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@ -1,34 +0,0 @@
language: cpp
sudo: true
matrix:
include:
- os: linux
env: CXXFLAGS="-std=c++11"
- os: osx
env: CXXFLAGS="-std=c++11"
install:
# install latest cmake
- |
if [ "$TRAVIS_OS_NAME" == "linux" ]; then
CMAKE_URL="https://cmake.org/files/v3.11/cmake-3.11.1-Linux-x86_64.tar.gz";
mkdir cmake_latest && travis_retry wget --no-check-certificate --quiet -O - ${CMAKE_URL} | tar --strip-components=1 -xz -C cmake_latest;
export PATH=$(pwd)/cmake_latest/bin:${PATH};
fi
- |
if [ "${TRAVIS_OS_NAME}" = "osx" ]; then
brew update;
brew uninstall cmake;
brew install cmake;
fi
- which ${CC}
- which ${CXX}
- which cmake
script:
- mkdir build
- cd build
- cmake -DBUILD_EXAMPLE=FALSE ..
- cmake --build .

@ -1,94 +1,75 @@
cmake_minimum_required(VERSION 3.13)
cmake_minimum_required(VERSION 3.11)
set(LIBNYQUIST_ROOT "${CMAKE_CURRENT_SOURCE_DIR}")
set(CMAKE_MODULE_PATH ${LIBNYQUIST_ROOT}/cmake)
include(CXXhelpers)
if (CMAKE_OSX_ARCHITECTURES)
if(CMAKE_OSX_SYSROOT MATCHES ".*iphoneos.*")
# RtAudio is not portable to ios currently
option(BUILD_EXAMPLE "Build example application" OFF)
else()
option(BUILD_EXAMPLE "Build example application" ON)
endif()
else()
option(BUILD_EXAMPLE "Build example application" ON)
endif()
option(BUILD_EXAMPLE "Build example application" ON)
#-------------------------------------------------------------------------------
# libopus
if (BUILD_LIBOPUS)
project(libopus)
project(libopus)
file(GLOB third_opus_src
"${LIBNYQUIST_ROOT}/third_party/opus/celt/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/opusfile/src/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/silk/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/silk/float/*.c"
)
set(lib_opus_src
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/analysis.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/mlp_data.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/mlp.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/opus_decoder.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/opus_multistream_decoder.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/opus_multistream_encoder.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/opus.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/repacketizer.c"
)
file(GLOB third_opus_src
"${LIBNYQUIST_ROOT}/third_party/opus/celt/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/libopus/src/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/opusfile/src/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/silk/*.c"
"${LIBNYQUIST_ROOT}/third_party/opus/silk/float/*.c"
)
add_library(libopus STATIC ${third_opus_src} ${lib_opus_src})
add_library(libopus STATIC ${third_opus_src})
set_cxx_version(libopus)
_set_compile_options(libopus)
set_cxx_version(libopus)
_set_compile_options(libopus)
if (WIN32)
_disable_warning(4244)
_disable_warning(4018)
endif()
target_include_directories(libopus PRIVATE
${LIBNYQUIST_ROOT}/third_party/libogg/include
${LIBNYQUIST_ROOT}/third_party/opus/celt
${LIBNYQUIST_ROOT}/third_party/opus/libopus/include
${LIBNYQUIST_ROOT}/third_party/opus/opusfile/include
${LIBNYQUIST_ROOT}/third_party/opus/opusfile/src/include
${LIBNYQUIST_ROOT}/third_party/opus/silk
${LIBNYQUIST_ROOT}/third_party/opus/silk/float)
if (MSVC_IDE)
# hack to get around the "Debug" and "Release" directories cmake tries to add on Windows
#set_target_properties(libnyquist PROPERTIES PREFIX "../")
set_target_properties(libopus PROPERTIES IMPORT_PREFIX "../")
endif()
target_compile_definitions(libopus PRIVATE OPUS_BUILD)
target_compile_definitions(libopus PRIVATE USE_ALLOCA)
set_target_properties(libopus
PROPERTIES
LIBRARY_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/lib"
ARCHIVE_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
RUNTIME_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
)
set_target_properties(libopus PROPERTIES OUTPUT_NAME_DEBUG libopus_d)
install (TARGETS libopus
LIBRARY DESTINATION lib
ARCHIVE DESTINATION lib
RUNTIME DESTINATION bin)
install (TARGETS libopus DESTINATION lib)
# folders
source_group(src FILES ${third_opus_src})
if (WIN32)
_disable_warning(4244)
_disable_warning(4018)
endif()
target_include_directories(libopus PRIVATE
${LIBNYQUIST_ROOT}/third_party/libogg/include
${LIBNYQUIST_ROOT}/third_party/opus/celt
${LIBNYQUIST_ROOT}/third_party/opus/libopus/include
${LIBNYQUIST_ROOT}/third_party/opus/opusfile/include
${LIBNYQUIST_ROOT}/third_party/opus/opusfile/src/include
${LIBNYQUIST_ROOT}/third_party/opus/silk
${LIBNYQUIST_ROOT}/third_party/opus/silk/float)
if (MSVC_IDE)
# hack to get around the "Debug" and "Release" directories cmake tries to add on Windows
#set_target_properties(libnyquist PROPERTIES PREFIX "../")
set_target_properties(libopus PROPERTIES IMPORT_PREFIX "../")
endif()
target_compile_definitions(libopus PRIVATE OPUS_BUILD)
target_compile_definitions(libopus PRIVATE USE_ALLOCA)
set_target_properties(libopus
PROPERTIES
LIBRARY_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/lib"
ARCHIVE_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
RUNTIME_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
)
set_target_properties(libopus PROPERTIES OUTPUT_NAME_DEBUG libopus_d)
install (TARGETS libopus
LIBRARY DESTINATION lib
ARCHIVE DESTINATION lib
RUNTIME DESTINATION bin)
install (TARGETS libopus DESTINATION lib)
# folders
source_group(src\ FILES ${third_opus_src})
#-------------------------------------------------------------------------------
# libwavpack
@ -163,7 +144,7 @@ install (TARGETS libwavpack DESTINATION lib)
#-------------------------------------------------------------------------------
# libnyquist static library
# libnyquist static library
project(libnyquist)
@ -172,7 +153,7 @@ file(GLOB nyquist_src "${LIBNYQUIST_ROOT}/src/*")
file(GLOB wavpack_src "${LIBNYQUIST_ROOT}/third_party/wavpack/src/*")
add_library(libnyquist STATIC
${nyquist_include}
${nyquist_include}
${nyquist_src}
)
@ -180,15 +161,13 @@ set_cxx_version(libnyquist)
_set_compile_options(libnyquist)
if (WIN32)
target_compile_definitions(libnyquist PRIVATE MODPLUG_STATIC)
_disable_warning(4244)
_disable_warning(4018)
endif()
target_include_directories(libnyquist
PUBLIC
$<INSTALL_INTERFACE:include>
$<BUILD_INTERFACE:${LIBNYQUIST_ROOT}/include>
PRIVATE
target_include_directories(libnyquist PRIVATE
${LIBNYQUIST_ROOT}/include
${LIBNYQUIST_ROOT}/include/libnyquist
${LIBNYQUIST_ROOT}/third_party
${LIBNYQUIST_ROOT}/third_party/FLAC/src/include
@ -221,7 +200,7 @@ set_target_properties(libnyquist
RUNTIME_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
)
target_link_libraries(libnyquist PRIVATE libwavpack)
#target_link_libraries(libnyquist libopus libwavpack)
install(TARGETS libnyquist
LIBRARY DESTINATION lib
@ -231,7 +210,7 @@ install(TARGETS libnyquist
install(TARGETS libnyquist DESTINATION lib)
# folders
source_group(src FILES ${nyquist_src})
source_group(src\ FILES ${nyquist_src})
#-------------------------------------------------------------------------------
@ -257,24 +236,15 @@ if(BUILD_EXAMPLE)
target_compile_definitions(${NQR_EXAMPLE_APP_NAME} PRIVATE __WINDOWS_WASAPI__)
elseif(APPLE)
target_compile_definitions(${NQR_EXAMPLE_APP_NAME} PRIVATE __MACOSX_CORE__)
elseif(LIBNYQUIST_JACK)
target_compile_definitions(${NQR_EXAMPLE_APP_NAME} PRIVATE __UNIX_JACK__)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PRIVATE jack pthread)
elseif(LIBNYQUIST_PULSE)
target_compile_definitions(${NQR_EXAMPLE_APP_NAME} PRIVATE __LINUX_PULSE__)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PRIVATE pulse pthread)
elseif(LIBNYQUIST_ASOUND)
target_compile_definitions(${NQR_EXAMPLE_APP_NAME} PRIVATE __LINUX_ALSA__)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PRIVATE asound pthread)
else()
message(FATAL, "On Linux, one of LIBNYQUIST_JACK, LIBNYQUIST_PULSE, or LIBNYQUIST_ASOUND must be set.")
message(FATAL, "Select an appropriate backend, such as __LINUX_ALSA__")
endif()
target_include_directories(${NQR_EXAMPLE_APP_NAME} PRIVATE
${LIBNYQUIST_ROOT}/include
${LIBNYQUIST_ROOT}/examples/src
${LIBNYQUIST_ROOT}/third_party
)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PRIVATE libnyquist)
set_target_properties(${NQR_EXAMPLE_APP_NAME}
PROPERTIES
@ -283,14 +253,6 @@ if(BUILD_EXAMPLE)
RUNTIME_OUTPUT_DIRECTORY "${CMAKE_BINARY_DIR}/bin"
)
if(APPLE)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PRIVATE
"-framework AudioToolbox"
"-framework AudioUnit"
"-framework Accelerate"
"-framework CoreAudio"
"-framework Cocoa"
)
ENDIF(APPLE)
target_link_libraries(${NQR_EXAMPLE_APP_NAME} PUBLIC libnyquist libopus libwavpack)
endif()

@ -1,4 +1,4 @@
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:

@ -1,14 +1,10 @@
# Libnyquist
[![Build status](https://ci.appveyor.com/api/projects/status/2xeuyuxy618ndf4r?svg=true)](https://ci.appveyor.com/project/ddiakopoulos/libnyquist)
Platform | Build Status |
-------- | ------------ |
Microsoft VS2017 x64 | [![Build status](https://ci.appveyor.com/api/projects/status/2xeuyuxy618ndf4r?svg=true)](https://ci.appveyor.com/project/ddiakopoulos/libnyquist) |
Clang (OSX) & GCC (Linux) | [![Build Status](https://travis-ci.org/ddiakopoulos/libnyquist.svg?branch=master)](https://travis-ci.org/ddiakopoulos/libnyquist) |
Libnyquist is a small C++11 library for reading sampled audio data from disk or memory. It's ideal to use as an audio asset frontend for games, audio sequencers, music players, and more.
Libnyquist is a small C++11 library for reading sampled audio data from disk or memory. It is intended to be used an audio loading frontend for games, audio sequencers, music players, and more.
The library does not include patent or license encumbered formats (such as AAC). For portability, libnyquist does not link against platform-specific APIs like Windows Media Foundation or CoreAudio, and instead bundles the source code of reference decoders as an implementation detail.
The library steers away from patent or GPL license encumbered formats (such as MP3 and AAC). For portability, libnyquist does not link against platform-specific APIs like Windows Media Foundation or CoreAudio, and instead bundles the source code of reference decoders as an implementation detail.
Libnyquist is meant to be statically linked, which is not the case with other popular libraries like libsndfile (which is licensed under the LGPL). Furthermore, the library is not concerned with supporting very rare encodings (for instance, A-law PCM or the SND format).
@ -16,15 +12,19 @@ While untested, there are no technical conditions that preclude compilation on o
## Format Support
Regardless of input bit depth, the library produces a channel-interleaved float vector, normalized between [-1.0,+1.0]. At present, the library does not provide comprehensive resampling functionality.
Regardless of input bit depth, the library hands over an interleaved float array, normalized between [-1.0,+1.0]. At present, the library does not provide resampling functionality.
* Wave (+ IMA-ADPCM encoding)
* MP3
* Ogg Vorbis
* Ogg Opus
* FLAC
* WavPack
* Musepack
* [MIDI files (with soundfonts)](midi-playback.md)
* libmodplug formats (669, amf, ams, dbm, dmf, dsm, far, it, j2b, mdl, med, mod, mt2, mtm, okt, pat, psm, ptm, s3m, stm, ult, umx, xm)
## Encoding
Simple but robust WAV format encoder now included. Extentions in the near future might include Ogg.
## Known Issues & Bugs
* See the Github issue tracker.

@ -2,12 +2,12 @@
# http://www.appveyor.com/docs/appveyor-yml
os: Visual Studio 2017
environment:
VisualStudioVersion: 14.0
platform:
- x64
configuration: Release
platform: x64
build_script:
- mkdir build
- cd build
- cmake -G "Visual Studio 15 2017 Win64" ..
- cmake --build . --config Release
build:
verbosity: minimal
project: libnyquist.vcxproj/v141/libnyquist.sln

@ -14,7 +14,7 @@ function(_disable_warning flag)
endfunction()
function(_set_compile_options proj)
if(MSVC)
if (WIN32)
target_compile_options(${proj} PRIVATE /arch:AVX /Zi )
endif()
endfunction()

@ -0,0 +1,8 @@
## MIDI Player Functionality
A SoundFont `.sf2` file is a file that contains many samples of different instruments playing different notes, enabling Libnyquist to render finally each instrument sound in the MIDI song. SoundFonts are available on the web, but the quality and size varies a lot, and they may have licensing issues (because of the copyright of the different samples), they may be designed for different styles of music (classical, jazz, pop...), and they may target different sets of instruments as well. As a general rule, most MIDI files are targeted to use the 128 sounds of [General MIDI standard (GM)](http://en.wikipedia.org/wiki/General_MIDI), so use a SoundFont that does conform to this standard. You probably want to try the [Fluid R3 GM (141 MB uncompressed)](http://google.com/search?q=Fluid+R3+GM+soundfont) as a starter pack, which is a very good general purpose SoundFont (and fully MIT licensed).
SoundFont files are large and often compressed in `.zip`, `.tar.gz` or `.sfArk` formats. `.sfArk` files can be unpacked with [sfark tool](http://www.melodymachine.com/sfark.htm). For now, libnyquist supports `.pat` (GUS) soundfonts only, which can be converted from `.sf2` files with the [unsf tool](http://alsa.opensrc.org/Unsf).
Back to the library, the compiler directive `PAT_CONFIG_FILE` tells libnyquist what soundfont file is hardcoded by default (defaults to `pat/timidity.cfg`, but might be `FluidR3_GM2/FluidR3_GM2-2.cfg`, or any other valid path during compilation). Additionally, there is a runtime check for the environment variable `MMPAT_PATH_TO_CFG`, which contains the path to the config file. For more info please check [this source file](third_party/libmodplug/src/load_pat.cpp)

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -37,12 +37,12 @@ static int rt_callback(void * output_buffer, void * input_buffer, uint32_t num_b
{
if (status) std::cerr << "[rtaudio] buffer over or underflow" << std::endl;
// Playback
// Playback
if (buffer.getAvailableRead()) buffer.read((float*) output_buffer, BUFFER_LENGTH);
else memset(output_buffer, 0, BUFFER_LENGTH * sizeof(float));
// Recording
if (record_buffer.getAvailableWrite()) record_buffer.write((float*) input_buffer, BUFFER_LENGTH / 2);
// Recording
if (record_buffer.getAvailableWrite()) record_buffer.write((float*) input_buffer, BUFFER_LENGTH / 2);
return 0;
}
@ -62,9 +62,9 @@ AudioDevice::~AudioDevice()
{
rtaudio->stopStream();
if (rtaudio->isStreamOpen())
{
{
rtaudio->closeStream();
}
}
}
}
@ -77,7 +77,7 @@ bool AudioDevice::Open(const int deviceId)
outputParams.nChannels = info.numChannels;
outputParams.firstChannel = 0;
RtAudio::StreamParameters inputParams;
RtAudio::StreamParameters inputParams;
inputParams.deviceId = rtaudio->getDefaultInputDevice();
inputParams.nChannels = 1;
inputParams.firstChannel = 0;
@ -129,21 +129,21 @@ bool AudioDevice::Play(const std::vector<float> & data)
bool AudioDevice::Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer)
{
uint64_t recordedSamples = 0;
uint64_t recordedSamples = 0;
// Allocate memory upfront (revisit this later...)
recordingBuffer.resize(lengthInSamples + (BUFFER_LENGTH)); // + a little padding
// Allocate memory upfront (revisit this later...)
recordingBuffer.resize(lengthInSamples + (BUFFER_LENGTH)); // + a little padding
while (recordedSamples < lengthInSamples)
{
if (record_buffer.getAvailableRead())
{
if (record_buffer.read(recordingBuffer.data() + recordedSamples, BUFFER_LENGTH / 2))
{
recordedSamples += (BUFFER_LENGTH / 2);
}
}
}
while (recordedSamples < lengthInSamples)
{
if (record_buffer.getAvailableRead())
{
if (record_buffer.read(recordingBuffer.data() + recordedSamples, BUFFER_LENGTH / 2))
{
recordedSamples += (BUFFER_LENGTH / 2);
}
}
}
return true;
return true;
}

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -39,27 +39,27 @@ static const int32_t BUFFER_LENGTH = FRAME_SIZE * CHANNELS;
struct AudioDeviceInfo
{
uint32_t id;
uint32_t numChannels;
uint32_t sampleRate;
uint32_t frameSize;
bool isPlaying = false;
uint32_t id;
uint32_t numChannels;
uint32_t sampleRate;
uint32_t frameSize;
bool isPlaying = false;
};
class AudioDevice
{
std::unique_ptr<RtAudio> rtaudio;
std::unique_ptr<RtAudio> rtaudio;
protected:
AudioDevice(const AudioDevice& r) = delete;
AudioDevice & operator = (const AudioDevice& r) = delete;
AudioDevice(const AudioDevice& r) = delete;
AudioDevice & operator = (const AudioDevice& r) = delete;
public:
AudioDeviceInfo info;
AudioDevice(int numChannels, int sampleRate, int deviceId = -1);
~AudioDevice();
static void ListAudioDevices();
bool Open(const int deviceId);
bool Play(const std::vector<float> & data);
bool Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer);
AudioDeviceInfo info;
AudioDevice(int numChannels, int sampleRate, int deviceId = -1);
~AudioDevice();
static void ListAudioDevices();
bool Open(const int deviceId);
bool Play(const std::vector<float> & data);
bool Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer);
};
#endif

@ -1,5 +1,5 @@
// Note to Visual Studio / Windows users: you must set the working directory manually on the project file
// to $(ProjectDir)../ since these settings are not saved directly in project. The loader
// to $(ProjectDir)../../../ since these settings are not saved directly in project. The loader
// will be unable to find the example assets unless the proper working directory is set.
#if defined(_MSC_VER)
@ -8,8 +8,9 @@
#include "AudioDevice.h"
#include "libnyquist/Decoders.h"
#include "libnyquist/Encoders.h"
#include "libnyquist/AudioDecoder.h"
#include "libnyquist/WavEncoder.h"
#include "libnyquist/PostProcess.h"
#include <thread>
@ -17,151 +18,138 @@ using namespace nqr;
int main(int argc, const char **argv) try
{
AudioDevice::ListAudioDevices();
AudioDevice::ListAudioDevices();
const int desiredSampleRate = 44100;
const int desiredChannelCount = 2;
AudioDevice myDevice(desiredChannelCount, desiredSampleRate);
myDevice.Open(myDevice.info.id);
int desiredSampleRate = 44100;
AudioDevice myDevice(2, desiredSampleRate);
myDevice.Open(myDevice.info.id);
std::shared_ptr<AudioData> fileData = std::make_shared<AudioData>();
std::shared_ptr<AudioData> fileData = std::make_shared<AudioData>();
NyquistIO loader;
NyquistIO loader;
if (argc > 1)
{
std::string cli_arg = std::string(argv[1]);
loader.Load(fileData.get(), cli_arg);
}
else
{
// Circular libnyquist testing
//loader.Load(fileData.get(), "libnyquist_example_output.opus");
if (argc > 1)
{
std::string cli_arg = std::string(argv[1]);
loader.Load(fileData.get(), cli_arg);
}
else
{
// Circular libnyquist testing
//loader.Load(fileData.get(), "encoded.opus");
// 1-channel wave
//loader.Load(fileData.get(), "test_data/1ch/44100/8/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/16/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/24/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/32/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/64/test.wav");
// 1-channel wave
//loader.Load(fileData.get(), "test_data/1ch/44100/8/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/16/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/24/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/32/test.wav");
//loader.Load(fileData.get(), "test_data/1ch/44100/64/test.wav");
// 2-channel wave
//loader.Load(fileData.get(), "test_data/2ch/44100/8/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/16/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/24/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/32/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/64/test.wav");
// 2-channel wave
//loader.Load(fileData.get(), "test_data/2ch/44100/8/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/16/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/24/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/32/test.wav");
//loader.Load(fileData.get(), "test_data/2ch/44100/64/test.wav");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_mono-ima4-reaper.wav");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_stereo-ima4-reaper.wav");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_mono-ima4-reaper.wav");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_stereo-ima4-reaper.wav");
// Multi-channel wave
//loader.Load(fileData.get(), "test_data/ad_hoc/6_channel_44k_16b.wav");
// Multi-channel wave
//loader.Load(fileData.get(), "test_data/ad_hoc/6_channel_44k_16b.wav");
// 1 + 2 channel ogg
//loader.Load(fileData.get(), "test_data/ad_hoc/LR_Stereo.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestLaugh_44k.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeatMono.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/BlockWoosh_Stereo.ogg");
// 1 + 2 channel flac
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr8_Stereo_Dithered.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Stereo.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Mono.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr24_Stereo.flac");
// 1 + 2 channel ogg
//loader.Load(fileData.get(), "test_data/ad_hoc/LR_Stereo.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestLaugh_44k.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeatMono.ogg");
//loader.Load(fileData.get(), "test_data/ad_hoc/BlockWoosh_Stereo.ogg");
// 1 + 2 channel flac
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr8_Stereo_Dithered.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Stereo.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Mono.flac");
//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr24_Stereo.flac");
//auto memory = ReadFile("test_data/ad_hoc/KittyPurr24_Stereo.flac"); // broken
//loader.Load(fileData.get(), "flac", memory.buffer); // broken
// Single-channel opus
//loader.Load(fileData.get(), "test_data/ad_hoc/detodos.opus"); // "Firefox: From All, To All"
// Single-channel opus
//loader.Load(fileData.get(), "test_data/ad_hoc/detodos.opus"); // "Firefox: From All, To All"
// 1 + 2 channel wavpack
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32_Mono.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int16.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int32.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24_Mono.wv");
// 1 + 2 channel wavpack
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32_Mono.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int16.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int32.wv");
//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24_Mono.wv");
// In-memory wavpack
auto memory = ReadFile("test_data/ad_hoc/TestBeat_Float32.wv");
loader.Load(fileData.get(), "wv", memory.buffer);
// 1 + 2 channel musepack
//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_stereo.mpc");
//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_mono.mpc");
// 1 + 2 channel musepack
//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_stereo.mpc");
//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_mono.mpc");
// In-memory ogg
//auto memory = ReadFile("test_data/ad_hoc/BlockWoosh_Stereo.ogg");
//loader.Load(fileData.get(), "ogg", memory.buffer);
auto memory = ReadFile("test_data/ad_hoc/BlockWoosh_Stereo.ogg");
loader.Load(fileData.get(), "ogg", memory.buffer);
}
// In-memory Mp3
//auto memory = ReadFile("test_data/ad_hoc/acetylene.mp3");
//loader.Load(fileData.get(), "mp3", memory.buffer);
}
/*
// Test Recording Capabilities of AudioDevice
fileData->samples.reserve(44100 * 5);
fileData->channelCount = 1;
fileData->frameSize = 32;
fileData->lengthSeconds = 5.0;
fileData->sampleRate = 44100;
std::cout << "Starting recording ..." << std::endl;
myDevice.Record(fileData->sampleRate * fileData->lengthSeconds, fileData->samples);
*/
/* Test Recording Capabilities of AudioDevice
fileData->samples.reserve(44100 * 5);
fileData->channelCount = 1;
fileData->frameSize = 32;
fileData->lengthSeconds = 5.0;
fileData->sampleRate = 44100;
std::cout << "Starting recording ..." << std::endl;
myDevice.Record(fileData->sampleRate * fileData->lengthSeconds, fileData->samples);
*/
if (fileData->sampleRate != desiredSampleRate)
{
std::cout << "[Warning - Sample Rate Mismatch] - file is sampled at " << fileData->sampleRate << " and output is " << desiredSampleRate << std::endl;
}
if (fileData->sampleRate != desiredSampleRate)
{
std::cout << "[Warning - Sample Rate Mismatch] - file is sampled at " << fileData->sampleRate << " and output is " << desiredSampleRate << std::endl;
}
// Resample
std::vector<float> outputBuffer;
outputBuffer.reserve(fileData->samples.size());
linear_resample(44100.0 / 48000.0, fileData->samples, outputBuffer, (uint32_t) fileData->samples.size());
std::cout << "Input Samples: " << fileData->samples.size() << std::endl;
std::cout << "Input Samples: " << fileData->samples.size() << std::endl;
std::cout << "Output Samples: " << outputBuffer.size() << std::endl;
// Convert mono to stereo for testing playback
if (fileData->channelCount == 1)
{
std::cout << "Playing MONO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
std::vector<float> stereoCopy(fileData->samples.size() * 2);
MonoToStereo(fileData->samples.data(), stereoCopy.data(), fileData->samples.size());
myDevice.Play(stereoCopy);
}
else
{
std::cout << "Playing STEREO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
myDevice.Play(fileData->samples);
}
// Convert mono to stereo for testing playback
if (fileData->channelCount == 1)
{
std::cout << "Playing MONO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
std::vector<float> stereoCopy(fileData->samples.size() * 2);
MonoToStereo(fileData->samples.data(), stereoCopy.data(), fileData->samples.size());
myDevice.Play(stereoCopy);
}
else
{
std::cout << "Playing for: " << fileData->lengthSeconds << " seconds..." << std::endl;
myDevice.Play(fileData->samples);
}
// Test Opus Encoding
{
// Resample
std::vector<float> outputBuffer;
std::cout << "Output Samples: " << outputBuffer.size() << std::endl;
fileData->samples = outputBuffer;
int encoderStatus = encode_opus_to_disk({ 1, PCM_FLT, DITHER_NONE }, fileData.get(), "encoded.opus");
std::cout << "Encoder Status: " << encoderStatus << std::endl;
outputBuffer.reserve(fileData->samples.size() * 2);
linear_resample(fileData->sampleRate / 48000.0f, fileData->samples, outputBuffer, (uint32_t)fileData->samples.size());
fileData->samples = outputBuffer;
int encoderStatus = encode_opus_to_disk({ fileData->channelCount, PCM_FLT, DITHER_NONE }, fileData.get(), "libnyquist_example_output.opus");
std::cout << "Encoder Status: " << encoderStatus << std::endl;
}
return EXIT_SUCCESS;
return EXIT_SUCCESS;
}
catch (const UnsupportedExtensionEx & e)
{
std::cerr << "Caught: " << e.what() << std::endl;
std::cerr << "Caught: " << e.what() << std::endl;
}
catch (const LoadPathNotImplEx & e)
{
std::cerr << "Caught: " << e.what() << std::endl;
std::cerr << "Caught: " << e.what() << std::endl;
}
catch (const LoadBufferNotImplEx & e)
{
std::cerr << "Caught: " << e.what() << std::endl;
std::cerr << "Caught: " << e.what() << std::endl;
}
catch (const std::exception & e)
{
std::cerr << "Caught: " << e.what() << std::endl;
std::cerr << "Caught: " << e.what() << std::endl;
}

@ -36,8 +36,6 @@
#include <atomic>
#include <assert.h>
#include <cstring>
#include <cstdlib>
template <typename T>
class RingBufferT
@ -45,7 +43,7 @@ class RingBufferT
public:
// Constructs a RingBufferT with size = 0
RingBufferT() : mData(nullptr), mAllocatedSize(0), mWriteIndex(0), mReadIndex(0) {}
RingBufferT() : mData(nullptr), mAllocatedSize(0), mWriteIndex(0), mReadIndex(0) {}
// Constructs a RingBufferT with \a count maximum elements.
RingBufferT(size_t count) : mAllocatedSize(0) { resize(count); }
@ -67,9 +65,9 @@ class RingBufferT
size_t allocatedSize = count + 1; // one bin is used to distinguish between the read and write indices when full.
if (mAllocatedSize)
mData = (T *) std::realloc(mData, allocatedSize * sizeof(T));
mData = (T *)::realloc(mData, allocatedSize * sizeof(T));
else
mData = (T *) std::calloc(allocatedSize, sizeof(T));
mData = (T *)::calloc(allocatedSize, sizeof(T));
assert(mData);

@ -0,0 +1,83 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef AUDIO_DECODER_H
#define AUDIO_DECODER_H
#include "Common.h"
#include <utility>
#include <map>
#include <memory>
#include <exception>
namespace nqr
{
struct UnsupportedExtensionEx : public std::runtime_error
{
UnsupportedExtensionEx() : std::runtime_error("Unsupported file extension") {}
};
struct LoadPathNotImplEx : public std::runtime_error
{
LoadPathNotImplEx() : std::runtime_error("Loading from path not implemented") {}
};
struct LoadBufferNotImplEx : public std::runtime_error
{
LoadBufferNotImplEx() : std::runtime_error("Loading from buffer not implemented") {}
};
struct BaseDecoder
{
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) = 0;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) = 0;
virtual std::vector<std::string> GetSupportedFileExtensions() = 0;
virtual ~BaseDecoder() {}
};
typedef std::pair<std::string, std::shared_ptr<nqr::BaseDecoder>> DecoderPair;
class NyquistIO
{
std::string ParsePathForExtension(const std::string & path) const;
std::shared_ptr<nqr::BaseDecoder> GetDecoderForExtension(const std::string & ext);
void BuildDecoderTable();
void AddDecoderToTable(std::shared_ptr<nqr::BaseDecoder> decoder);
std::map<std::string, std::shared_ptr<BaseDecoder>> decoderTable;
NO_MOVE(NyquistIO);
public:
NyquistIO();
~NyquistIO();
void Load(AudioData * data, const std::string & path);
void Load(AudioData * data, const std::string & extension, const std::vector<uint8_t> & buffer);
bool IsFileSupported(const std::string & path) const;
};
} // end namespace nqr
#endif

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -37,12 +37,13 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include <type_traits>
#include <numeric>
#include <array>
#include <map>
#include <random>
#include "PostProcess.h"
#include "Dither.h"
namespace nqr
{
/////////////////
// Util Macros //
/////////////////
@ -255,35 +256,6 @@ inline void hermite_resample(const double rate, const std::vector<float> & input
// Conversion Utilities //
//////////////////////////
enum DitherType
{
DITHER_NONE,
DITHER_TRIANGLE
};
class Dither
{
std::uniform_real_distribution<float> distribution;
std::mt19937 gen;
float previous;
DitherType d;
public:
Dither(DitherType d) : distribution(-0.5f, +0.5f), previous(0.f), d(d) {}
float operator()(float s)
{
if (d == DITHER_TRIANGLE)
{
const float value = distribution(gen);
s = s + value - previous;
previous = value;
return s;
}
else return s;
}
};
// Signed maxes, defined for readabilty/convenience
#define NQR_INT16_MAX 32767.f
#define NQR_INT24_MAX 8388608.f
@ -331,9 +303,6 @@ void ConvertToFloat32(float * dst, const int16_t * src, const size_t N, PCMForma
void ConvertFromFloat32(uint8_t * dst, const float * src, const size_t N, PCMFormat f, DitherType t = DITHER_NONE);
int GetFormatBitsPerSample(PCMFormat f);
PCMFormat MakeFormatForBits(int bits, bool floatingPt, bool isSigned);
//////////////////////////
// User Data + File Ops //
//////////////////////////
@ -367,327 +336,9 @@ struct NyquistFileBuffer
};
NyquistFileBuffer ReadFile(const std::string & pathToFile);
////////////////////
// Encoding Utils //
////////////////////
struct EncoderParams
{
int channelCount;
PCMFormat targetFormat;
DitherType dither;
};
enum EncoderError
{
NoError,
InsufficientSampleData,
FileIOError,
UnsupportedSamplerate,
UnsupportedChannelConfiguration,
UnsupportedBitdepth,
UnsupportedChannelMix,
BufferTooBig,
};
//////////////////////
// Wav Format Utils //
//////////////////////
enum WaveFormatCode
{
FORMAT_UNKNOWN = 0x0, // Unknown Wave Format
FORMAT_PCM = 0x1, // PCM Format
FORMAT_ADPCM = 0x2, // Microsoft ADPCM Format
FORMAT_IEEE = 0x3, // IEEE float/double
FORMAT_ALAW = 0x6, // 8-bit ITU-T G.711 A-law
FORMAT_MULAW = 0x7, // 8-bit ITU-T G.711 µ-law
FORMAT_IMA_ADPCM = 0x11, // IMA ADPCM Format
FORMAT_EXT = 0xFFFE // Set via subformat
};
struct RiffChunkHeader
{
uint32_t id_riff; // Chunk ID: 'RIFF'
uint32_t file_size; // Entire file in bytes
uint32_t id_wave; // Chunk ID: 'WAVE'
};
struct WaveChunkHeader
{
uint32_t fmt_id; // Chunk ID: 'fmt '
uint32_t chunk_size; // Size in bytes
uint16_t format; // Format code
uint16_t channel_count; // Num interleaved channels
uint32_t sample_rate; // SR
uint32_t data_rate; // Data rate
uint16_t frame_size; // 1 frame = channels * bits per sample (also known as block align)
uint16_t bit_depth; // Bits per sample
};
struct BextChunk
{
uint32_t fmt_id; // Chunk ID: 'bext'
uint32_t chunk_size; // Size in bytes
uint8_t description[256]; // Description of the sound (ascii)
uint8_t origin[32]; // Name of the originator (ascii)
uint8_t origin_ref[32]; // Reference of the originator (ascii)
uint8_t orgin_date[10]; // yyyy-mm-dd (ascii)
uint8_t origin_time[8]; // hh-mm-ss (ascii)
uint64_t time_ref; // First sample count since midnight
uint32_t version; // Version of the BWF
uint8_t uimd[64]; // Byte 0 of SMPTE UMID
uint8_t reserved[188]; // 190 bytes, reserved for future use & set to NULL
};
struct FactChunk
{
uint32_t fact_id; // Chunk ID: 'fact'
uint32_t chunk_size; // Size in bytes
uint32_t sample_length; // number of samples per channel
};
struct ExtensibleData
{
uint16_t size;
uint16_t valid_bits_per_sample;
uint32_t channel_mask;
struct GUID
{
uint32_t data0;
uint16_t data1;
uint16_t data2;
uint16_t data3;
uint8_t data4[6];
};
};
template<class C, class R>
std::basic_ostream<C,R> & operator << (std::basic_ostream<C,R> & a, const WaveChunkHeader & b)
{
return a <<
"Format ID:\t\t" << b.fmt_id <<
"\nChunk Size:\t\t" << b.chunk_size <<
"\nFormat Code:\t\t" << b.format <<
"\nChannels:\t\t" << b.channel_count <<
"\nSample Rate:\t\t" << b.sample_rate <<
"\nData Rate:\t\t" << b.data_rate <<
"\nFrame Size:\t\t" << b.frame_size <<
"\nBit Depth:\t\t" << b.bit_depth << std::endl;
}
//@todo expose speaker/channel/layout masks in the API:
enum SpeakerChannelMask
{
SPEAKER_FRONT_LEFT = 0x00000001,
SPEAKER_FRONT_RIGHT = 0x00000002,
SPEAKER_FRONT_CENTER = 0x00000004,
SPEAKER_LOW_FREQUENCY = 0x00000008,
SPEAKER_BACK_LEFT = 0x00000010,
SPEAKER_BACK_RIGHT = 0x00000020,
SPEAKER_FRONT_LEFT_OF_CENTER = 0x00000040,
SPEAKER_FRONT_RIGHT_OF_CENTER = 0x00000080,
SPEAKER_BACK_CENTER = 0x00000100,
SPEAKER_SIDE_LEFT = 0x00000200,
SPEAKER_SIDE_RIGHT = 0x00000400,
SPEAKER_TOP_CENTER = 0x00000800,
SPEAKER_TOP_FRONT_LEFT = 0x00001000,
SPEAKER_TOP_FRONT_CENTER = 0x00002000,
SPEAKER_TOP_FRONT_RIGHT = 0x00004000,
SPEAKER_TOP_BACK_LEFT = 0x00008000,
SPEAKER_TOP_BACK_CENTER = 0x00010000,
SPEAKER_TOP_BACK_RIGHT = 0x00020000,
SPEAKER_RESERVED = 0x7FFC0000,
SPEAKER_ALL = 0x80000000
};
enum SpeakerLayoutMask
{
SPEAKER_MONO = (SPEAKER_FRONT_CENTER),
SPEAKER_STEREO = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT),
SPEAKER_2POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY),
SPEAKER_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER),
SPEAKER_QUAD = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_4POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_5POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_7POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER),
SPEAKER_5POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT),
SPEAKER_7POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT),
};
//@todo verify mask values
inline int ComputeChannelMask(const size_t channels)
{
switch (channels)
{
case 1: return SPEAKER_MONO;
case 2: return SPEAKER_STEREO;
case 3: return SPEAKER_2POINT1;
case 4: return SPEAKER_QUAD;
case 5: return SPEAKER_4POINT1;
case 6: return SPEAKER_5POINT1;
default: return -1;
}
}
/////////////////////
// Chunk utilities //
/////////////////////
struct ChunkHeaderInfo
{
uint32_t offset; // Byte offset into chunk
uint32_t size; // Size of the chunk in bytes
};
inline uint32_t GenerateChunkCode(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
{
#ifdef ARCH_CPU_LITTLE_ENDIAN
return ((uint32_t)((a) | ((b) << 8) | ((c) << 16) | (((uint32_t)(d)) << 24)));
#else
return ((uint32_t)((((uint32_t)(a)) << 24) | ((b) << 16) | ((c) << 8) | (d)));
#endif
}
inline char * GenerateChunkCodeChar(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
{
auto chunk = GenerateChunkCode(a, b, c, d);
char * outArr = new char[4];
uint32_t t = 0x000000FF;
for (size_t i = 0; i < 4; i++)
{
outArr[i] = chunk & t;
chunk >>= 8;
}
return outArr;
}
inline ChunkHeaderInfo ScanForChunk(const std::vector<uint8_t> & fileData, uint32_t chunkMarker)
{
// D[n] aligned to 16 bytes now
const uint16_t * d = reinterpret_cast<const uint16_t *>(fileData.data());
for (size_t i = 0; i < fileData.size() / sizeof(uint16_t); i++)
{
// This will be in machine endianess
uint32_t m = Pack(Read16(d[i]), Read16(d[i + 1]));
if (m == chunkMarker)
{
uint32_t cSz = Pack(Read16(d[i + 2]), Read16(d[i + 3]));
return { (uint32_t(i * sizeof(uint16_t))), cSz }; // return i in bytes to the start of the data
}
else continue;
}
return { 0, 0 };
};
inline WaveChunkHeader MakeWaveHeader(const EncoderParams param, const int sampleRate)
{
WaveChunkHeader header;
int bitdepth = GetFormatBitsPerSample(param.targetFormat);
header.fmt_id = GenerateChunkCode('f', 'm', 't', ' ');
header.chunk_size = 16;
header.format = (param.targetFormat <= PCMFormat::PCM_32) ? WaveFormatCode::FORMAT_PCM : WaveFormatCode::FORMAT_IEEE;
header.channel_count = param.channelCount;
header.sample_rate = sampleRate;
header.data_rate = sampleRate * param.channelCount * (bitdepth / 8);
header.frame_size = param.channelCount * (bitdepth / 8);
header.bit_depth = bitdepth;
return header;
}
// @todo expose this in the FLAC API
inline std::map<int, std::string> GetFlacQualityTable()
{
return {
{ 0, "0 (Fastest)" },
{ 1, "1" },
{ 2, "2" },
{ 3, "3" },
{ 4, "4" },
{ 5, "5 (Default)" },
{ 6, "6" },
{ 7, "7" },
{ 8, "8 (Highest)" },
};
}
template <typename T>
inline void DeinterleaveStereo(T * c1, T * c2, T const * src, size_t count)
{
auto src_end = src + count;
while (src != src_end)
{
*c1 = src[0];
*c2 = src[1];
c1++;
c2++;
src += 2;
}
}
template<typename T>
void InterleaveChannels(const T * src, T * dest, size_t numFramesPerChannel, size_t numChannels, size_t N)
{
for (size_t ch = 0; ch < numChannels; ch++)
{
size_t x = ch;
const T * srcChannel = &src[ch * numFramesPerChannel];
for (size_t i = 0; i < N; i++)
{
dest[x] = srcChannel[i];
x += numChannels;
}
}
}
template<typename T>
void DeinterleaveChannels(const T * src, T * dest, size_t numFramesPerChannel, size_t numChannels, size_t N)
{
for (size_t ch = 0; ch < numChannels; ch++)
{
size_t x = ch;
T *destChannel = &dest[ch * numFramesPerChannel];
for (size_t i = 0; i < N; i++)
{
destChannel[i] = (T)src[x];
x += numChannels;
}
}
}
template <typename T>
void StereoToMono(const T * src, T * dest, size_t N)
{
for (size_t i = 0, j = 0; i < N; i += 2, ++j)
{
dest[j] = (src[i] + src[i + 1]) / 2.0f;
}
}
template <typename T>
void MonoToStereo(const T * src, T * dest, size_t N)
{
for (size_t i = 0, j = 0; i < N; ++i, j += 2)
{
dest[j] = src[i];
dest[j + 1] = src[i];
}
}
inline void TrimSilenceInterleaved(std::vector<float> & buffer, float v, bool fromFront, bool fromEnd)
{
//@todo implement me!
}
int GetFormatBitsPerSample(PCMFormat f);
PCMFormat MakeFormatForBits(int bits, bool floatingPt, bool isSigned);
} // end namespace nqr

@ -1,136 +0,0 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef AUDIO_DECODER_H
#define AUDIO_DECODER_H
#include "Common.h"
#include <utility>
#include <map>
#include <memory>
#include <exception>
namespace nqr
{
struct BaseDecoder
{
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) = 0;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) = 0;
virtual std::vector<std::string> GetSupportedFileExtensions() = 0;
virtual ~BaseDecoder() {}
};
typedef std::pair< std::string, std::shared_ptr<nqr::BaseDecoder> > DecoderPair;
class NyquistIO
{
std::string ParsePathForExtension(const std::string & path) const;
std::shared_ptr<nqr::BaseDecoder> GetDecoderForExtension(const std::string & ext);
void BuildDecoderTable();
void AddDecoderToTable(std::shared_ptr<nqr::BaseDecoder> decoder);
std::map< std::string, std::shared_ptr<BaseDecoder> > decoderTable;
NO_MOVE(NyquistIO);
public:
NyquistIO();
~NyquistIO();
void Load(AudioData * data, const std::string & path);
void Load(AudioData * data, const std::vector<uint8_t> & buffer);
void Load(AudioData * data, const std::string & extension, const std::vector<uint8_t> & buffer);
bool IsFileSupported(const std::string & path) const;
};
struct UnsupportedExtensionEx : public std::runtime_error { UnsupportedExtensionEx() : std::runtime_error("Unsupported file extension") {} };
struct LoadPathNotImplEx : public std::runtime_error { LoadPathNotImplEx() : std::runtime_error("Loading from path not implemented") {} };
struct LoadBufferNotImplEx : public std::runtime_error { LoadBufferNotImplEx() : std::runtime_error("Loading from buffer not implemented") {} };
struct WavDecoder final : public nqr::BaseDecoder
{
WavDecoder() = default;
virtual ~WavDecoder() {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct WavPackDecoder final : public nqr::BaseDecoder
{
WavPackDecoder() = default;
virtual ~WavPackDecoder() override {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct VorbisDecoder final : public nqr::BaseDecoder
{
VorbisDecoder() = default;
virtual ~VorbisDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct OpusDecoder final : public nqr::BaseDecoder
{
OpusDecoder() = default;
virtual ~OpusDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct MusepackDecoder final : public nqr::BaseDecoder
{
MusepackDecoder() = default;
virtual ~MusepackDecoder() override {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct Mp3Decoder final : public nqr::BaseDecoder
{
Mp3Decoder() = default;
virtual ~Mp3Decoder() override {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
struct FlacDecoder final : public nqr::BaseDecoder
{
FlacDecoder() = default;
virtual ~FlacDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,69 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef DITHER_OPERATIONS_H
#define DITHER_OPERATIONS_H
#include <random>
namespace nqr
{
enum DitherType
{
DITHER_NONE,
DITHER_TRIANGLE
};
class Dither
{
std::uniform_real_distribution<float> distribution;
std::mt19937 rndGen;
float prev = 0.0f;
DitherType d;
public:
Dither(DitherType d) : distribution(-0.5f, +0.5f), d(d) {}
float operator()(float s)
{
if (d == DITHER_TRIANGLE)
{
const float value = distribution(rndGen);
s = s + value - prev;
prev = value;
return s;
}
else
{
return s;
}
}
};
} // end namespace nqr
#endif

@ -0,0 +1,65 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC_DECODER_H
#define FLAC_DECODER_H
// http://lists.xiph.org/pipermail/flac-dev/2012-March/003276.html
#define FLAC__NO_DLL
#include "AudioDecoder.h"
#include <map>
namespace nqr
{
//@todo expose this in API
inline std::map<int, std::string> GetQualityTable()
{
return {
{ 0, "0 (Fastest)" },
{ 1, "1" },
{ 2, "2" },
{ 3, "3" },
{ 4, "4" },
{ 5, "5 (Default)" },
{ 6, "6" },
{ 7, "7" },
{ 8, "8 (Highest)" },
};
}
struct FlacDecoder final : public nqr::BaseDecoder
{
FlacDecoder() = default;
virtual ~FlacDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,141 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef IMA4_UTIL_H
#define IMA4_UTIL_H
#include "AudioDecoder.h"
namespace nqr
{
struct ADPCMState
{
int frame_size;
int firstDataBlockByte;
int dataSize;
int currentByte;
const uint8_t * inBuffer;
};
static const int ima_index_table[16] =
{
-1, -1, -1, -1, // +0 / +3 : - the step
2, 4, 6, 8, // +4 / +7 : + the step
-1, -1, -1, -1, // -0 / -3 : - the step
2, 4, 6, 8, // -4 / -7 : + the step
};
static inline int ima_clamp_index(int index)
{
if (index < 0) return 0;
else if (index > 88) return 88;
return index;
}
static inline int16_t ima_clamp_predict(int16_t predict)
{
if (predict < -32768) return -32768;
else if (predict > 32767) return 32767;
return predict;
}
static const int ima_step_table[89] =
{
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34,
37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494,
544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552,
1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026,
4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623,
27086, 29794, 32767
};
// Decodes an IMA ADPCM nibble to a 16 bit pcm sample
static inline int16_t decode_nibble(uint8_t nibble, int16_t & p, int & s)
{
// Compute a delta to add to the predictor value
int diff = ima_step_table[s] >> 3;
if (nibble & 4) diff += ima_step_table[s];
if (nibble & 2) diff += ima_step_table[s] >> 1;
if (nibble & 1) diff += ima_step_table[s] >> 2;
// Sign
if (nibble & 8) diff = -diff;
// Add delta
p += diff;
s += ima_index_table[nibble];
s = ima_clamp_index(s);
return ima_clamp_predict(p);
}
void decode_ima_adpcm(ADPCMState & state, int16_t * outBuffer, uint32_t num_channels)
{
const uint8_t * data = state.inBuffer;
// Loop over the interleaved channels
for (uint32_t ch = 0; ch < num_channels; ch++)
{
const int byteOffset = ch * 4;
// Header Structure:
// Byte0: packed low byte of the initial predictor
// Byte1: packed high byte of the initial predictor
// Byte2: initial step index
// Byte3: Reserved empty value
int16_t predictor = ((int16_t)data[byteOffset + 1] << 8) | data[byteOffset];
int stepIndex = data[byteOffset + 2];
uint8_t reserved = data[byteOffset + 3];
if (reserved != 0) throw std::runtime_error("adpcm decode error");
int byteIdx = num_channels * 4 + byteOffset; //the byte index of the first data word for this channel
int idx = ch;
// Decode nibbles of the remaining data
while (byteIdx < state.frame_size)
{
for (int j = 0; j < 4; j++)
{
outBuffer[idx] = decode_nibble(data[byteIdx] & 0xf, predictor, stepIndex); // low nibble
idx += num_channels;
outBuffer[idx] = decode_nibble(data[byteIdx] >> 4, predictor, stepIndex); // high nibble
idx += num_channels;
byteIdx++;
}
byteIdx += (num_channels - 1) << 2; // Jump to the next data word for the current channel
}
}
}
} // end namespace nqr
#endif

@ -0,0 +1,45 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef MODPLUG_DECODER_H
#define MODPLUG_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
struct ModplugDecoder : public nqr::BaseDecoder
{
ModplugDecoder() {};
virtual ~ModplugDecoder() {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override;
virtual std::vector<std::string> GetSupportedFileExtensions() override;
};
} // end namespace nqr
#endif

@ -0,0 +1,45 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef MUSEPACK_DECODER_H
#define MUSEPACK_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
struct MusepackDecoder final : public nqr::BaseDecoder
{
MusepackDecoder() = default;
virtual ~MusepackDecoder() override {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,50 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef OPUS_DECODER_H
#define OPUS_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
// Opus is a general-purpose codec designed to replace Vorbis at some point. Primarily, it's a low
// delay format making it suitable for high-quality, real time streaming. It's not really
// an archival format or something designed for heavy DSP post-processing since
// it's fundamentally limited to encode/decode at 48khz.
// https://mf4.xiph.org/jenkins/view/opus/job/opusfile-unix/ws/doc/html/index.html
struct OpusDecoder final : public nqr::BaseDecoder
{
OpusDecoder() = default;
virtual ~OpusDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,104 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef POSTPROCESS_H
#define POSTPROCESS_H
#include <vector>
namespace nqr
{
template <typename T>
inline void DeinterleaveStereo(T * c1, T * c2, T const * src, size_t count)
{
auto src_end = src + count;
while (src != src_end)
{
*c1 = src[0];
*c2 = src[1];
c1++;
c2++;
src += 2;
}
}
template<typename T>
void InterleaveChannels(const T * src, T * dest, size_t numFramesPerChannel, size_t numChannels, size_t N)
{
for (size_t ch = 0; ch < numChannels; ch++)
{
size_t x = ch;
const T * srcChannel = &src[ch * numFramesPerChannel];
for(size_t i = 0; i < N; i++)
{
dest[x] = srcChannel[i];
x += numChannels;
}
}
}
template<typename T>
void DeinterleaveChannels(const T * src, T * dest, size_t numFramesPerChannel, size_t numChannels, size_t N)
{
for(size_t ch = 0; ch < numChannels; ch++)
{
size_t x = ch;
T *destChannel = &dest[ch * numFramesPerChannel];
for (size_t i = 0; i < N; i++)
{
destChannel[i] = (T) src[x];
x += numChannels;
}
}
}
template <typename T>
void StereoToMono(const T * src, T * dest, size_t N)
{
for (size_t i = 0, j = 0; i < N; i += 2, ++j)
{
dest[j] = (src[i] + src[i + 1]) / 2.0f;
}
}
template <typename T>
void MonoToStereo(const T * src, T * dest, size_t N)
{
for(size_t i = 0, j = 0; i < N; ++i, j += 2)
{
dest[j] = src[i];
dest[j + 1] = src[i];
}
}
inline void TrimSilenceInterleaved(std::vector<float> & buffer, float v, bool fromFront, bool fromEnd)
{
//@todo implement me!
}
} // end namespace nqr
#endif

@ -0,0 +1,84 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef RIFF_UTILS_H
#define RIFF_UTILS_H
#include "Common.h"
#include "WavDecoder.h"
#include "Dither.h"
namespace nqr
{
/////////////////////
// Chunk utilities //
/////////////////////
struct EncoderParams
{
int channelCount;
PCMFormat targetFormat;
DitherType dither;
};
struct ChunkHeaderInfo
{
uint32_t offset; // Byte offset into chunk
uint32_t size; // Size of the chunk in bytes
};
inline uint32_t GenerateChunkCode(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
{
#ifdef ARCH_CPU_LITTLE_ENDIAN
return ((uint32_t) ((a) | ((b) << 8) | ((c) << 16) | (((uint32_t) (d)) << 24)));
#else
return ((uint32_t) ((((uint32_t) (a)) << 24) | ((b) << 16) | ((c) << 8) | (d)));
#endif
}
inline char * GenerateChunkCodeChar(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
{
auto chunk = GenerateChunkCode(a, b, c, d);
char * outArr = new char[4];
uint32_t t = 0x000000FF;
for(size_t i = 0; i < 4; i++)
{
outArr[i] = chunk & t;
chunk >>= 8;
}
return outArr;
}
ChunkHeaderInfo ScanForChunk(const std::vector<uint8_t> & fileData, uint32_t chunkMarker);
WaveChunkHeader MakeWaveHeader(const EncoderParams param, const int sampleRate);
} // end namespace nqr
#endif

@ -0,0 +1,44 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef VORBIS_DECODER_H
#define VORBIS_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
struct VorbisDecoder final : public nqr::BaseDecoder
{
VorbisDecoder() = default;
virtual ~VorbisDecoder() override {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,182 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef WAVE_DECODER_H
#define WAVE_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
enum WaveFormatCode
{
FORMAT_UNKNOWN = 0x0, // Unknown Wave Format
FORMAT_PCM = 0x1, // PCM Format
FORMAT_ADPCM = 0x2, // Microsoft ADPCM Format
FORMAT_IEEE = 0x3, // IEEE float/double
FORMAT_ALAW = 0x6, // 8-bit ITU-T G.711 A-law
FORMAT_MULAW = 0x7, // 8-bit ITU-T G.711 µ-law
FORMAT_IMA_ADPCM = 0x11, // IMA ADPCM Format
FORMAT_EXT = 0xFFFE // Set via subformat
};
struct RiffChunkHeader
{
uint32_t id_riff; // Chunk ID: 'RIFF'
uint32_t file_size; // Entire file in bytes
uint32_t id_wave; // Chunk ID: 'WAVE'
};
struct WaveChunkHeader
{
uint32_t fmt_id; // Chunk ID: 'fmt '
uint32_t chunk_size; // Size in bytes
uint16_t format; // Format code
uint16_t channel_count; // Num interleaved channels
uint32_t sample_rate; // SR
uint32_t data_rate; // Data rate
uint16_t frame_size; // 1 frame = channels * bits per sample (also known as block align)
uint16_t bit_depth; // Bits per sample
};
struct BextChunk
{
uint32_t fmt_id; // Chunk ID: 'bext'
uint32_t chunk_size; // Size in bytes
uint8_t description[256]; // Description of the sound (ascii)
uint8_t origin[32]; // Name of the originator (ascii)
uint8_t origin_ref[32]; // Reference of the originator (ascii)
uint8_t orgin_date[10]; // yyyy-mm-dd (ascii)
uint8_t origin_time[8]; // hh-mm-ss (ascii)
uint64_t time_ref; // First sample count since midnight
uint32_t version; // Version of the BWF
uint8_t uimd[64]; // Byte 0 of SMPTE UMID
uint8_t reserved[188]; // 190 bytes, reserved for future use & set to NULL
};
struct FactChunk
{
uint32_t fact_id; // Chunk ID: 'fact'
uint32_t chunk_size; // Size in bytes
uint32_t sample_length; // number of samples per channel
};
struct ExtensibleData
{
uint16_t size;
uint16_t valid_bits_per_sample;
uint32_t channel_mask;
struct GUID
{
uint32_t data0;
uint16_t data1;
uint16_t data2;
uint16_t data3;
uint8_t data4[6];
};
};
template<class C, class R>
std::basic_ostream<C,R> & operator << (std::basic_ostream<C,R> & a, const WaveChunkHeader & b)
{
return a <<
"Format ID:\t\t" << b.fmt_id <<
"\nChunk Size:\t\t" << b.chunk_size <<
"\nFormat Code:\t\t" << b.format <<
"\nChannels:\t\t" << b.channel_count <<
"\nSample Rate:\t\t" << b.sample_rate <<
"\nData Rate:\t\t" << b.data_rate <<
"\nFrame Size:\t\t" << b.frame_size <<
"\nBit Depth:\t\t" << b.bit_depth << std::endl;
}
//@todo expose speaker/channel/layout masks in the API:
enum SpeakerChannelMask
{
SPEAKER_FRONT_LEFT = 0x00000001,
SPEAKER_FRONT_RIGHT = 0x00000002,
SPEAKER_FRONT_CENTER = 0x00000004,
SPEAKER_LOW_FREQUENCY = 0x00000008,
SPEAKER_BACK_LEFT = 0x00000010,
SPEAKER_BACK_RIGHT = 0x00000020,
SPEAKER_FRONT_LEFT_OF_CENTER = 0x00000040,
SPEAKER_FRONT_RIGHT_OF_CENTER = 0x00000080,
SPEAKER_BACK_CENTER = 0x00000100,
SPEAKER_SIDE_LEFT = 0x00000200,
SPEAKER_SIDE_RIGHT = 0x00000400,
SPEAKER_TOP_CENTER = 0x00000800,
SPEAKER_TOP_FRONT_LEFT = 0x00001000,
SPEAKER_TOP_FRONT_CENTER = 0x00002000,
SPEAKER_TOP_FRONT_RIGHT = 0x00004000,
SPEAKER_TOP_BACK_LEFT = 0x00008000,
SPEAKER_TOP_BACK_CENTER = 0x00010000,
SPEAKER_TOP_BACK_RIGHT = 0x00020000,
SPEAKER_RESERVED = 0x7FFC0000,
SPEAKER_ALL = 0x80000000
};
enum SpeakerLayoutMask
{
SPEAKER_MONO = (SPEAKER_FRONT_CENTER),
SPEAKER_STEREO = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT),
SPEAKER_2POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY),
SPEAKER_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER),
SPEAKER_QUAD = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_4POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_5POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT),
SPEAKER_7POINT1 = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER),
SPEAKER_5POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT),
SPEAKER_7POINT1_SURROUND = (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT),
};
//@todo verify mask values
inline int ComputeChannelMask(const size_t channels)
{
switch (channels)
{
case 1: return SPEAKER_MONO;
case 2: return SPEAKER_STEREO;
case 3: return SPEAKER_2POINT1;
case 4: return SPEAKER_QUAD;
case 5: return SPEAKER_4POINT1;
case 6: return SPEAKER_5POINT1;
default: return -1;
}
}
struct WavDecoder final : public nqr::BaseDecoder
{
WavDecoder() = default;
virtual ~WavDecoder() {}
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,22 +23,36 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef NYQUIST_ENCODERS_H
#define NYQUIST_ENCODERS_H
#ifndef WAVE_ENCODER_H
#define WAVE_ENCODER_H
#include "Common.h"
#include "WavDecoder.h"
#include "RiffUtils.h"
namespace nqr
{
// A simplistic encoder that takes a buffer of audio, conforms it to the user's
// EncoderParams preference, and writes to disk. Be warned, does not support resampling!
// @todo support dithering, samplerate conversion, etc.
int encode_wav_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
enum EncoderError
{
NoError,
InsufficientSampleData,
FileIOError,
UnsupportedSamplerate,
UnsupportedChannelConfiguration,
UnsupportedBitdepth,
UnsupportedChannelMix,
BufferTooBig,
};
// A simplistic encoder that takes a buffer of audio, conforms it to the user's
// EncoderParams preference, and writes to disk. Be warned, does not support resampling!
// @todo support dithering, samplerate conversion, etc.
int encode_wav_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
// Assume data adheres to EncoderParams, except for bit depth and fmt which are re-formatted
// to satisfy the Ogg/Opus spec.
int encode_opus_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
int encode_opus_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
} // end namespace nqr
#endif // end NYQUIST_ENCODERS_H
#endif

@ -0,0 +1,45 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef WAVEPACK_DECODER_H
#define WAVEPACK_DECODER_H
#include "AudioDecoder.h"
namespace nqr
{
struct WavPackDecoder final : public nqr::BaseDecoder
{
WavPackDecoder() = default;
virtual ~WavPackDecoder() override {};
virtual void LoadFromPath(nqr::AudioData * data, const std::string & path) override final;
virtual void LoadFromBuffer(nqr::AudioData * data, const std::vector<uint8_t> & memory) override final;
virtual std::vector<std::string> GetSupportedFileExtensions() override final;
};
} // end namespace nqr
#endif

@ -0,0 +1,137 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "AudioDecoder.h"
#include "WavDecoder.h"
#include "WavPackDecoder.h"
#include "FlacDecoder.h"
#include "VorbisDecoder.h"
#include "OpusDecoder.h"
#include "MusepackDecoder.h"
#include "ModplugDecoder.h"
using namespace nqr;
NyquistIO::NyquistIO()
{
BuildDecoderTable();
}
NyquistIO::~NyquistIO() { }
void NyquistIO::Load(AudioData * data, const std::string & path)
{
if (IsFileSupported(path))
{
if (decoderTable.size())
{
auto fileExtension = ParsePathForExtension(path);
auto decoder = GetDecoderForExtension(fileExtension);
try
{
decoder->LoadFromPath(data, path);
}
catch (const std::exception & e)
{
std::cerr << "Caught internal exception: " << e.what() << std::endl;
}
}
else throw std::runtime_error("No available decoders.");
}
else
{
throw UnsupportedExtensionEx();
}
}
void NyquistIO::Load(AudioData * data, const std::string & extension, const std::vector<uint8_t> & buffer)
{
if (decoderTable.find(extension) == decoderTable.end())
{
throw UnsupportedExtensionEx();
}
if (decoderTable.size())
{
auto decoder = GetDecoderForExtension(extension);
try
{
decoder->LoadFromBuffer(data, buffer);
}
catch (const std::exception & e)
{
std::cerr << "Caught internal exception: " << e.what() << std::endl;
}
}
else
{
throw std::runtime_error("No available decoders.");
}
}
bool NyquistIO::IsFileSupported(const std::string & path) const
{
auto fileExtension = ParsePathForExtension(path);
if (decoderTable.find(fileExtension) == decoderTable.end()) return false;
else return true;
}
std::string NyquistIO::ParsePathForExtension(const std::string & path) const
{
if (path.find_last_of(".") != std::string::npos) return path.substr(path.find_last_of(".") + 1);
return std::string("");
}
std::shared_ptr<BaseDecoder> NyquistIO::GetDecoderForExtension(const std::string & ext)
{
if (decoderTable.size()) return decoderTable[ext];
else throw std::runtime_error("No available decoders.");
return nullptr;
}
void NyquistIO::AddDecoderToTable(std::shared_ptr<nqr::BaseDecoder> decoder)
{
auto supportedExtensions = decoder->GetSupportedFileExtensions();
for (const auto ext : supportedExtensions)
{
if (decoderTable.count(ext) >= 1) throw std::runtime_error("decoder already exists for extension.");
decoderTable.insert(DecoderPair(ext, decoder));
}
}
void NyquistIO::BuildDecoderTable()
{
AddDecoderToTable(std::make_shared<WavDecoder>());
AddDecoderToTable(std::make_shared<WavPackDecoder>());
AddDecoderToTable(std::make_shared<FlacDecoder>());
AddDecoderToTable(std::make_shared<VorbisDecoder>());
AddDecoderToTable(std::make_shared<OpusDecoder>());
AddDecoderToTable(std::make_shared<MusepackDecoder>());
AddDecoderToTable(std::make_shared<ModplugDecoder>());
}

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -24,189 +24,34 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Common.h"
#include "Decoders.h"
#include <cstring>
#include <unordered_map>
using namespace nqr;
NyquistIO::NyquistIO() { BuildDecoderTable(); }
NyquistIO::~NyquistIO() { }
void NyquistIO::Load(AudioData * data, const std::string & path)
{
if (IsFileSupported(path))
{
if (decoderTable.size())
{
auto fileExtension = ParsePathForExtension(path);
auto decoder = GetDecoderForExtension(fileExtension);
try
{
decoder->LoadFromPath(data, path);
}
catch (const std::exception & e)
{
std::cerr << "NyquistIO::Load(" << path << ") caught internal exception: " << e.what() << std::endl;
throw;
}
}
else throw std::runtime_error("No available decoders.");
}
else
{
throw UnsupportedExtensionEx();
}
}
void NyquistIO::Load(AudioData * data, const std::vector<uint8_t> & buffer)
{
const std::map<std::vector<int16_t>, std::string> magic_map{
{{ 'w', 'v', 'p', 'k' }, "wv" },
{{ 'M', 'P', 'C', 'K' }, "mpc" },
{{ 0xFF, 0xFB }, "mp3" }, // ÿû, mp3 without ID3 header
{{ 'I', 'D', '3' }, "mp3" }, // mp3 with ID3 header
{{ 'O', 'g', 'g', 'S' }, "ogg_or_vorbis" }, // see `match_ogg_subtype`
{{ 'f', 'L', 'a', 'C' }, "flac" },
{{ 0x52, 0x49, 0x46, 0x46, -0x1, -0x1, -0x1, -0x1, 0x57, 0x41, 0x56, 0x45 }, "wav" } // RIFF....WAVE
};
auto match_magic = [](const uint8_t * data, const std::vector<int16_t> & magic)
{
for (int i = 0; i < magic.size(); ++i)
{
if (magic[i] != data[i] && magic[i] != -0x1) // -0x1 skips things that don't contribute to the magic number
{
return false;
}
}
return true;
};
auto match_ogg_subtype = [](const uint8_t * data)
{
std::string header; // arbitrarily read the first 64 bytes as ascii characters
for (int i = 0; i < 64; ++i) header += data[i];
std::size_t found_opus = header.find("OpusHead");
if (found_opus != std::string::npos) return "opus";
std::size_t found_vorbis = header.find("vorbis");
if (found_vorbis != std::string::npos) return "ogg";
return "none";
};
std::string ext = "none";
for (auto & filetype : magic_map)
{
if (match_magic(buffer.data(), filetype.first))
{
ext = filetype.second;
}
if (ext == "ogg_or_vorbis")
{
ext = match_ogg_subtype(buffer.data());
}
}
NyquistIO::Load(data, ext, buffer);
}
void NyquistIO::Load(AudioData * data, const std::string & extension, const std::vector<uint8_t> & buffer)
{
if (decoderTable.find(extension) == decoderTable.end())
{
throw UnsupportedExtensionEx();
}
if (decoderTable.size())
{
auto decoder = GetDecoderForExtension(extension);
try
{
decoder->LoadFromBuffer(data, buffer);
}
catch (const std::exception & e)
{
std::cerr << "caught internal loading exception: " << e.what() << std::endl;
throw;
}
}
else throw std::runtime_error("fatal: no decoders available");
}
bool NyquistIO::IsFileSupported(const std::string & path) const
{
auto fileExtension = ParsePathForExtension(path);
if (decoderTable.find(fileExtension) == decoderTable.end()) return false;
else return true;
}
std::string NyquistIO::ParsePathForExtension(const std::string & path) const
{
if (path.find_last_of(".") != std::string::npos) return path.substr(path.find_last_of(".") + 1);
return std::string("");
}
std::shared_ptr<BaseDecoder> NyquistIO::GetDecoderForExtension(const std::string & ext)
{
if (decoderTable.size()) return decoderTable[ext];
else throw std::runtime_error("No available decoders.");
return nullptr;
}
void NyquistIO::AddDecoderToTable(std::shared_ptr<nqr::BaseDecoder> decoder)
{
auto supportedExtensions = decoder->GetSupportedFileExtensions();
for (const auto ext : supportedExtensions)
{
if (decoderTable.count(ext) >= 1) throw std::runtime_error("decoder already exists for extension");
decoderTable.insert(DecoderPair(ext, decoder));
}
}
void NyquistIO::BuildDecoderTable()
{
AddDecoderToTable(std::make_shared<WavDecoder>());
AddDecoderToTable(std::make_shared<WavPackDecoder>());
AddDecoderToTable(std::make_shared<FlacDecoder>());
AddDecoderToTable(std::make_shared<VorbisDecoder>());
AddDecoderToTable(std::make_shared<OpusDecoder>());
AddDecoderToTable(std::make_shared<MusepackDecoder>());
AddDecoderToTable(std::make_shared<Mp3Decoder>());
}
NyquistFileBuffer nqr::ReadFile(const std::string & pathToFile)
{
//std::cout << "[Debug] Open: " << pathToFile << std::endl;
FILE * audioFile = fopen(pathToFile.c_str(), "rb");
if (!audioFile)
{
throw std::runtime_error("file not found");
}
fseek(audioFile, 0, SEEK_END);
size_t lengthInBytes = ftell(audioFile);
fseek(audioFile, 0, SEEK_SET);
// Allocate temporary buffer
std::vector<uint8_t> fileBuffer(lengthInBytes);
size_t elementsRead = fread(fileBuffer.data(), 1, lengthInBytes, audioFile);
if (elementsRead == 0 || fileBuffer.size() < 64)
{
throw std::runtime_error("error reading file or file too small");
}
NyquistFileBuffer data = {std::move(fileBuffer), elementsRead};
fclose(audioFile);
@ -219,7 +64,7 @@ NyquistFileBuffer nqr::ReadFile(const std::string & pathToFile)
void nqr::ConvertToFloat32(float * dst, const uint8_t * src, const size_t N, PCMFormat f)
{
assert(f != PCM_END);
if (f == PCM_U8)
{
const uint8_t * dataPtr = reinterpret_cast<const uint8_t *>(src);
@ -255,12 +100,12 @@ void nqr::ConvertToFloat32(float * dst, const uint8_t * src, const size_t N, PCM
for (size_t i = 0; i < N; ++i)
dst[i] = int32_to_float32(Read32(dataPtr[i]));
}
//@todo add int64 format
else if (f == PCM_FLT)
{
std::memcpy(dst, src, N * sizeof(float));
memcpy(dst, src, N * sizeof(float));
/* const float * dataPtr = reinterpret_cast<const float *>(src);
for (size_t i = 0; i < N; ++i)
dst[i] = (float) Read32(dataPtr[i]); */
@ -277,7 +122,7 @@ void nqr::ConvertToFloat32(float * dst, const uint8_t * src, const size_t N, PCM
void nqr::ConvertToFloat32(float * dst, const int32_t * src, const size_t N, PCMFormat f)
{
assert(f != PCM_END);
if (f == PCM_16)
{
for (size_t i = 0; i < N; ++i)
@ -316,7 +161,7 @@ void nqr::ConvertFromFloat32(uint8_t * dst, const float * src, const size_t N, P
assert(f != PCM_END);
Dither dither(t);
if (f == PCM_U8)
{
uint8_t * destPtr = reinterpret_cast<uint8_t *>(dst);

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,24 +23,22 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
// http://lists.xiph.org/pipermail/flac-dev/2012-March/003276.html
#define FLAC__NO_DLL
#include "FlacDecoder.h"
#include "FLAC/all.h"
#include "FLAC/stream_decoder.h"
#include "AudioDecoder.h"
#include <cstring>
using namespace nqr;
// FLAC is a big-endian format. All values are unsigned.
class FlacDecoderInternal
{
public:
FlacDecoderInternal(AudioData * d, const std::string & filepath) : d(d)
{
decoderInternal = FLAC__stream_decoder_new();

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -79,7 +79,7 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include "FLAC/all.h"
#if defined(_WIN32) || defined(_WIN64)
#if defined(_MSC_VER)
#include "FLAC/src/win_utf8_io.c"
#endif

@ -0,0 +1,130 @@
/*
Copyright (c) 2016, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "ModplugDecoder.h"
using namespace nqr;
#ifndef MODPLUG_STATIC
#define MODPLUG_STATIC
#endif
#include "libmodplug/src/modplug.h"
class ModplugInternal
{
public:
ModplugInternal(AudioData * d, const std::vector<uint8_t> & fileData) : d(d)
{
ModPlug_Settings mps;
ModPlug_GetSettings(&mps);
mps.mChannels = 2;
mps.mBits = 32;
mps.mFrequency = 44100;
mps.mResamplingMode = MODPLUG_RESAMPLE_FIR; //_LINEAR, _SPLINE
mps.mStereoSeparation = 128;
mps.mMaxMixChannels = 64;
mps.mLoopCount = 0; // forever: -1;
mps.mFlags = MODPLUG_ENABLE_OVERSAMPLING; // _NOISE_REDUCTION, _REVERB, _MEGABASS, _SURROUND
ModPlug_SetSettings(&mps);
mpf = ModPlug_Load((const void*)fileData.data(), fileData.size());
if (!mpf) throw std::runtime_error("could not load module");
d->sampleRate = 44100;
d->channelCount = 2;
d->sourceFormat = MakeFormatForBits(32, true, false);
auto len_ms = ModPlug_GetLength(mpf);
d->lengthSeconds = (double) len_ms / 1000.0;
auto totalSamples = (44100LL * len_ms) / 1000;
d->samples.resize(totalSamples * d->channelCount);
auto read_func = [&]()
{
const float invf = 1 / (float)0x7fffffff;
float *ptr = d->samples.data();
float *end = d->samples.data() + d->samples.size();
while (ptr < end)
{
int res = ModPlug_Read(mpf, (void*)ptr, (end - ptr) * sizeof(float));
int samples_read = res / (sizeof(float) * 2);
if (totalSamples < samples_read)
{
samples_read = totalSamples;
}
for (int i = 0; i < samples_read; ++i)
{
*ptr++ = *((int*)ptr) * invf;
*ptr++ = *((int*)ptr) * invf;
}
totalSamples -= samples_read;
}
return ptr >= end;
};
if (!read_func())
{
throw std::runtime_error("could not read any data");
}
ModPlug_Unload(mpf);
}
private:
ModPlugFile * mpf;
NO_MOVE(ModplugInternal);
AudioData * d;
};
//////////////////////
// Public Interface //
//////////////////////
void ModplugDecoder::LoadFromPath(AudioData * data, const std::string & path)
{
auto fileBuffer = nqr::ReadFile(path);
ModplugInternal decoder(data, fileBuffer.buffer);
}
void ModplugDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & memory)
{
ModplugInternal decoder(data, memory);
}
std::vector<std::string> ModplugDecoder::GetSupportedFileExtensions()
{
return {"pat","mid", "mod","s3m","xm","it","669","amf","ams","dbm","dmf","dsm","far","mdl","med","mtm","okt","ptm","stm","ult","umx","mt2","psm"};
}

@ -0,0 +1,102 @@
/*
Copyright (c) 2016, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#if (_MSC_VER)
#pragma warning (push)
#pragma warning (disable: 181 111 4267 4996 4244 4701 4702 4133 4100 4127 4206 4312 4505 4365 4005 4013 4334)
#endif
#ifdef __clang__
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wconversion"
#pragma clang diagnostic ignored "-Wshadow"
#pragma clang diagnostic ignored "-Wdeprecated-register"
#define HAVE_SETENV
#endif
#ifndef MODPLUG_STATIC
#define MODPLUG_STATIC
#endif
#define HAVE_SINF
#include "libmodplug/src/load_669.cpp"
#include "libmodplug/src/load_abc.cpp"
#include "libmodplug/src/load_amf.cpp"
#include "libmodplug/src/load_ams.cpp"
#include "libmodplug/src/load_dbm.cpp"
#include "libmodplug/src/load_dmf.cpp"
#include "libmodplug/src/load_dsm.cpp"
#include "libmodplug/src/load_far.cpp"
#include "libmodplug/src/load_it.cpp"
#include "libmodplug/src/load_j2b.cpp"
#include "libmodplug/src/load_mdl.cpp"
#include "libmodplug/src/load_med.cpp"
#define none none_alt
#define MMFILE MMFILE_alt
#define mmfseek mmfseek_alt
#define mmftell mmftell_alt
#define mmreadUBYTES mmreadUBYTES_alt
#include "libmodplug/src/load_mid.cpp"
#include "libmodplug/src/load_mod.cpp"
#include "libmodplug/src/load_mt2.cpp"
#include "libmodplug/src/load_mtm.cpp"
#include "libmodplug/src/load_okt.cpp"
#undef MMFILE
#undef mmfseek
#undef mmftell
#undef mmreadUBYTES
#define MMFILE MMFILE_alt2
#define mmfseek mmfseek_alt2
#define mmftell mmftell_alt2
#define mmreadUBYTES mmreadUBYTES_alt2
#include "libmodplug/src/load_pat.cpp"
#include "libmodplug/src/load_psm.cpp"
#include "libmodplug/src/load_ptm.cpp"
#include "libmodplug/src/load_s3m.cpp"
#include "libmodplug/src/load_stm.cpp"
#include "libmodplug/src/load_ult.cpp"
#include "libmodplug/src/load_umx.cpp"
#include "libmodplug/src/load_wav.cpp"
#include "libmodplug/src/load_xm.cpp"
#include "libmodplug/src/mmcmp.cpp"
#include "libmodplug/src/modplug.cpp"
#include "libmodplug/src/sndfile.cpp"
#include "libmodplug/src/sndmix.cpp"
#include "libmodplug/src/fastmix.cpp"
#include "libmodplug/src/snd_dsp.cpp"
#include "libmodplug/src/snd_flt.cpp"
#include "libmodplug/src/snd_fx.cpp"
#ifdef __clang__
#pragma clang diagnostic pop
#endif
#if (_MSC_VER)
#pragma warning (pop)
#endif

@ -1,80 +0,0 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
using namespace nqr;
#include "mpc/mpcdec.h"
#include "mpc/reader.h"
#include "musepack/libmpcdec/decoder.h"
#include "musepack/libmpcdec/internal.h"
#define MINIMP3_FLOAT_OUTPUT
#define MINIMP3_IMPLEMENTATION
#include "minimp3/minimp3.h"
#include "minimp3/minimp3_ex.h"
#include <cstdlib>
#include <cstring>
void mp3_decode_internal(AudioData * d, const std::vector<uint8_t> & fileData)
{
mp3dec_t mp3d;
mp3dec_file_info_t info;
mp3dec_load_buf(&mp3d, (const uint8_t*)fileData.data(), fileData.size(), &info, 0, 0);
d->sampleRate = info.hz;
d->channelCount = info.channels;
d->sourceFormat = MakeFormatForBits(32, true, false);
d->lengthSeconds = ((float)info.samples / (float)d->channelCount) / (float)d->sampleRate;
if (info.samples == 0) throw std::runtime_error("mp3: could not read any data");
d->samples.resize(info.samples);
std::memcpy(d->samples.data(), info.buffer, sizeof(float) * info.samples);
std::free(info.buffer);
}
//////////////////////
// Public Interface //
//////////////////////
void Mp3Decoder::LoadFromPath(AudioData * data, const std::string & path)
{
auto fileBuffer = nqr::ReadFile(path);
mp3_decode_internal(data, fileBuffer.buffer);
}
void Mp3Decoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & memory)
{
mp3_decode_internal(data, memory);
}
std::vector<std::string> Mp3Decoder::GetSupportedFileExtensions()
{
return {"mp3"};
}

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,7 +23,7 @@
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
#include "MusepackDecoder.h"
using namespace nqr;

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,19 +23,13 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
#include "OpusDecoder.h"
#include "opus/opusfile/include/opusfile.h"
using namespace nqr;
static const int OPUS_SAMPLE_RATE = 48000;
// Opus is a general-purpose codec designed to replace Vorbis at some point. Primarily, it's a low
// delay format making it suitable for high-quality, real time streaming. It's not really
// an archival format or something designed for heavy DSP post-processing since
// it's fundamentally limited to encode/decode at 48khz.
// https://mf4.xiph.org/jenkins/view/opus/job/opusfile-unix/ws/doc/html/index.html
class OpusDecoderInternal
{

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:

@ -0,0 +1,66 @@
/*
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "RiffUtils.h"
using namespace nqr;
ChunkHeaderInfo nqr::ScanForChunk(const std::vector<uint8_t> & fileData, uint32_t chunkMarker)
{
// D[n] aligned to 16 bytes now
const uint16_t * d = reinterpret_cast<const uint16_t *>(fileData.data());
for (size_t i = 0; i < fileData.size() / sizeof(uint16_t); i++)
{
// This will be in machine endianess
uint32_t m = Pack(Read16(d[i]), Read16(d[i+1]));
if (m == chunkMarker)
{
uint32_t cSz = Pack(Read16(d[i+2]), Read16(d[i+3]));
return {(uint32_t (i * sizeof(uint16_t))), cSz}; // return i in bytes to the start of the data
}
else continue;
}
return {0, 0};
};
WaveChunkHeader nqr::MakeWaveHeader(const EncoderParams param, const int sampleRate)
{
WaveChunkHeader header;
int bitdepth = GetFormatBitsPerSample(param.targetFormat);
header.fmt_id = GenerateChunkCode('f', 'm', 't', ' ');
header.chunk_size = 16;
header.format = (param.targetFormat <= PCMFormat::PCM_32) ? WaveFormatCode::FORMAT_PCM : WaveFormatCode::FORMAT_IEEE;
header.channel_count = param.channelCount;
header.sample_rate = sampleRate;
header.data_rate = sampleRate * param.channelCount * (bitdepth / 8);
header.frame_size = param.channelCount * (bitdepth / 8);
header.bit_depth = bitdepth;
return header;
}

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,11 +23,9 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
#include "VorbisDecoder.h"
#include "libvorbis/include/vorbis/vorbisfile.h"
#include <string.h>
using namespace nqr;
class VorbisDecoderInternal

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,116 +23,13 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
#include "WavDecoder.h"
#include "RiffUtils.h"
#include "IMA4Util.h"
#include <cstring>
using namespace nqr;
struct ADPCMState
{
int frame_size;
int firstDataBlockByte;
int dataSize;
int currentByte;
const uint8_t * inBuffer;
};
static const int ima_index_table[16] =
{
-1, -1, -1, -1, // +0 / +3 : - the step
2, 4, 6, 8, // +4 / +7 : + the step
-1, -1, -1, -1, // -0 / -3 : - the step
2, 4, 6, 8, // -4 / -7 : + the step
};
static inline int ima_clamp_index(int index)
{
if (index < 0) return 0;
else if (index > 88) return 88;
return index;
}
static inline int16_t ima_clamp_predict(int16_t predict)
{
if (predict < -32768) return -32768;
else if (predict > 32767) return 32767;
return predict;
}
static const int ima_step_table[89] =
{
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34,
37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494,
544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552,
1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026,
4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623,
27086, 29794, 32767
};
// Decodes an IMA ADPCM nibble to a 16 bit pcm sample
static inline int16_t decode_nibble(uint8_t nibble, int16_t & p, int & s)
{
// Compute a delta to add to the predictor value
int diff = ima_step_table[s] >> 3;
if (nibble & 4) diff += ima_step_table[s];
if (nibble & 2) diff += ima_step_table[s] >> 1;
if (nibble & 1) diff += ima_step_table[s] >> 2;
// Sign
if (nibble & 8) diff = -diff;
// Add delta
p += diff;
s += ima_index_table[nibble];
s = ima_clamp_index(s);
return ima_clamp_predict(p);
}
void decode_ima_adpcm(ADPCMState & state, int16_t * outBuffer, uint32_t num_channels)
{
const uint8_t * data = state.inBuffer;
// Loop over the interleaved channels
for (uint32_t ch = 0; ch < num_channels; ch++)
{
const int byteOffset = ch * 4;
// Header Structure:
// Byte0: packed low byte of the initial predictor
// Byte1: packed high byte of the initial predictor
// Byte2: initial step index
// Byte3: Reserved empty value
int16_t predictor = ((int16_t)data[byteOffset + 1] << 8) | data[byteOffset];
int stepIndex = data[byteOffset + 2];
uint8_t reserved = data[byteOffset + 3];
if (reserved != 0) throw std::runtime_error("adpcm decode error");
int byteIdx = num_channels * 4 + byteOffset; //the byte index of the first data word for this channel
int idx = ch;
// Decode nibbles of the remaining data
while (byteIdx < state.frame_size)
{
for (int j = 0; j < 4; j++)
{
outBuffer[idx] = decode_nibble(data[byteIdx] & 0xf, predictor, stepIndex); // low nibble
idx += num_channels;
outBuffer[idx] = decode_nibble(data[byteIdx] >> 4, predictor, stepIndex); // high nibble
idx += num_channels;
byteIdx++;
}
byteIdx += (num_channels - 1) << 2; // Jump to the next data word for the current channel
}
}
}
//////////////////////
// Public Interface //
//////////////////////
@ -220,7 +117,10 @@ void WavDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & m
bool grabExtensibleData = false;
bool adpcmEncoded = false;
if (wavHeader.format == WaveFormatCode::FORMAT_IEEE)
if (wavHeader.format == WaveFormatCode::FORMAT_PCM)
{
}
else if (wavHeader.format == WaveFormatCode::FORMAT_IEEE)
{
scanForFact = true;
}
@ -231,7 +131,8 @@ void WavDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & m
}
else if (wavHeader.format == WaveFormatCode::FORMAT_EXT)
{
// Used when (1) PCM data has more than 16 bits; (2) channels > 2; (3) bits/sample !== container size; (4) channel/speaker mapping specified;
// Used when (1) PCM data has more than 16 bits; (2) channels > 2; (3) bits/sample !== container size; (4) channel/speaker mapping specified
//std::cout << "[format id] extended" << std::endl;
scanForFact = true;
grabExtensibleData = true;
}

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,7 +23,7 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Encoders.h"
#include "WavEncoder.h"
#include <fstream>
using namespace nqr;
@ -53,8 +53,7 @@ static inline void to_bytes(uint32_t value, char * arr)
int nqr::encode_wav_to_disk(const EncoderParams p, const AudioData * d, const std::string & path)
{
if (!d->samples.size())
return EncoderError::InsufficientSampleData;
assert(d->samples.size() > 0);
// Cast away const because we know what we are doing (Hopefully?)
float * sampleData = const_cast<float *>(d->samples.data());

@ -1,5 +1,5 @@
/*
Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
Copyright (c) 2015, Dimitri Diakopoulos All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
@ -23,10 +23,9 @@ OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Decoders.h"
#include "WavPackDecoder.h"
#include "wavpack.h"
#include <string.h>
#include <cstring>
using namespace nqr;
@ -41,30 +40,95 @@ public:
context = WavpackOpenFileInput(path.c_str(), errorStr, OPEN_WVC | OPEN_NORMALIZE, 0);
if (!context) throw std::runtime_error("Not a WavPack file");
auto bitdepth = WavpackGetBitsPerSample(context);
d->sampleRate = WavpackGetSampleRate(context);
d->channelCount = WavpackGetNumChannels(context);
d->lengthSeconds = double(getLengthInSeconds());
d->frameSize = d->channelCount * bitdepth;
//@todo support channel masks
// WavpackGetChannelMask
auto totalSamples = size_t(getTotalSamples());
int mode = WavpackGetMode(context);
bool isFloatingPoint = (MODE_FLOAT & mode);
d->sourceFormat = MakeFormatForBits(bitdepth, isFloatingPoint, false);
auto totalSamples = size_t(WavpackGetNumSamples(context));
decode(totalSamples);
/// From the WavPack docs:
/// "... required memory at "buffer" is 4 * samples * num_channels bytes. The
/// audio data is returned right-justified in 32-bit longs in the endian
/// mode native to the executing processor."
d->samples.resize(totalSamples * d->channelCount);
if (!isFloatingPoint)
internalBuffer.resize(totalSamples * d->channelCount);
if (!readInternal(totalSamples))
throw std::runtime_error("could not read any data");
// Next, process internal buffer into the user-visible samples array
if (!isFloatingPoint)
ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples * d->channelCount, d->sourceFormat);
}
WavPackInternal(AudioData * d, const std::vector<uint8_t> & memory) : d(d)
WavPackInternal(AudioData * d, const std::vector<uint8_t> & memory) : d(d), data(std::move(memory)), dataPos(0)
{
WavpackStreamReader64 reader =
{
read_bytes,
write_bytes,
get_pos,
set_pos_abs,
set_pos_rel,
push_back_byte,
get_length,
can_seek,
truncate_here,
close,
};
char errorStr[128];
context = WavpackOpenRawDecoder((void *) memory.data(), memory.size(), nullptr, 0, 0, errorStr, OPEN_WVC | OPEN_NORMALIZE, 0);
// Since we are using OpenRawDecoder, WavpackGetNumSamples won't work.
// Instead, find the first block and get totalSamples from its header.
WavpackHeader wph;
auto headerOffset = readNextHeader(memory, &wph, 0);
if (!context || headerOffset == -1)
context = WavpackOpenFileInputEx64(&reader, this, nullptr, errorStr, OPEN_WVC | OPEN_NORMALIZE, 0);
if (!context)
{
throw std::runtime_error("Not a WavPack file");
}
auto bitdepth = WavpackGetBitsPerSample(context);
d->sampleRate = WavpackGetSampleRate(context);
d->channelCount = WavpackGetNumChannels(context);
d->lengthSeconds = double(getLengthInSeconds());
d->frameSize = d->channelCount * bitdepth;
//@todo support channel masks
// WavpackGetChannelMask
auto totalSamples = size_t(getTotalSamples());
int mode = WavpackGetMode(context);
bool isFloatingPoint = (MODE_FLOAT & mode);
d->sourceFormat = MakeFormatForBits(bitdepth, isFloatingPoint, false);
auto totalSamples = wph.total_samples;
decode(totalSamples);
d->samples.resize(totalSamples * d->channelCount);
if (!isFloatingPoint)
internalBuffer.resize(totalSamples * d->channelCount);
if (!readInternal(totalSamples))
throw std::runtime_error("could not read any data");
// Next, process internal buffer into the user-visible samples array
if (!isFloatingPoint)
ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples * d->channelCount, d->sourceFormat);
}
~WavPackInternal()
@ -98,68 +162,124 @@ public:
// EOF
//if (framesRead == 0) break;
totalFramesRead += framesRead;
framesRemaining -= framesRead;
}
return totalFramesRead;
}
private:
void decode(size_t totalSamples) {
auto bitdepth = WavpackGetBitsPerSample(context);
d->sampleRate = WavpackGetSampleRate(context);
d->channelCount = WavpackGetNumChannels(context);
d->lengthSeconds = double(totalSamples / WavpackGetSampleRate(context));
d->frameSize = d->channelCount * bitdepth;
//@todo support channel masks
// WavpackGetChannelMask
int mode = WavpackGetMode(context);
bool isFloatingPoint = (MODE_FLOAT & mode);
d->sourceFormat = MakeFormatForBits(bitdepth, isFloatingPoint, false);
d->samples.resize(totalSamples * d->channelCount);
if (!isFloatingPoint)
internalBuffer.resize(totalSamples * d->channelCount);
if (!readInternal(totalSamples))
throw std::runtime_error("could not read any data");
// Next, process internal buffer into the user-visible samples array
if (!isFloatingPoint)
ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples * d->channelCount, d->sourceFormat);
static int32_t read_bytes(void * id, void * data, int32_t byte_count)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
int32_t readLength = std::min<size_t>(byte_count, decoder->data.size() - decoder->dataPos);
if (readLength > 0)
{
std::memcpy(data, decoder->data.data(), readLength);
decoder->dataPos += readLength;
return readLength;
}
else return 0;
}
return 0;
}
int64_t readNextHeader(const std::vector<uint8_t> & memory, WavpackHeader *wphdr, size_t startOffset) {
/// Based on read_next_header function in wavpack's openutils.c.
/// This will find the position of the next WavPack header in the given vector, at or after startOffset.
/// If a header is found, it will write the header to *wphdr and return the position of its first byte in the vector.
/// Otherwise, it will return -1.
unsigned char* sp;
for (size_t i = startOffset; i < memory.size(); i++) {
sp = const_cast<unsigned char *>(memory.data() + i);
auto headerStartPoint = sp;
if (*sp++ == 'w' && *sp == 'v' && *++sp == 'p' && *++sp == 'k' &&
!(*++sp & 1) && sp [2] < 16 && !sp [3] && (sp [2] || sp [1] || *sp >= 24) && sp [5] == 4 &&
sp [4] >= (MIN_STREAM_VERS & 0xff) && sp [4] <= (MAX_STREAM_VERS & 0xff) && sp [18] < 3 && !sp [19]) {
memcpy (wphdr, headerStartPoint, sizeof (*wphdr));
WavpackLittleEndianToNative (wphdr, (char*)WavpackHeaderFormat);
return i;
}
static int32_t write_bytes(void * id, void * data, int32_t byte_count)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
int32_t writeLength = std::min<size_t>(byte_count, decoder->data.size() - decoder->dataPos);
if (writeLength > 0)
{
std::memcpy(decoder->data.data(), data, writeLength);
decoder->dataPos += writeLength;
return writeLength;
}
else return 0;
}
return 0;
}
static int64_t get_pos(void *id)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
return decoder->dataPos;
}
return -1;
return 0;
}
static int set_pos_abs(void *id, int64_t pos)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
size_t newPos = std::min<size_t>(pos, decoder->data.size());
decoder->dataPos = newPos;
return newPos;
}
return 0;
}
static int set_pos_rel(void *id, int64_t delta, int mode)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
size_t newPos = 0;
if (mode == SEEK_SET) newPos = delta;
else if (mode == SEEK_CUR) newPos = decoder->dataPos + delta;
else if (mode == SEEK_END) newPos = decoder->data.size() + delta;
newPos = std::min<size_t>(newPos, decoder->data.size());
decoder->dataPos = newPos;
return newPos;
}
return 0;
}
static int push_back_byte(void *id, int c)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
decoder->dataPos--;
decoder->data[decoder->dataPos] = c;
return 1;
}
return 0;
}
static int64_t get_length(void *id)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
return decoder->data.size();
}
return 0;
}
static int can_seek(void *id)
{
if (id != nullptr) return 1;
return 0;
}
static int truncate_here(void *id)
{
if (id != nullptr)
{
WavPackInternal *decoder = (WavPackInternal *)id;
decoder->data.resize(decoder->dataPos);
return 1;
}
return 0;
}
static int close(void *id)
{
if (id != nullptr) return 1;
return 0;
}
private:
NO_MOVE(WavPackInternal);
@ -168,7 +288,9 @@ private:
WavpackContext * context; //@todo unique_ptr
AudioData * d;
std::vector<uint8_t> data;
size_t dataPos;
std::vector<int32_t> internalBuffer;
inline int64_t getTotalSamples() const { return WavpackGetNumSamples(context); }

Binary file not shown.

@ -61,12 +61,12 @@
#define flac_min(a,b) __min(a,b)
#endif
#ifndef flac_min
#define flac_min(x,y) ((x) <= (y) ? (x) : (y))
#ifndef MIN
#define MIN(x,y) ((x) <= (y) ? (x) : (y))
#endif
#ifndef flac_max
#define flac_max(x,y) ((x) >= (y) ? (x) : (y))
#ifndef MAX
#define MAX(x,y) ((x) >= (y) ? (x) : (y))
#endif
#endif

@ -0,0 +1,27 @@
*~
aclocal.m4
autom4te.cache
config.guess
config.log
config.status
config.sub
configure
depcomp
install-sh
libmodplug.pc
libtool
ltmain.sh
Makefile
Makefile.in
missing
src/.deps
src/.libs
*.la
*.lo
src/config.h
src/stamp-h1
*.o
*.po
*.Plo
*.swp
*.swo

@ -0,0 +1,22 @@
XMMS plugin:
Kenton Varda <temporal@gauge3d.org>
Konstanty Bialkowski <konstanty@ieee.org>
General Maintainence:
Konstanty Bialkowski <konstanty@ieee.org>
Sound Engine:
Olivier Lapicque <olivierl@jps.net>
BZip2 support:
Colin DeVilbiss <crdevilb@mtu.edu>
Spline and Fir resamplers:
Markus Fick <marf@gmx.net>
Endianness Fixes:
Adam Goode <adam@evdebs.org>
Endianness Fixes + Implementation of C 24bit,32bit functions:
Marco Trillo <toad@arsystel.com>
Fixes to AGC/Clipping, Frequency Limit, Other Fixes:
Alistair John Strachan <s0348365@sms.ed.ac.uk>
AMD64 Fix (long long vs long vs int)
Tyler Montbriand <tsm@accesscomm.ca>

@ -0,0 +1 @@
ModPlug-XMMS and libmodplug are now in the public domain.

@ -0,0 +1,225 @@
libmodplug - the library which was part of the Modplug-xmms project
Web page: http://modplug-xmms.sf.net/
Based on the ModPlug sound engine by Olivier Lapicque <olivierl@jps.net>
XMMS plugin by Kenton Varda <temporal@gauge3d.org> (~2002)
Maintainer is now Konstanty Bialkowski <konstanty@ieee.org> (~2006)
On Wed 14 Aug 2013 the repository was forked / cloned to GitHub.
The current release is libmodplug v0.8.8.5.
History
-------
Olivier Lapicque, author of Modplug, which is arguably the best quality
MOD-playing software available, has placed his sound rendering code in the
public domain. This library and plugin is based on that code.
This code was originally part of modplug-xmms, and was split into a library - libmodplug
and the modplug-xmms code. Also since then an example rendering project called modplugplay and
modplug123 were introduced. They are still available on the sourceforge website.
For more information on libmodplug, the library for decoding mod-like music
formats, see libmodplug/README.
Contents
--------
1. Requirements
2. Features
3. Options
4. Troubleshooting
---------------
1. Requirements
---------------
- POSIX OS (Linux or other unix*)
- XMMS 1.0.0 or higher (only for modplug-xmms plugin).
* This library is only guaranteed to work on Linux. I have received
conflicting reports on whether or not it will work on Solaris x86.
One person reported that the plugin compiled fine with the
"-fpermissive" compiler flag, which I have added. Others had far
more trouble. Note that a recent change to the library should allow
it to work on PPC and other big-endian systems.
* Under linux there is also modplugplay contributed, which allows command
line playing of mod files under Linux. (Available at http://modplug-xmms.sf.net/)
-----------
2. Features
-----------
- Plays 22 different mod formats, including:
MOD, S3M, XM, IT, 669, AMF (both of them), AMS, DBM, DMF, DSM, FAR,
MDL, MED, MTM, OKT, PTM, STM, ULT, UMX, MT2, PSM
- Plays zip, rar, gzip, and bzip2 compressed mods. The following
extensions are recognized:
zip: MDZ, S3Z, XMZ, ITZ
rar: MDR, S3R, XMR, ITR
gzip: MDGZ, S3GZ, XMGZ, ITGZ
You can also load plain old ZIP, RAR, and GZ files. If ModPlug finds
a mod in them, it will play it.
Note: To play these formats, you need to have the associated
decompression utilities (unzip, gunzip, unrar) installed.
Note(2): The format of the mod is NOT determined from the extension on
compressed mods. For example, if you zipped a UMX mod and gave it the
extension MDZ, it would work fine.
- plays timidity's GUS patch files (*.pat):
a multi sample pat file with n samples can be played with a Frere Jacques
canon with n voices.
- plays all types of MIDI files (*.mid):
uses the timidity .pat files for samples (when available)
recognizes environment variables:
MMPAT_PATH_TO_CFG set to the directory where the file "timidity.cfg" and
the subdirectory "instruments" can be found,
default: "/usr/local/share/timidity".
MMMID_SPEED for experimenting with the mod speed (1 thru 9)
MMMID_VERBOSE for feedback on the conversion process
MMMID_DEBUG for sake of completeness, only useful for maintainers
- plays textfiles written in the ABC music notation (*.abc):
uses the timidity .pat files for samples (when available)
recognizes environment variables:
MMPAT_PATH_TO_CFG set to the directory where the file "timidity.cfg" and
the subdirectory "instruments" can be found,
default: "/usr/local/share/timidity".
MMABC_NO_RANDOM_PICK when not set and the abc file contains multiple songs
(X:n) the first song to be played will be picked at random another click
on the play button advances to the next
song in the file (or the first when the last song has been
played), when set it can be 0 (zero) or not numeric
to let it play all songs in the file, a positive number n to
let it play the n-th song in the file, a negative number -n to
let it play the n-th song in the file and advancing to the next
song when the play button is clicked.
MMABC_DUMPTRACKS when set it gives diagnostic information on stdout,
values can be:
all - every event is printed
nonotes - only the control events (looping, breaks etc.) are printed
any other value prints the control events and every note event
immediately succeeding the control events.
- Slightly better sound quality than Mikmod. Vastly superior quality
over Winamp.
- All XMMS calls are supported except for the band gains on the
equalizer. The preamp is supported, but MOD music is not anywhere
near as cheap to equalize as MP3. Thus, equalization does is not
supported in this version. However, a variable bass boost option
is available in the configuration dialog (see below).
- Tons of playback options (see below).
----------
3. Options
----------
All of the following items are configurable from the plugin
configuration dialog box.
Sampling rate: Higher is better. Note that the sound is rendered at a
higher sampling rate and converted down to increase quality.
Bits per sample: 8-bit or 16-bit sound. Note that all computations are
done at 32-bit and converted down to the sampling rate you specify.
Channels: mono/stereo. Note that all computations are done in stereo.
If you choose mono, the channels will be mixed.
Resampling: Method used to convert samples to different sampling rates.
"Nearest" is the fastest setting (but sounds terrible), while
"8-tap fir" is the best-quality setting.
Noise Reduction: Reduces noise. :)
Fast Playlist Info: When this option is on, names of songs in your
playlist will load considerably faster, but song lengths will not be
shown and only MOD, S3M, XM, and IT formats will have their names shown.
Don't worry, though, because all the data which is skipped will still be
loaded when you actually play the song. This should probably always be
on.
Reverb: A nice reverb effect. The depth and delay of the reverb can be
tuned to your liking using the sliders.
Bass boost: Variable bass boost effect. The "range" slider controls the
frequency range of the bass boost. If you increase this value, higher
frequencies will be boosted, but the overall volume increase will be
less. (you can compensate by using the volume slider:)
Surround: Dolby Pro-Logic surround effect. Depth and delay can be fine
tuned.
Preamp: A global volume boost. Note that setting the preamp too high
will cause clipping (nasty clicks and pops).
Looping: Some mods have loops built-in. Normally, these loops are
ignored because otherwise the same mod would play forever. However,
you can choose to respect the loops, or even set a number of times to
follow a loop.
------------------
4. Troubleshooting
------------------
Problem:
Some of my files load up, but show garbled info in the playlist and/or
don't play correctly.
Possible cause:
The mod is in a different format than its file format suggests.
Modplug-XMMS uses a combination of file extension and contents to figure
out what format a mod is in, and can be thrown off if a mod is
incorrectly labeled.
Solution:
Turning off "fast info" in the configuration may fix the problem. This
will cause Modplug-XMMS to detect all basic mod types by content, but
archive types will still be detected by extension. If this doesn't
solve the problem, then you probably have files which are actually
compressed archives but are not labeled as such. For example, you may
have a file "aws_anew.xm" which is actually a ZIP archive. You will
have to either unzip these files or rename them to have an extension
associated with their type. In the case of a ZIP, you can use any of
the extensions "ZIP, MDZ, S3Z, XMZ, ITZ". (Note that these five types
are all treated exactly the same -- the actual format of the mod is
detected by contents.)
Problem:
Everything appears to be working, but no sound is being generated.
MP3's play just fine.
Possible cause:
Modplug has a relatively low default volume, and you may just not be
hearing it. (Note: Yes, more that one person has e-mailed me with
this problem.)
Solution:
Turn up your volume. You may wish to do this via the "preamp"
setting in the ModPlug configuration. This way, you won't have to
turn down your volume again when you play an MP3.
Problem:
You have a mod which is rendered incorrectly by ModPlug-XMMS.
Possible cause:
This could be our fault. :)
Solution:
First, test the mod using the Windows version of ModPlug, if you can.
If it sounds wrong there, then send the mod and a bug report to
Olivier Lapicque <olivierl@jps.net>. If the mod plays correctly in
Windows, however, then the bug is my fault. In that case, e-mail
me (Konstanty) <konstanty@ieee.org>. (previously Kenton Varda at
<temporal@gauge3d.org>).
Problem:
I have a problem which is not listed here, or an idea for a cool
feature.
Solution:
E-mail me (Konstanty) at <konstanty@ieee.org>. I would be
happy to hear any suggestions or problems you have.

File diff suppressed because it is too large Load Diff

@ -0,0 +1,134 @@
#ifndef _ITDEFS_H_
#define _ITDEFS_H_
#pragma pack(1)
typedef struct tagITFILEHEADER
{
DWORD id; // 0x4D504D49
CHAR songname[26];
WORD reserved1; // 0x1004
WORD ordnum;
WORD insnum;
WORD smpnum;
WORD patnum;
WORD cwtv;
WORD cmwt;
WORD flags;
WORD special;
BYTE globalvol;
BYTE mv;
BYTE speed;
BYTE tempo;
BYTE sep;
BYTE zero;
WORD msglength;
DWORD msgoffset;
DWORD reserved2;
BYTE chnpan[64];
BYTE chnvol[64];
} ITFILEHEADER;
typedef struct tagITENVELOPE
{
BYTE flags;
BYTE num;
BYTE lpb;
BYTE lpe;
BYTE slb;
BYTE sle;
BYTE data[25*3];
BYTE reserved;
} ITENVELOPE;
// Old Impulse Instrument Format (cmwt < 0x200)
typedef struct tagITOLDINSTRUMENT
{
DWORD id; // IMPI = 0x49504D49
CHAR filename[12]; // DOS file name
BYTE zero;
BYTE flags;
BYTE vls;
BYTE vle;
BYTE sls;
BYTE sle;
WORD reserved1;
WORD fadeout;
BYTE nna;
BYTE dnc;
WORD trkvers;
BYTE nos;
BYTE reserved2;
CHAR name[26];
WORD reserved3[3];
BYTE keyboard[240];
BYTE volenv[200];
BYTE nodes[50];
} ITOLDINSTRUMENT;
// Impulse Instrument Format
typedef struct tagITINSTRUMENT
{
DWORD id;
CHAR filename[12];
BYTE zero;
BYTE nna;
BYTE dct;
BYTE dca;
WORD fadeout;
signed char pps;
BYTE ppc;
BYTE gbv;
BYTE dfp;
BYTE rv;
BYTE rp;
WORD trkvers;
BYTE nos;
BYTE reserved1;
CHAR name[26];
BYTE ifc;
BYTE ifr;
BYTE mch;
BYTE mpr;
WORD mbank;
BYTE keyboard[240];
ITENVELOPE volenv;
ITENVELOPE panenv;
ITENVELOPE pitchenv;
BYTE dummy[4]; // was 7, but IT v2.17 saves 554 bytes
} ITINSTRUMENT;
// IT Sample Format
typedef struct ITSAMPLESTRUCT
{
DWORD id; // 0x53504D49
CHAR filename[12];
BYTE zero;
BYTE gvl;
BYTE flags;
BYTE vol;
CHAR name[26];
BYTE cvt;
BYTE dfp;
DWORD length;
DWORD loopbegin;
DWORD loopend;
DWORD C5Speed;
DWORD susloopbegin;
DWORD susloopend;
DWORD samplepointer;
BYTE vis;
BYTE vid;
BYTE vir;
BYTE vit;
} ITSAMPLESTRUCT;
#pragma pack()
extern BYTE autovibit2xm[8];
extern BYTE autovibxm2it[8];
#endif

@ -0,0 +1,195 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
////////////////////////////////////////////////////////////
// 669 Composer / UNIS 669 module loader
////////////////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
typedef struct tagFILEHEADER669
{
WORD sig; // 'if' or 'JN'
signed char songmessage[108]; // Song Message
BYTE samples; // number of samples (1-64)
BYTE patterns; // number of patterns (1-128)
BYTE restartpos;
BYTE orders[128];
BYTE tempolist[128];
BYTE breaks[128];
} FILEHEADER669;
typedef struct tagSAMPLE669
{
BYTE filename[13];
BYTE length[4]; // when will somebody think about DWORD align ???
BYTE loopstart[4];
BYTE loopend[4];
} SAMPLE669;
DWORD lengthArrayToDWORD(const BYTE length[4]) {
DWORD len = (length[3] << 24) +
(length[2] << 16) +
(length[1] << 8) +
(length[0]);
return(len);
}
BOOL CSoundFile::Read669(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
BOOL b669Ext;
const FILEHEADER669 *pfh = (const FILEHEADER669 *)lpStream;
const SAMPLE669 *psmp = (const SAMPLE669 *)(lpStream + 0x1F1);
DWORD dwMemPos = 0;
if ((!lpStream) || (dwMemLength < sizeof(FILEHEADER669))) return FALSE;
if ((bswapLE16(pfh->sig) != 0x6669) && (bswapLE16(pfh->sig) != 0x4E4A)) return FALSE;
b669Ext = (bswapLE16(pfh->sig) == 0x4E4A) ? TRUE : FALSE;
if ((!pfh->samples) || (pfh->samples > 64) || (pfh->restartpos >= 128)
|| (!pfh->patterns) || (pfh->patterns > 128)) return FALSE;
DWORD dontfuckwithme = 0x1F1 + pfh->samples * sizeof(SAMPLE669) + pfh->patterns * 0x600;
if (dontfuckwithme > dwMemLength) return FALSE;
for (UINT ichk=0; ichk<pfh->samples; ichk++)
{
DWORD len = lengthArrayToDWORD(psmp[ichk].length);
dontfuckwithme += len;
}
if (dontfuckwithme > dwMemLength) return FALSE;
// That should be enough checking: this must be a 669 module.
m_nType = MOD_TYPE_669;
m_dwSongFlags |= SONG_LINEARSLIDES;
m_nMinPeriod = 28 << 2;
m_nMaxPeriod = 1712 << 3;
m_nDefaultTempo = 125;
m_nDefaultSpeed = 6;
m_nChannels = 8;
memcpy(m_szNames[0], pfh->songmessage, 16);
m_nSamples = pfh->samples;
for (UINT nins=1; nins<=m_nSamples; nins++, psmp++)
{
DWORD len = lengthArrayToDWORD(psmp->length);
DWORD loopstart = lengthArrayToDWORD(psmp->loopstart);
DWORD loopend = lengthArrayToDWORD(psmp->loopend);
if (len > MAX_SAMPLE_LENGTH) len = MAX_SAMPLE_LENGTH;
if ((loopend > len) && (!loopstart)) loopend = 0;
if (loopend > len) loopend = len;
if (loopstart + 4 >= loopend) loopstart = loopend = 0;
Ins[nins].nLength = len;
Ins[nins].nLoopStart = loopstart;
Ins[nins].nLoopEnd = loopend;
if (loopend) Ins[nins].uFlags |= CHN_LOOP;
memcpy(m_szNames[nins], psmp->filename, 13);
Ins[nins].nVolume = 256;
Ins[nins].nGlobalVol = 64;
Ins[nins].nPan = 128;
}
// Song Message
m_lpszSongComments = new char[109];
memcpy(m_lpszSongComments, pfh->songmessage, 108);
m_lpszSongComments[108] = 0;
// Reading Orders
memcpy(Order, pfh->orders, 128);
m_nRestartPos = pfh->restartpos;
if (Order[m_nRestartPos] >= pfh->patterns) m_nRestartPos = 0;
// Reading Pattern Break Locations
for (UINT npan=0; npan<8; npan++)
{
ChnSettings[npan].nPan = (npan & 1) ? 0x30 : 0xD0;
ChnSettings[npan].nVolume = 64;
}
// Reading Patterns
dwMemPos = 0x1F1 + pfh->samples * 25;
for (UINT npat=0; npat<pfh->patterns; npat++)
{
Patterns[npat] = AllocatePattern(64, m_nChannels);
if (!Patterns[npat]) break;
PatternSize[npat] = 64;
MODCOMMAND *m = Patterns[npat];
const BYTE *p = lpStream + dwMemPos;
for (UINT row=0; row<64; row++)
{
MODCOMMAND *mspeed = m;
if ((row == pfh->breaks[npat]) && (row != 63))
{
for (UINT i=0; i<8; i++)
{
m[i].command = CMD_PATTERNBREAK;
m[i].param = 0;
}
}
for (UINT n=0; n<8; n++, m++, p+=3)
{
UINT note = p[0] >> 2;
UINT instr = ((p[0] & 0x03) << 4) | (p[1] >> 4);
UINT vol = p[1] & 0x0F;
if (p[0] < 0xFE)
{
m->note = note + 37;
m->instr = instr + 1;
}
if (p[0] <= 0xFE)
{
m->volcmd = VOLCMD_VOLUME;
m->vol = (vol << 2) + 2;
}
if (p[2] != 0xFF)
{
UINT command = p[2] >> 4;
UINT param = p[2] & 0x0F;
switch(command)
{
case 0x00: command = CMD_PORTAMENTOUP; break;
case 0x01: command = CMD_PORTAMENTODOWN; break;
case 0x02: command = CMD_TONEPORTAMENTO; break;
case 0x03: command = CMD_MODCMDEX; param |= 0x50; break;
case 0x04: command = CMD_VIBRATO; param |= 0x40; break;
case 0x05: if (param) command = CMD_SPEED; else command = 0; param += 2; break;
case 0x06: if (param == 0) { command = CMD_PANNINGSLIDE; param = 0xFE; } else
if (param == 1) { command = CMD_PANNINGSLIDE; param = 0xEF; } else
command = 0;
break;
default: command = 0;
}
if (command)
{
if (command == CMD_SPEED) mspeed = NULL;
m->command = command;
m->param = param;
}
}
}
if ((!row) && (mspeed))
{
for (UINT i=0; i<8; i++) if (!mspeed[i].command)
{
mspeed[i].command = CMD_SPEED;
mspeed[i].param = pfh->tempolist[npat] + 2;
break;
}
}
}
dwMemPos += 0x600;
}
// Reading Samples
for (UINT n=1; n<=m_nSamples; n++)
{
UINT len = Ins[n].nLength;
if (dwMemPos >= dwMemLength) break;
if (len > 4) ReadSample(&Ins[n], RS_PCM8U, (LPSTR)(lpStream+dwMemPos), dwMemLength - dwMemPos);
dwMemPos += len;
}
return TRUE;
}

File diff suppressed because it is too large Load Diff

@ -0,0 +1,422 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
///////////////////////////////////////////////////
//
// AMF module loader
//
// There is 2 types of AMF files:
// - ASYLUM Music Format
// - Advanced Music Format(DSM)
//
///////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#define AMFLOG
//#pragma warning(disable:4244)
#pragma pack(1)
typedef struct _AMFFILEHEADER
{
UCHAR szAMF[3];
UCHAR version;
CHAR title[32];
UCHAR numsamples;
UCHAR numorders;
USHORT numtracks;
UCHAR numchannels;
} AMFFILEHEADER;
typedef struct _AMFSAMPLE
{
UCHAR type;
CHAR samplename[32];
CHAR filename[13];
ULONG offset;
ULONG length;
USHORT c2spd;
UCHAR volume;
} AMFSAMPLE;
#pragma pack()
#ifdef AMFLOG
extern void Log(LPCSTR, ...);
#endif
VOID AMF_Unpack(MODCOMMAND *pPat, const BYTE *pTrack, UINT nRows, UINT nChannels)
//-------------------------------------------------------------------------------
{
UINT lastinstr = 0;
UINT nTrkSize = bswapLE16(*(USHORT *)pTrack);
nTrkSize += (UINT)pTrack[2] << 16;
pTrack += 3;
while (nTrkSize--)
{
UINT row = pTrack[0];
UINT cmd = pTrack[1];
UINT arg = pTrack[2];
if (row >= nRows) break;
MODCOMMAND *m = pPat + row * nChannels;
if (cmd < 0x7F) // note+vol
{
m->note = cmd+1;
if (!m->instr) m->instr = lastinstr;
m->volcmd = VOLCMD_VOLUME;
m->vol = arg;
} else
if (cmd == 0x7F) // duplicate row
{
signed char rdelta = (signed char)arg;
int rowsrc = (int)row + (int)rdelta;
if ((rowsrc >= 0) && (rowsrc < (int)nRows)) memcpy(m, &pPat[rowsrc*nChannels],sizeof(pPat[rowsrc*nChannels]));
} else
if (cmd == 0x80) // instrument
{
m->instr = arg+1;
lastinstr = m->instr;
} else
if (cmd == 0x83) // volume
{
m->volcmd = VOLCMD_VOLUME;
m->vol = arg;
} else
// effect
{
UINT command = cmd & 0x7F;
UINT param = arg;
switch(command)
{
// 0x01: Set Speed
case 0x01: command = CMD_SPEED; break;
// 0x02: Volume Slide
// 0x0A: Tone Porta + Vol Slide
// 0x0B: Vibrato + Vol Slide
case 0x02: command = CMD_VOLUMESLIDE;
case 0x0A: if (command == 0x0A) command = CMD_TONEPORTAVOL;
case 0x0B: if (command == 0x0B) command = CMD_VIBRATOVOL;
if (param & 0x80) param = (-(signed char)param)&0x0F;
else param = (param&0x0F)<<4;
break;
// 0x04: Porta Up/Down
case 0x04: if (param & 0x80) { command = CMD_PORTAMENTOUP; param = (-(signed char)param)&0x7F; }
else { command = CMD_PORTAMENTODOWN; } break;
// 0x06: Tone Portamento
case 0x06: command = CMD_TONEPORTAMENTO; break;
// 0x07: Tremor
case 0x07: command = CMD_TREMOR; break;
// 0x08: Arpeggio
case 0x08: command = CMD_ARPEGGIO; break;
// 0x09: Vibrato
case 0x09: command = CMD_VIBRATO; break;
// 0x0C: Pattern Break
case 0x0C: command = CMD_PATTERNBREAK; break;
// 0x0D: Position Jump
case 0x0D: command = CMD_POSITIONJUMP; break;
// 0x0F: Retrig
case 0x0F: command = CMD_RETRIG; break;
// 0x10: Offset
case 0x10: command = CMD_OFFSET; break;
// 0x11: Fine Volume Slide
case 0x11: if (param) { command = CMD_VOLUMESLIDE;
if (param & 0x80) param = 0xF0|((-(signed char)param)&0x0F);
else param = 0x0F|((param&0x0F)<<4);
} else command = 0; break;
// 0x12: Fine Portamento
// 0x16: Extra Fine Portamento
case 0x12:
case 0x16: if (param) { int mask = (command == 0x16) ? 0xE0 : 0xF0;
command = (param & 0x80) ? CMD_PORTAMENTOUP : CMD_PORTAMENTODOWN;
if (param & 0x80) param = mask|((-(signed char)param)&0x0F);
else param |= mask;
} else command = 0; break;
// 0x13: Note Delay
case 0x13: command = CMD_S3MCMDEX; param = 0xD0|(param & 0x0F); break;
// 0x14: Note Cut
case 0x14: command = CMD_S3MCMDEX; param = 0xC0|(param & 0x0F); break;
// 0x15: Set Tempo
case 0x15: command = CMD_TEMPO; break;
// 0x17: Panning
case 0x17: param = (param+64)&0x7F;
if (m->command) { if (!m->volcmd) { m->volcmd = VOLCMD_PANNING; m->vol = param/2; } command = 0; }
else { command = CMD_PANNING8; }
// Unknown effects
default: command = param = 0;
}
if (command)
{
m->command = command;
m->param = param;
}
}
pTrack += 3;
}
}
BOOL CSoundFile::ReadAMF(LPCBYTE lpStream, const DWORD dwMemLength)
//-----------------------------------------------------------
{
const AMFFILEHEADER *pfh = (AMFFILEHEADER *)lpStream;
DWORD dwMemPos;
if ((!lpStream) || (dwMemLength < 2048)) return FALSE;
if ((!strncmp((LPCTSTR)lpStream, "ASYLUM Music Format V1.0", 25)) && (dwMemLength > 4096))
{
UINT numorders, numpats, numsamples;
dwMemPos = 32;
numpats = lpStream[dwMemPos+3];
numorders = lpStream[dwMemPos+4];
numsamples = 64;
dwMemPos += 6;
if ((!numpats) || (numpats > MAX_PATTERNS) || (!numorders)
|| (numpats*64*32 + 294 + 37*64 >= dwMemLength)) return FALSE;
m_nType = MOD_TYPE_AMF0;
m_nChannels = 8;
m_nInstruments = 0;
m_nSamples = 31;
m_nDefaultTempo = 125;
m_nDefaultSpeed = 6;
for (UINT iOrd=0; iOrd<MAX_ORDERS; iOrd++)
{
Order[iOrd] = (iOrd < numorders) ? lpStream[dwMemPos+iOrd] : 0xFF;
}
dwMemPos = 294; // ???
for (UINT iSmp=0; iSmp<numsamples; iSmp++)
{
MODINSTRUMENT *psmp = &Ins[iSmp+1];
memcpy(m_szNames[iSmp+1], lpStream+dwMemPos, 22);
m_szNames[iSmp+1][21] = '\0';
psmp->nFineTune = MOD2XMFineTune(lpStream[dwMemPos+22]);
psmp->nVolume = lpStream[dwMemPos+23];
psmp->nGlobalVol = 64;
if (psmp->nVolume > 0x40) psmp->nVolume = 0x40;
psmp->nVolume <<= 2;
psmp->nLength = bswapLE32(*((LPDWORD)(lpStream+dwMemPos+25)));
psmp->nLoopStart = bswapLE32(*((LPDWORD)(lpStream+dwMemPos+29)));
psmp->nLoopEnd = psmp->nLoopStart + bswapLE32(*((LPDWORD)(lpStream+dwMemPos+33)));
if ((psmp->nLoopEnd > psmp->nLoopStart) && (psmp->nLoopEnd <= psmp->nLength))
{
psmp->uFlags = CHN_LOOP;
} else
{
psmp->nLoopStart = psmp->nLoopEnd = 0;
}
if ((psmp->nLength) && (iSmp>31)) m_nSamples = iSmp+1;
dwMemPos += 37;
}
for (UINT iPat=0; iPat<numpats; iPat++)
{
MODCOMMAND *p = AllocatePattern(64, m_nChannels);
if (!p) break;
Patterns[iPat] = p;
PatternSize[iPat] = 64;
const UCHAR *pin = lpStream + dwMemPos;
for (UINT i=0; i<8*64; i++)
{
p->note = 0;
if (pin[0])
{
p->note = pin[0] + 13;
}
p->instr = pin[1];
p->command = pin[2];
p->param = pin[3];
if (p->command > 0x0F)
{
#ifdef AMFLOG
Log("0x%02X.0x%02X ?", p->command, p->param);
#endif
p->command = 0;
}
ConvertModCommand(p);
pin += 4;
p++;
}
dwMemPos += 64*32;
}
// Read samples
for (UINT iData=0; iData<m_nSamples; iData++)
{
MODINSTRUMENT *psmp = &Ins[iData+1];
if (psmp->nLength)
{
if (dwMemPos > dwMemLength) return FALSE;
dwMemPos += ReadSample(psmp, RS_PCM8S, (LPCSTR)(lpStream+dwMemPos), dwMemLength-dwMemPos);
}
}
return TRUE;
}
////////////////////////////
// DSM/AMF
USHORT *ptracks[MAX_PATTERNS];
DWORD sampleseekpos[MAX_SAMPLES];
if ((pfh->szAMF[0] != 'A') || (pfh->szAMF[1] != 'M') || (pfh->szAMF[2] != 'F')
|| (pfh->version < 10) || (pfh->version > 14) || (!bswapLE16(pfh->numtracks))
|| (!pfh->numorders) || (pfh->numorders > MAX_PATTERNS)
|| (!pfh->numsamples) || (pfh->numsamples >= MAX_SAMPLES)
|| (pfh->numchannels < 4) || (pfh->numchannels > 32))
return FALSE;
memcpy(m_szNames[0], pfh->title, 32);
m_szNames[0][31] = '\0';
dwMemPos = sizeof(AMFFILEHEADER);
m_nType = MOD_TYPE_AMF;
m_nChannels = pfh->numchannels;
m_nSamples = pfh->numsamples;
m_nInstruments = 0;
// Setup Channel Pan Positions
if (pfh->version >= 11)
{
signed char *panpos = (signed char *)(lpStream + dwMemPos);
UINT nchannels = (pfh->version >= 13) ? 32 : 16;
for (UINT i=0; i<nchannels; i++)
{
int pan = (panpos[i] + 64) * 2;
if (pan < 0) pan = 0;
if (pan > 256) { pan = 128; ChnSettings[i].dwFlags |= CHN_SURROUND; }
ChnSettings[i].nPan = pan;
}
dwMemPos += nchannels;
} else
{
for (UINT i=0; i<16; i++)
{
ChnSettings[i].nPan = (lpStream[dwMemPos+i] & 1) ? 0x30 : 0xD0;
}
dwMemPos += 16;
}
// Get Tempo/Speed
m_nDefaultTempo = 125;
m_nDefaultSpeed = 6;
if (pfh->version >= 13)
{
if (lpStream[dwMemPos] >= 32) m_nDefaultTempo = lpStream[dwMemPos];
if (lpStream[dwMemPos+1] <= 32) m_nDefaultSpeed = lpStream[dwMemPos+1];
dwMemPos += 2;
}
// Setup sequence list
for (UINT iOrd=0; iOrd<MAX_ORDERS; iOrd++)
{
Order[iOrd] = 0xFF;
if (iOrd < pfh->numorders)
{
Order[iOrd] = iOrd;
PatternSize[iOrd] = 64;
if (pfh->version >= 14)
{
PatternSize[iOrd] = bswapLE16(*(USHORT *)(lpStream+dwMemPos));
dwMemPos += 2;
}
ptracks[iOrd] = (USHORT *)(lpStream+dwMemPos);
dwMemPos += m_nChannels * sizeof(USHORT);
}
}
if (dwMemPos + m_nSamples * (sizeof(AMFSAMPLE)+8) > dwMemLength) return TRUE;
// Read Samples
UINT maxsampleseekpos = 0;
for (UINT iIns=0; iIns<m_nSamples; iIns++)
{
MODINSTRUMENT *pins = &Ins[iIns+1];
const AMFSAMPLE *psh = (AMFSAMPLE *)(lpStream + dwMemPos);
dwMemPos += sizeof(AMFSAMPLE);
memcpy(m_szNames[iIns+1], psh->samplename, 32);
m_szNames[iIns+1][31] = '\0';
memcpy(pins->name, psh->filename, 13);
pins->name[12] = '\0';
pins->nLength = bswapLE32(psh->length);
pins->nC4Speed = bswapLE16(psh->c2spd);
pins->nGlobalVol = 64;
pins->nVolume = psh->volume * 4;
if (pfh->version >= 11)
{
pins->nLoopStart = bswapLE32(*(DWORD *)(lpStream+dwMemPos));
pins->nLoopEnd = bswapLE32(*(DWORD *)(lpStream+dwMemPos+4));
dwMemPos += 8;
} else
{
pins->nLoopStart = bswapLE16(*(WORD *)(lpStream+dwMemPos));
pins->nLoopEnd = pins->nLength;
dwMemPos += 2;
}
sampleseekpos[iIns] = 0;
if ((psh->type) && (bswapLE32(psh->offset) < dwMemLength-1))
{
sampleseekpos[iIns] = bswapLE32(psh->offset);
if (bswapLE32(psh->offset) > maxsampleseekpos)
maxsampleseekpos = bswapLE32(psh->offset);
if ((pins->nLoopEnd > pins->nLoopStart + 2)
&& (pins->nLoopEnd <= pins->nLength)) pins->uFlags |= CHN_LOOP;
}
}
// Read Track Mapping Table
USHORT *pTrackMap = (USHORT *)(lpStream+dwMemPos);
UINT realtrackcnt = 0;
dwMemPos += pfh->numtracks * sizeof(USHORT);
for (UINT iTrkMap=0; iTrkMap<pfh->numtracks; iTrkMap++)
{
if (realtrackcnt < pTrackMap[iTrkMap]) realtrackcnt = pTrackMap[iTrkMap];
}
// Store tracks positions
BYTE **pTrackData = new BYTE *[realtrackcnt];
memset(pTrackData, 0, sizeof(BYTE *) * realtrackcnt);
for (UINT iTrack=0; iTrack<realtrackcnt; iTrack++) if (dwMemPos <= dwMemLength - 3)
{
UINT nTrkSize = bswapLE16(*(USHORT *)(lpStream+dwMemPos));
nTrkSize += (UINT)lpStream[dwMemPos+2] << 16;
if (dwMemPos + nTrkSize * 3 + 3 <= dwMemLength)
{
pTrackData[iTrack] = (BYTE *)(lpStream + dwMemPos);
}
dwMemPos += nTrkSize * 3 + 3;
}
// Create the patterns from the list of tracks
for (UINT iPat=0; iPat<pfh->numorders; iPat++)
{
MODCOMMAND *p = AllocatePattern(PatternSize[iPat], m_nChannels);
if (!p) break;
Patterns[iPat] = p;
for (UINT iChn=0; iChn<m_nChannels; iChn++)
{
UINT nTrack = bswapLE16(ptracks[iPat][iChn]);
if ((nTrack) && (nTrack <= pfh->numtracks))
{
UINT realtrk = bswapLE16(pTrackMap[nTrack-1]);
if (realtrk)
{
realtrk--;
if ((realtrk < realtrackcnt) && (pTrackData[realtrk]))
{
AMF_Unpack(p+iChn, pTrackData[realtrk], PatternSize[iPat], m_nChannels);
}
}
}
}
}
delete[] pTrackData;
// Read Sample Data
for (UINT iSeek=1; iSeek<=maxsampleseekpos; iSeek++)
{
if (dwMemPos >= dwMemLength) break;
for (UINT iSmp=0; iSmp<m_nSamples; iSmp++) if (iSeek == sampleseekpos[iSmp])
{
MODINSTRUMENT *pins = &Ins[iSmp+1];
dwMemPos += ReadSample(pins, RS_PCM8U, (LPCSTR)(lpStream+dwMemPos), dwMemLength-dwMemPos);
break;
}
}
return TRUE;
}

@ -0,0 +1,628 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
//////////////////////////////////////////////
// AMS module loader //
//////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#pragma pack(1)
typedef struct AMSFILEHEADER
{
char szHeader[7]; // "Extreme" // changed from CHAR
BYTE verlo, verhi; // 0x??,0x01
BYTE chncfg;
BYTE samples;
WORD patterns;
WORD orders;
BYTE vmidi;
WORD extra;
} AMSFILEHEADER;
typedef struct AMSSAMPLEHEADER
{
DWORD length;
DWORD loopstart;
DWORD loopend;
BYTE finetune_and_pan;
WORD samplerate; // C-2 = 8363
BYTE volume; // 0-127
BYTE infobyte;
} AMSSAMPLEHEADER;
#pragma pack()
BOOL CSoundFile::ReadAMS(LPCBYTE lpStream, DWORD dwMemLength)
//-----------------------------------------------------------
{
BYTE pkinf[MAX_SAMPLES];
AMSFILEHEADER *pfh = (AMSFILEHEADER *)lpStream;
DWORD dwMemPos;
UINT tmp, tmp2;
if ((!lpStream) || (dwMemLength < 1024)) return FALSE;
if ((pfh->verhi != 0x01) || (strncmp(pfh->szHeader, "Extreme", 7))
|| (!pfh->patterns) || (!pfh->orders) || (!pfh->samples) || (pfh->samples >= MAX_SAMPLES)
|| (pfh->patterns > MAX_PATTERNS) || (pfh->orders > MAX_ORDERS))
{
return ReadAMS2(lpStream, dwMemLength);
}
dwMemPos = sizeof(AMSFILEHEADER) + pfh->extra;
if (dwMemPos + pfh->samples * sizeof(AMSSAMPLEHEADER) + 256 >= dwMemLength) return FALSE;
m_nType = MOD_TYPE_AMS;
m_nInstruments = 0;
m_nChannels = (pfh->chncfg & 0x1F) + 1;
m_nSamples = pfh->samples;
for (UINT nSmp=1; nSmp<=m_nSamples; nSmp++, dwMemPos += sizeof(AMSSAMPLEHEADER))
{
AMSSAMPLEHEADER *psh = (AMSSAMPLEHEADER *)(lpStream + dwMemPos);
MODINSTRUMENT *pins = &Ins[nSmp];
pins->nLength = psh->length;
pins->nLoopStart = psh->loopstart;
pins->nLoopEnd = psh->loopend;
pins->nGlobalVol = 64;
pins->nVolume = psh->volume << 1;
pins->nC4Speed = psh->samplerate;
pins->nPan = (psh->finetune_and_pan & 0xF0);
if (pins->nPan < 0x80) pins->nPan += 0x10;
pins->nFineTune = MOD2XMFineTune(psh->finetune_and_pan & 0x0F);
pins->uFlags = (psh->infobyte & 0x80) ? CHN_16BIT : 0;
if ((pins->nLoopEnd <= pins->nLength) && (pins->nLoopStart+4 <= pins->nLoopEnd)) pins->uFlags |= CHN_LOOP;
pkinf[nSmp] = psh->infobyte;
}
// Read Song Name
tmp = lpStream[dwMemPos++];
if (dwMemPos + tmp + 1 >= dwMemLength) return TRUE;
tmp2 = (tmp < 32) ? tmp : 31;
if (tmp2) memcpy(m_szNames[0], lpStream+dwMemPos, tmp2);
m_szNames[0][tmp2] = 0;
dwMemPos += tmp;
// Read sample names
for (UINT sNam=1; sNam<=m_nSamples; sNam++)
{
if (dwMemPos + 32 >= dwMemLength) return TRUE;
tmp = lpStream[dwMemPos++];
tmp2 = (tmp < 32) ? tmp : 31;
if (tmp2) memcpy(m_szNames[sNam], lpStream+dwMemPos, tmp2);
dwMemPos += tmp;
}
// Skip Channel names
for (UINT cNam=0; cNam<m_nChannels; cNam++)
{
if (dwMemPos + 32 >= dwMemLength) return TRUE;
tmp = lpStream[dwMemPos++];
dwMemPos += tmp;
}
// Read Pattern Names
m_lpszPatternNames = new char[pfh->patterns * 32]; // changed from CHAR
if (!m_lpszPatternNames) return TRUE;
m_nPatternNames = pfh->patterns;
memset(m_lpszPatternNames, 0, m_nPatternNames * 32);
for (UINT pNam=0; pNam < m_nPatternNames; pNam++)
{
if (dwMemPos + 32 >= dwMemLength) return TRUE;
tmp = lpStream[dwMemPos++];
tmp2 = (tmp < 32) ? tmp : 31;
if (tmp2) memcpy(m_lpszPatternNames+pNam*32, lpStream+dwMemPos, tmp2);
dwMemPos += tmp;
}
// Read Song Comments
tmp = *((WORD *)(lpStream+dwMemPos));
dwMemPos += 2;
if (dwMemPos + tmp >= dwMemLength) return TRUE;
if (tmp)
{
m_lpszSongComments = new char[tmp+1]; // changed from CHAR
if (!m_lpszSongComments) return TRUE;
memset(m_lpszSongComments, 0, tmp+1);
memcpy(m_lpszSongComments, lpStream + dwMemPos, tmp);
dwMemPos += tmp;
}
// Read Order List
for (UINT iOrd=0; iOrd<pfh->orders; iOrd++, dwMemPos += 2)
{
UINT n = *((WORD *)(lpStream+dwMemPos));
Order[iOrd] = (BYTE)n;
}
// Read Patterns
for (UINT iPat=0; iPat<pfh->patterns; iPat++)
{
if (dwMemPos + 4 >= dwMemLength) return TRUE;
UINT len = *((DWORD *)(lpStream + dwMemPos));
dwMemPos += 4;
if ((len >= dwMemLength) || (dwMemPos + len > dwMemLength)) return TRUE;
PatternSize[iPat] = 64;
MODCOMMAND *m = AllocatePattern(PatternSize[iPat], m_nChannels);
if (!m) return TRUE;
Patterns[iPat] = m;
const BYTE *p = lpStream + dwMemPos;
UINT row = 0, i = 0;
while ((row < PatternSize[iPat]) && (i+2 < len))
{
BYTE b0 = p[i++];
BYTE b1 = p[i++];
BYTE b2 = 0;
UINT ch = b0 & 0x3F;
// Note+Instr
if (!(b0 & 0x40))
{
b2 = p[i++];
if (ch < m_nChannels)
{
if (b1 & 0x7F) m[ch].note = (b1 & 0x7F) + 25;
m[ch].instr = b2;
}
if (b1 & 0x80)
{
b0 |= 0x40;
b1 = p[i++];
}
}
// Effect
if (b0 & 0x40)
{
anothercommand:
if (b1 & 0x40)
{
if (ch < m_nChannels)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = b1 & 0x3F;
}
} else
{
b2 = p[i++];
if (ch < m_nChannels)
{
UINT cmd = b1 & 0x3F;
if (cmd == 0x0C)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = b2 >> 1;
} else
if (cmd == 0x0E)
{
if (!m[ch].command)
{
UINT command = CMD_S3MCMDEX;
UINT param = b2;
switch(param & 0xF0)
{
case 0x00: if (param & 0x08) { param &= 0x07; param |= 0x90; } else {command=param=0;} break;
case 0x10: command = CMD_PORTAMENTOUP; param |= 0xF0; break;
case 0x20: command = CMD_PORTAMENTODOWN; param |= 0xF0; break;
case 0x30: param = (param & 0x0F) | 0x10; break;
case 0x40: param = (param & 0x0F) | 0x30; break;
case 0x50: param = (param & 0x0F) | 0x20; break;
case 0x60: param = (param & 0x0F) | 0xB0; break;
case 0x70: param = (param & 0x0F) | 0x40; break;
case 0x90: command = CMD_RETRIG; param &= 0x0F; break;
case 0xA0: if (param & 0x0F) { command = CMD_VOLUMESLIDE; param = (param << 4) | 0x0F; } else command=param=0; break;
case 0xB0: if (param & 0x0F) { command = CMD_VOLUMESLIDE; param |= 0xF0; } else command=param=0; break;
}
m[ch].command = command;
m[ch].param = param;
}
} else
{
m[ch].command = cmd;
m[ch].param = b2;
ConvertModCommand(&m[ch]);
}
}
}
if (b1 & 0x80)
{
b1 = p[i++];
if (i <= len) goto anothercommand;
}
}
if (b0 & 0x80)
{
row++;
m += m_nChannels;
}
}
dwMemPos += len;
}
// Read Samples
for (UINT iSmp=1; iSmp<=m_nSamples; iSmp++) if (Ins[iSmp].nLength)
{
if (dwMemPos >= dwMemLength - 9) return TRUE;
UINT flags = (Ins[iSmp].uFlags & CHN_16BIT) ? RS_AMS16 : RS_AMS8;
dwMemPos += ReadSample(&Ins[iSmp], flags, (LPSTR)(lpStream+dwMemPos), dwMemLength-dwMemPos);
}
return TRUE;
}
/////////////////////////////////////////////////////////////////////
// AMS 2.2 loader
#pragma pack(1)
typedef struct AMS2FILEHEADER
{
DWORD dwHdr1; // AMShdr
WORD wHdr2;
BYTE b1A; // 0x1A
BYTE titlelen; // 30-bytes max
CHAR szTitle[30]; // [titlelen]
} AMS2FILEHEADER;
typedef struct AMS2SONGHEADER
{
WORD version;
BYTE instruments;
WORD patterns;
WORD orders;
WORD bpm;
BYTE speed;
BYTE channels;
BYTE commands;
BYTE rows;
WORD flags;
} AMS2SONGHEADER;
typedef struct AMS2INSTRUMENT
{
BYTE samples;
BYTE notemap[NOTE_MAX];
} AMS2INSTRUMENT;
typedef struct AMS2ENVELOPE
{
BYTE speed;
BYTE sustain;
BYTE loopbegin;
BYTE loopend;
BYTE points;
BYTE info[3];
} AMS2ENVELOPE;
typedef struct AMS2SAMPLE
{
DWORD length;
DWORD loopstart;
DWORD loopend;
WORD frequency;
BYTE finetune;
WORD c4speed;
CHAR transpose;
BYTE volume;
BYTE flags;
} AMS2SAMPLE;
#pragma pack()
BOOL CSoundFile::ReadAMS2(LPCBYTE lpStream, DWORD dwMemLength)
//------------------------------------------------------------
{
const AMS2FILEHEADER *pfh = (AMS2FILEHEADER *)lpStream;
AMS2SONGHEADER *psh;
DWORD dwMemPos;
BYTE smpmap[16];
BYTE packedsamples[MAX_SAMPLES];
if ((pfh->dwHdr1 != 0x68534D41) || (pfh->wHdr2 != 0x7264)
|| (pfh->b1A != 0x1A) || (pfh->titlelen > 30)) return FALSE;
dwMemPos = pfh->titlelen + 8;
psh = (AMS2SONGHEADER *)(lpStream + dwMemPos);
if (((psh->version & 0xFF00) != 0x0200) || (!psh->instruments)
|| (psh->instruments >= MAX_INSTRUMENTS) || (!psh->patterns) || (!psh->orders)) return FALSE;
dwMemPos += sizeof(AMS2SONGHEADER);
if (pfh->titlelen)
{
memcpy(m_szNames, pfh->szTitle, pfh->titlelen);
m_szNames[0][pfh->titlelen] = 0;
}
m_nType = MOD_TYPE_AMS;
m_nChannels = 32;
m_nDefaultTempo = psh->bpm >> 8;
m_nDefaultSpeed = psh->speed;
m_nInstruments = psh->instruments;
m_nSamples = 0;
if (psh->flags & 0x40) m_dwSongFlags |= SONG_LINEARSLIDES;
for (UINT nIns=1; nIns<=m_nInstruments; nIns++)
{
UINT insnamelen = lpStream[dwMemPos];
CHAR *pinsname = (CHAR *)(lpStream+dwMemPos+1);
dwMemPos += insnamelen + 1;
AMS2INSTRUMENT *pins = (AMS2INSTRUMENT *)(lpStream + dwMemPos);
dwMemPos += sizeof(AMS2INSTRUMENT);
if (dwMemPos + 1024 >= dwMemLength) return TRUE;
AMS2ENVELOPE *volenv, *panenv, *pitchenv;
volenv = (AMS2ENVELOPE *)(lpStream+dwMemPos);
dwMemPos += 5 + volenv->points*3;
panenv = (AMS2ENVELOPE *)(lpStream+dwMemPos);
dwMemPos += 5 + panenv->points*3;
pitchenv = (AMS2ENVELOPE *)(lpStream+dwMemPos);
dwMemPos += 5 + pitchenv->points*3;
INSTRUMENTHEADER *penv = new INSTRUMENTHEADER;
if (!penv) return TRUE;
memset(smpmap, 0, sizeof(smpmap));
memset(penv, 0, sizeof(INSTRUMENTHEADER));
for (UINT ismpmap=0; ismpmap<pins->samples; ismpmap++)
{
if ((ismpmap >= 16) || (m_nSamples+1 >= MAX_SAMPLES)) break;
m_nSamples++;
smpmap[ismpmap] = m_nSamples;
}
penv->nGlobalVol = 64;
penv->nPan = 128;
penv->nPPC = 60;
Headers[nIns] = penv;
if (insnamelen)
{
if (insnamelen > 31) insnamelen = 31;
memcpy(penv->name, pinsname, insnamelen);
penv->name[insnamelen] = 0;
}
for (UINT inotemap=0; inotemap<NOTE_MAX; inotemap++)
{
penv->NoteMap[inotemap] = inotemap+1;
penv->Keyboard[inotemap] = smpmap[pins->notemap[inotemap] & 0x0F];
}
// Volume Envelope
{
UINT pos = 0;
penv->nVolEnv = (volenv->points > 16) ? 16 : volenv->points;
penv->nVolSustainBegin = penv->nVolSustainEnd = volenv->sustain;
penv->nVolLoopStart = volenv->loopbegin;
penv->nVolLoopEnd = volenv->loopend;
for (UINT i=0; i<penv->nVolEnv; i++)
{
penv->VolEnv[i] = (BYTE)((volenv->info[i*3+2] & 0x7F) >> 1);
pos += volenv->info[i*3] + ((volenv->info[i*3+1] & 1) << 8);
penv->VolPoints[i] = (WORD)pos;
}
}
penv->nFadeOut = (((lpStream[dwMemPos+2] & 0x0F) << 8) | (lpStream[dwMemPos+1])) << 3;
UINT envflags = lpStream[dwMemPos+3];
if (envflags & 0x01) penv->dwFlags |= ENV_VOLLOOP;
if (envflags & 0x02) penv->dwFlags |= ENV_VOLSUSTAIN;
if (envflags & 0x04) penv->dwFlags |= ENV_VOLUME;
dwMemPos += 5;
// Read Samples
for (UINT ismp=0; ismp<pins->samples; ismp++)
{
MODINSTRUMENT *psmp = ((ismp < 16) && (smpmap[ismp])) ? &Ins[smpmap[ismp]] : NULL;
UINT smpnamelen = lpStream[dwMemPos];
if ((psmp) && (smpnamelen) && (smpnamelen <= 22))
{
memcpy(m_szNames[smpmap[ismp]], lpStream+dwMemPos+1, smpnamelen);
}
dwMemPos += smpnamelen + 1;
if (psmp)
{
AMS2SAMPLE *pams = (AMS2SAMPLE *)(lpStream+dwMemPos);
psmp->nGlobalVol = 64;
psmp->nPan = 128;
psmp->nLength = pams->length;
psmp->nLoopStart = pams->loopstart;
psmp->nLoopEnd = pams->loopend;
psmp->nC4Speed = pams->c4speed;
psmp->RelativeTone = pams->transpose;
psmp->nVolume = pams->volume / 2;
packedsamples[smpmap[ismp]] = pams->flags;
if (pams->flags & 0x04) psmp->uFlags |= CHN_16BIT;
if (pams->flags & 0x08) psmp->uFlags |= CHN_LOOP;
if (pams->flags & 0x10) psmp->uFlags |= CHN_PINGPONGLOOP;
}
dwMemPos += sizeof(AMS2SAMPLE);
}
}
if (dwMemPos + 256 >= dwMemLength) return TRUE;
// Comments
{
UINT composernamelen = lpStream[dwMemPos];
if (composernamelen)
{
m_lpszSongComments = new char[composernamelen+1]; // changed from CHAR
if (m_lpszSongComments)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos+1, composernamelen);
m_lpszSongComments[composernamelen] = 0;
}
}
dwMemPos += composernamelen + 1;
// channel names
for (UINT i=0; i<32; i++)
{
UINT chnnamlen = lpStream[dwMemPos];
if ((chnnamlen) && (chnnamlen < MAX_CHANNELNAME))
{
memcpy(ChnSettings[i].szName, lpStream+dwMemPos+1, chnnamlen);
}
dwMemPos += chnnamlen + 1;
if (dwMemPos + chnnamlen + 256 >= dwMemLength) return TRUE;
}
// packed comments (ignored)
UINT songtextlen = *((LPDWORD)(lpStream+dwMemPos));
dwMemPos += songtextlen;
if (dwMemPos + 256 >= dwMemLength) return TRUE;
}
// Order List
{
for (UINT i=0; i<MAX_ORDERS; i++)
{
Order[i] = 0xFF;
if (dwMemPos + 2 >= dwMemLength) return TRUE;
if (i < psh->orders)
{
Order[i] = lpStream[dwMemPos];
dwMemPos += 2;
}
}
}
// Pattern Data
for (UINT ipat=0; ipat<psh->patterns; ipat++)
{
if (dwMemPos+8 >= dwMemLength) return TRUE;
UINT packedlen = *((LPDWORD)(lpStream+dwMemPos));
UINT numrows = 1 + (UINT)(lpStream[dwMemPos+4]);
//UINT patchn = 1 + (UINT)(lpStream[dwMemPos+5] & 0x1F);
//UINT patcmds = 1 + (UINT)(lpStream[dwMemPos+5] >> 5);
UINT patnamlen = lpStream[dwMemPos+6];
dwMemPos += 4;
if ((ipat < MAX_PATTERNS) && (packedlen < dwMemLength-dwMemPos) && (numrows >= 8))
{
if ((patnamlen) && (patnamlen < MAX_PATTERNNAME))
{
char s[MAX_PATTERNNAME]; // changed from CHAR
memcpy(s, lpStream+dwMemPos+3, patnamlen);
s[patnamlen] = 0;
SetPatternName(ipat, s);
}
PatternSize[ipat] = numrows;
Patterns[ipat] = AllocatePattern(numrows, m_nChannels);
if (!Patterns[ipat]) return TRUE;
// Unpack Pattern Data
LPCBYTE psrc = lpStream + dwMemPos;
UINT pos = 3 + patnamlen;
UINT row = 0;
while ((pos < packedlen) && (row < numrows))
{
MODCOMMAND *m = Patterns[ipat] + row * m_nChannels;
UINT byte1 = psrc[pos++];
UINT ch = byte1 & 0x1F;
// Read Note + Instr
if (!(byte1 & 0x40))
{
UINT byte2 = psrc[pos++];
UINT note = byte2 & 0x7F;
if (note) m[ch].note = (note > 1) ? (note-1) : 0xFF;
m[ch].instr = psrc[pos++];
// Read Effect
while (byte2 & 0x80)
{
byte2 = psrc[pos++];
if (byte2 & 0x40)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = byte2 & 0x3F;
} else
{
UINT command = byte2 & 0x3F;
UINT param = psrc[pos++];
if (command == 0x0C)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = param / 2;
} else
if (command < 0x10)
{
m[ch].command = command;
m[ch].param = param;
ConvertModCommand(&m[ch]);
} else
{
// TODO: AMS effects
}
}
}
}
if (byte1 & 0x80) row++;
}
}
dwMemPos += packedlen;
}
// Read Samples
for (UINT iSmp=1; iSmp<=m_nSamples; iSmp++) if (Ins[iSmp].nLength)
{
if (dwMemPos >= dwMemLength - 9) return TRUE;
UINT flags;
if (packedsamples[iSmp] & 0x03)
{
flags = (Ins[iSmp].uFlags & CHN_16BIT) ? RS_AMS16 : RS_AMS8;
} else
{
flags = (Ins[iSmp].uFlags & CHN_16BIT) ? RS_PCM16S : RS_PCM8S;
}
dwMemPos += ReadSample(&Ins[iSmp], flags, (LPSTR)(lpStream+dwMemPos), dwMemLength-dwMemPos);
}
return TRUE;
}
/////////////////////////////////////////////////////////////////////
// AMS Sample unpacking
void AMSUnpack(const char *psrc, UINT inputlen, char *pdest, UINT dmax, char packcharacter)
{
UINT tmplen = dmax;
signed char *amstmp = new signed char[tmplen];
if (!amstmp) return;
// Unpack Loop
{
signed char *p = amstmp;
UINT i=0, j=0;
while ((i < inputlen) && (j < tmplen))
{
signed char ch = psrc[i++];
if (ch == packcharacter)
{
BYTE ch2 = psrc[i++];
if (ch2)
{
ch = psrc[i++];
while (ch2--)
{
p[j++] = ch;
if (j >= tmplen) break;
}
} else p[j++] = packcharacter;
} else p[j++] = ch;
}
}
// Bit Unpack Loop
{
signed char *p = amstmp;
UINT bitcount = 0x80, dh;
UINT k=0;
for (UINT i=0; i<dmax; i++)
{
BYTE al = *p++;
dh = 0;
for (UINT count=0; count<8; count++)
{
UINT bl = al & bitcount;
bl = ((bl|(bl<<8)) >> ((dh+8-count) & 7)) & 0xFF;
bitcount = ((bitcount|(bitcount<<8)) >> 1) & 0xFF;
pdest[k++] |= bl;
if (k >= dmax)
{
k = 0;
dh++;
}
}
bitcount = ((bitcount|(bitcount<<8)) >> dh) & 0xFF;
}
}
// Delta Unpack
{
signed char old = 0;
for (UINT i=0; i<dmax; i++)
{
int pos = ((LPBYTE)pdest)[i];
if ((pos != 128) && (pos & 0x80)) pos = -(pos & 0x7F);
old -= (signed char)pos;
pdest[i] = old;
}
}
delete[] amstmp;
}

@ -0,0 +1,368 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
///////////////////////////////////////////////////////////////
//
// DigiBooster Pro Module Loader (*.dbm)
//
// Note: this loader doesn't handle multiple songs
//
///////////////////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#define DBM_FILE_MAGIC 0x304d4244
#define DBM_ID_NAME 0x454d414e
#define DBM_NAMELEN 0x2c000000
#define DBM_ID_INFO 0x4f464e49
#define DBM_INFOLEN 0x0a000000
#define DBM_ID_SONG 0x474e4f53
#define DBM_ID_INST 0x54534e49
#define DBM_ID_VENV 0x564e4556
#define DBM_ID_PATT 0x54544150
#define DBM_ID_SMPL 0x4c504d53
#pragma pack(1)
typedef struct DBMFILEHEADER
{
DWORD dbm_id; // "DBM0" = 0x304d4244
WORD trkver; // Tracker version: 02.15
WORD reserved;
DWORD name_id; // "NAME" = 0x454d414e
DWORD name_len; // name length: always 44
CHAR songname[44];
DWORD info_id; // "INFO" = 0x4f464e49
DWORD info_len; // 0x0a000000
WORD instruments;
WORD samples;
WORD songs;
WORD patterns;
WORD channels;
DWORD song_id; // "SONG" = 0x474e4f53
DWORD song_len;
CHAR songname2[44];
WORD orders;
// WORD orderlist[0]; // orderlist[orders] in words
} DBMFILEHEADER;
typedef struct DBMINSTRUMENT
{
CHAR name[30];
WORD sampleno;
WORD volume;
DWORD finetune;
DWORD loopstart;
DWORD looplen;
WORD panning;
WORD flags;
} DBMINSTRUMENT;
typedef struct DBMENVELOPE
{
WORD instrument;
BYTE flags;
BYTE numpoints;
BYTE sustain1;
BYTE loopbegin;
BYTE loopend;
BYTE sustain2;
WORD volenv[2*32];
} DBMENVELOPE;
typedef struct DBMPATTERN
{
WORD rows;
DWORD packedsize;
BYTE patterndata[2]; // [packedsize]
} DBMPATTERN;
typedef struct DBMSAMPLE
{
DWORD flags;
DWORD samplesize;
BYTE sampledata[2]; // [samplesize]
} DBMSAMPLE;
#pragma pack()
BOOL CSoundFile::ReadDBM(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const DBMFILEHEADER *pfh = (DBMFILEHEADER *)lpStream;
DWORD dwMemPos;
UINT nOrders, nSamples, nInstruments, nPatterns;
if ((!lpStream) || (dwMemLength <= sizeof(DBMFILEHEADER)) || (!pfh->channels)
|| (pfh->dbm_id != bswapLE32(DBM_FILE_MAGIC)) || (!pfh->songs) || (pfh->song_id != bswapLE32(DBM_ID_SONG))
|| (pfh->name_id != bswapLE32(DBM_ID_NAME)) || (pfh->name_len != bswapLE32(DBM_NAMELEN))
|| (pfh->info_id != bswapLE32(DBM_ID_INFO)) || (pfh->info_len != bswapLE32(DBM_INFOLEN))) return FALSE;
dwMemPos = sizeof(DBMFILEHEADER);
nOrders = bswapBE16(pfh->orders);
if (dwMemPos + 2 * nOrders + 8*3 >= dwMemLength) return FALSE;
nInstruments = bswapBE16(pfh->instruments);
nSamples = bswapBE16(pfh->samples);
nPatterns = bswapBE16(pfh->patterns);
m_nType = MOD_TYPE_DBM;
m_nChannels = bswapBE16(pfh->channels);
if (m_nChannels < 4) m_nChannels = 4;
if (m_nChannels > 64) m_nChannels = 64;
memcpy(m_szNames[0], (pfh->songname[0]) ? pfh->songname : pfh->songname2, 32);
m_szNames[0][31] = 0;
for (UINT iOrd=0; iOrd < nOrders; iOrd++)
{
Order[iOrd] = lpStream[dwMemPos+iOrd*2+1];
if (iOrd >= MAX_ORDERS-2) break;
}
dwMemPos += 2*nOrders;
while (dwMemPos + 10 < dwMemLength)
{
DWORD chunk_id = ((LPDWORD)(lpStream+dwMemPos))[0];
DWORD chunk_size = bswapBE32(((LPDWORD)(lpStream+dwMemPos))[1]);
DWORD chunk_pos;
dwMemPos += 8;
chunk_pos = dwMemPos;
if ((dwMemPos + chunk_size > dwMemLength) || (chunk_size > dwMemLength)) break;
dwMemPos += chunk_size;
// Instruments
if (chunk_id == bswapLE32(DBM_ID_INST))
{
if (nInstruments >= MAX_INSTRUMENTS) nInstruments = MAX_INSTRUMENTS-1;
for (UINT iIns=0; iIns<nInstruments; iIns++)
{
MODINSTRUMENT *psmp;
INSTRUMENTHEADER *penv;
DBMINSTRUMENT *pih;
UINT nsmp;
if (chunk_pos + sizeof(DBMINSTRUMENT) > dwMemPos) break;
if ((penv = new INSTRUMENTHEADER) == NULL) break;
pih = (DBMINSTRUMENT *)(lpStream+chunk_pos);
nsmp = bswapBE16(pih->sampleno);
psmp = ((nsmp) && (nsmp < MAX_SAMPLES)) ? &Ins[nsmp] : NULL;
memset(penv, 0, sizeof(INSTRUMENTHEADER));
memcpy(penv->name, pih->name, 30);
if (psmp)
{
memcpy(m_szNames[nsmp], pih->name, 30);
m_szNames[nsmp][30] = 0;
}
Headers[iIns+1] = penv;
penv->nFadeOut = 1024; // ???
penv->nGlobalVol = 64;
penv->nPan = bswapBE16(pih->panning);
if ((penv->nPan) && (penv->nPan < 256))
penv->dwFlags = ENV_SETPANNING;
else
penv->nPan = 128;
penv->nPPC = 5*12;
for (UINT i=0; i<NOTE_MAX; i++)
{
penv->Keyboard[i] = nsmp;
penv->NoteMap[i] = i+1;
}
// Sample Info
if (psmp)
{
DWORD sflags = bswapBE16(pih->flags);
psmp->nVolume = bswapBE16(pih->volume) * 4;
if ((!psmp->nVolume) || (psmp->nVolume > 256)) psmp->nVolume = 256;
psmp->nGlobalVol = 64;
psmp->nC4Speed = bswapBE32(pih->finetune);
int f2t = FrequencyToTranspose(psmp->nC4Speed);
psmp->RelativeTone = f2t >> 7;
psmp->nFineTune = f2t & 0x7F;
if ((pih->looplen) && (sflags & 3))
{
psmp->nLoopStart = bswapBE32(pih->loopstart);
psmp->nLoopEnd = psmp->nLoopStart + bswapBE32(pih->looplen);
psmp->uFlags |= CHN_LOOP;
psmp->uFlags &= ~CHN_PINGPONGLOOP;
if (sflags & 2) psmp->uFlags |= CHN_PINGPONGLOOP;
}
}
chunk_pos += sizeof(DBMINSTRUMENT);
m_nInstruments = iIns+1;
}
} else
// Volume Envelopes
if (chunk_id == bswapLE32(DBM_ID_VENV))
{
UINT nEnvelopes = lpStream[chunk_pos+1];
chunk_pos += 2;
for (UINT iEnv=0; iEnv<nEnvelopes; iEnv++)
{
DBMENVELOPE *peh;
UINT nins;
if (chunk_pos + sizeof(DBMENVELOPE) > dwMemPos) break;
peh = (DBMENVELOPE *)(lpStream+chunk_pos);
nins = bswapBE16(peh->instrument);
if ((nins) && (nins < MAX_INSTRUMENTS) && (Headers[nins]) && (peh->numpoints))
{
INSTRUMENTHEADER *penv = Headers[nins];
if (peh->flags & 1) penv->dwFlags |= ENV_VOLUME;
if (peh->flags & 2) penv->dwFlags |= ENV_VOLSUSTAIN;
if (peh->flags & 4) penv->dwFlags |= ENV_VOLLOOP;
penv->nVolEnv = peh->numpoints + 1;
if (penv->nVolEnv > MAX_ENVPOINTS) penv->nVolEnv = MAX_ENVPOINTS;
penv->nVolLoopStart = peh->loopbegin;
penv->nVolLoopEnd = peh->loopend;
penv->nVolSustainBegin = penv->nVolSustainEnd = peh->sustain1;
for (UINT i=0; i<penv->nVolEnv; i++)
{
penv->VolPoints[i] = bswapBE16(peh->volenv[i*2]);
penv->VolEnv[i] = (BYTE)bswapBE16(peh->volenv[i*2+1]);
}
}
chunk_pos += sizeof(DBMENVELOPE);
}
} else
// Packed Pattern Data
if (chunk_id == bswapLE32(DBM_ID_PATT))
{
if (nPatterns > MAX_PATTERNS) nPatterns = MAX_PATTERNS;
for (UINT iPat=0; iPat<nPatterns; iPat++)
{
DBMPATTERN *pph;
DWORD pksize;
UINT nRows;
if (chunk_pos + sizeof(DBMPATTERN) > dwMemPos) break;
pph = (DBMPATTERN *)(lpStream+chunk_pos);
pksize = bswapBE32(pph->packedsize);
if ((chunk_pos + pksize + 6 > dwMemPos) || (pksize > dwMemPos)) break;
nRows = bswapBE16(pph->rows);
if ((nRows >= 4) && (nRows <= 256))
{
MODCOMMAND *m = AllocatePattern(nRows, m_nChannels);
if (m)
{
LPBYTE pkdata = (LPBYTE)&pph->patterndata;
UINT row = 0;
UINT i = 0;
PatternSize[iPat] = nRows;
Patterns[iPat] = m;
while ((i+3<pksize) && (row < nRows))
{
UINT ch = pkdata[i++];
if (ch)
{
BYTE b = pkdata[i++];
ch--;
if (ch < m_nChannels)
{
if (b & 0x01)
{
UINT note = pkdata[i++];
if (note == 0x1F) note = 0xFF; else
if ((note) && (note < 0xFE))
{
note = ((note >> 4)*12) + (note & 0x0F) + 13;
}
m[ch].note = note;
}
if (b & 0x02) m[ch].instr = pkdata[i++];
if (b & 0x3C)
{
UINT cmd1 = 0xFF, param1 = 0, cmd2 = 0xFF, param2 = 0;
if (b & 0x04) cmd1 = (UINT)pkdata[i++];
if (b & 0x08) param1 = pkdata[i++];
if (b & 0x10) cmd2 = (UINT)pkdata[i++];
if (b & 0x20) param2 = pkdata[i++];
if (cmd1 == 0x0C)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = param1;
cmd1 = 0xFF;
} else
if (cmd2 == 0x0C)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = param2;
cmd2 = 0xFF;
}
if ((cmd1 > 0x13) || ((cmd1 >= 0x10) && (cmd2 < 0x10)))
{
cmd1 = cmd2;
param1 = param2;
cmd2 = 0xFF;
}
if (cmd1 <= 0x13)
{
m[ch].command = cmd1;
m[ch].param = param1;
ConvertModCommand(&m[ch]);
}
}
} else
{
if (b & 0x01) i++;
if (b & 0x02) i++;
if (b & 0x04) i++;
if (b & 0x08) i++;
if (b & 0x10) i++;
if (b & 0x20) i++;
}
} else
{
row++;
m += m_nChannels;
}
}
}
}
chunk_pos += 6 + pksize;
}
} else
// Reading Sample Data
if (chunk_id == bswapLE32(DBM_ID_SMPL))
{
if (nSamples >= MAX_SAMPLES) nSamples = MAX_SAMPLES-1;
m_nSamples = nSamples;
for (UINT iSmp=1; iSmp<=nSamples; iSmp++)
{
MODINSTRUMENT *pins;
DBMSAMPLE *psh;
DWORD samplesize;
DWORD sampleflags;
if (chunk_pos + sizeof(DBMSAMPLE) >= dwMemPos) break;
psh = (DBMSAMPLE *)(lpStream+chunk_pos);
chunk_pos += 8;
samplesize = bswapBE32(psh->samplesize);
sampleflags = bswapBE32(psh->flags);
pins = &Ins[iSmp];
pins->nLength = samplesize;
if (sampleflags & 2)
{
pins->uFlags |= CHN_16BIT;
samplesize <<= 1;
}
if ((chunk_pos+samplesize > dwMemPos) || (samplesize > dwMemLength)) break;
if (sampleflags & 3)
{
ReadSample(pins, (pins->uFlags & CHN_16BIT) ? RS_PCM16M : RS_PCM8S,
(LPSTR)(psh->sampledata), samplesize);
}
chunk_pos += samplesize;
}
}
}
return TRUE;
}

@ -0,0 +1,608 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
///////////////////////////////////////////////////////
// DMF DELUSION DIGITAL MUSIC FILEFORMAT (X-Tracker) //
///////////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#define DMFLOG
//#pragma warning(disable:4244)
#pragma pack(1)
typedef struct DMFHEADER
{
DWORD id; // "DDMF" = 0x464d4444
BYTE version; // 4
CHAR trackername[8]; // "XTRACKER"
CHAR songname[30];
CHAR composer[20];
BYTE date[3];
} DMFHEADER;
typedef struct DMFINFO
{
DWORD id; // "INFO"
DWORD infosize;
} DMFINFO;
typedef struct DMFSEQU
{
DWORD id; // "SEQU"
DWORD seqsize;
WORD loopstart;
WORD loopend;
WORD sequ[2];
} DMFSEQU;
typedef struct DMFPATT
{
DWORD id; // "PATT"
DWORD patsize;
WORD numpat; // 1-1024
BYTE tracks;
BYTE firstpatinfo;
} DMFPATT;
typedef struct DMFTRACK
{
BYTE tracks;
BYTE beat; // [hi|lo] -> hi=ticks per beat, lo=beats per measure
WORD ticks; // max 512
DWORD jmpsize;
} DMFTRACK;
typedef struct DMFSMPI
{
DWORD id;
DWORD size;
BYTE samples;
} DMFSMPI;
typedef struct DMFSAMPLE
{
DWORD len;
DWORD loopstart;
DWORD loopend;
WORD c3speed;
BYTE volume;
BYTE flags;
} DMFSAMPLE;
#pragma pack()
#ifdef DMFLOG
extern void Log(LPCSTR s, ...);
#endif
BOOL CSoundFile::ReadDMF(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const DMFHEADER *pfh = (DMFHEADER *)lpStream;
DMFINFO *psi;
DMFSEQU *sequ;
DWORD dwMemPos;
BYTE infobyte[32];
BYTE smplflags[MAX_SAMPLES], hasSMPI = 0;
if ((!lpStream) || (dwMemLength < 1024)) return FALSE;
if ((pfh->id != 0x464d4444) || (!pfh->version) || (pfh->version & 0xF0)) return FALSE;
dwMemPos = 66;
memcpy(m_szNames[0], pfh->songname, 30);
m_szNames[0][30] = 0;
m_nType = MOD_TYPE_DMF;
m_nChannels = 0;
#ifdef DMFLOG
Log("DMF version %d: \"%s\": %d bytes (0x%04X)\n", pfh->version, m_szNames[0], dwMemLength, dwMemLength);
#endif
while (dwMemPos + 7 < dwMemLength)
{
DWORD id = *((LPDWORD)(lpStream+dwMemPos));
switch(id)
{
// "INFO"
case 0x4f464e49:
// "CMSG"
case 0x47534d43:
psi = (DMFINFO *)(lpStream+dwMemPos);
if (id == 0x47534d43) dwMemPos++;
if ((psi->infosize > dwMemLength) || (psi->infosize + dwMemPos + 8 > dwMemLength)) goto dmfexit;
if ((psi->infosize >= 8) && (!m_lpszSongComments))
{
m_lpszSongComments = new char[psi->infosize]; // changed from CHAR
if (m_lpszSongComments)
{
for (UINT i=0; i<psi->infosize-1; i++)
{
CHAR c = lpStream[dwMemPos+8+i];
if ((i % 40) == 39)
m_lpszSongComments[i] = 0x0d;
else
m_lpszSongComments[i] = (c < ' ') ? ' ' : c;
}
m_lpszSongComments[psi->infosize-1] = 0;
}
}
dwMemPos += psi->infosize + 8 - 1;
break;
// "SEQU"
case 0x55514553:
sequ = (DMFSEQU *)(lpStream+dwMemPos);
if ((sequ->seqsize >= dwMemLength) || (dwMemPos + sequ->seqsize + 12 > dwMemLength)) goto dmfexit;
{
UINT nseq = sequ->seqsize >> 1;
if (nseq >= MAX_ORDERS-1) nseq = MAX_ORDERS-1;
if (sequ->loopstart < nseq) m_nRestartPos = sequ->loopstart;
for (UINT i=0; i<nseq; i++) Order[i] = (BYTE)sequ->sequ[i];
}
dwMemPos += sequ->seqsize + 8;
break;
// "PATT"
case 0x54544150:
if (!m_nChannels)
{
DMFPATT *patt = (DMFPATT *)(lpStream+dwMemPos);
UINT numpat;
DWORD dwPos = dwMemPos + 11;
if ((patt->patsize >= dwMemLength) || (dwMemPos + patt->patsize + 8 > dwMemLength)) goto dmfexit;
numpat = patt->numpat;
if (numpat > MAX_PATTERNS) numpat = MAX_PATTERNS;
m_nChannels = patt->tracks;
if (m_nChannels < patt->firstpatinfo) m_nChannels = patt->firstpatinfo;
if (m_nChannels > 32) m_nChannels = 32;
if (m_nChannels < 4) m_nChannels = 4;
for (UINT npat=0; npat<numpat; npat++)
{
DMFTRACK *pt = (DMFTRACK *)(lpStream+dwPos);
#ifdef DMFLOG
Log("Pattern #%d: %d tracks, %d rows\n", npat, pt->tracks, pt->ticks);
#endif
UINT tracks = pt->tracks;
if (tracks > 32) tracks = 32;
UINT ticks = pt->ticks;
if (ticks > 256) ticks = 256;
if (ticks < 16) ticks = 16;
dwPos += 8;
if ((pt->jmpsize >= dwMemLength) || (dwPos + pt->jmpsize + 4 >= dwMemLength)) break;
PatternSize[npat] = (WORD)ticks;
MODCOMMAND *m = AllocatePattern(PatternSize[npat], m_nChannels);
if (!m) goto dmfexit;
Patterns[npat] = m;
DWORD d = dwPos;
dwPos += pt->jmpsize;
UINT ttype = 1;
UINT tempo = 125;
UINT glbinfobyte = 0;
UINT pbeat = (pt->beat & 0xf0) ? pt->beat>>4 : 8;
BOOL tempochange = (pt->beat & 0xf0) ? TRUE : FALSE;
memset(infobyte, 0, sizeof(infobyte));
for (UINT row=0; row<ticks; row++)
{
MODCOMMAND *p = &m[row*m_nChannels];
// Parse track global effects
if (!glbinfobyte)
{
BYTE info = lpStream[d++];
BYTE infoval = 0;
if ((info & 0x80) && (d < dwPos)) glbinfobyte = lpStream[d++];
info &= 0x7f;
if ((info) && (d < dwPos)) infoval = lpStream[d++];
switch(info)
{
case 1: ttype = 0; tempo = infoval; tempochange = TRUE; break;
case 2: ttype = 1; tempo = infoval; tempochange = TRUE; break;
case 3: pbeat = infoval>>4; tempochange = ttype; break;
#ifdef DMFLOG
default: if (info) Log("GLB: %02X.%02X\n", info, infoval);
#endif
}
} else
{
glbinfobyte--;
}
// Parse channels
for (UINT i=0; i<tracks; i++) if (!infobyte[i])
{
MODCOMMAND cmd = {0,0,0,0,0,0};
BYTE info = lpStream[d++];
if (info & 0x80) infobyte[i] = lpStream[d++];
// Instrument
if (info & 0x40)
{
cmd.instr = lpStream[d++];
}
// Note
if (info & 0x20)
{
cmd.note = lpStream[d++];
if ((cmd.note) && (cmd.note < 0xfe)) cmd.note &= 0x7f;
if ((cmd.note) && (cmd.note < 128)) cmd.note += 24;
}
// Volume
if (info & 0x10)
{
cmd.volcmd = VOLCMD_VOLUME;
cmd.vol = (lpStream[d++]+3)>>2;
}
// Effect 1
if (info & 0x08)
{
BYTE efx = lpStream[d++];
BYTE eval = lpStream[d++];
switch(efx)
{
// 1: Key Off
case 1: if (!cmd.note) cmd.note = 0xFE; break;
// 2: Set Loop
// 4: Sample Delay
case 4: if (eval&0xe0) { cmd.command = CMD_S3MCMDEX; cmd.param = (eval>>5)|0xD0; } break;
// 5: Retrig
case 5: if (eval&0xe0) { cmd.command = CMD_RETRIG; cmd.param = (eval>>5); } break;
// 6: Offset
case 6: cmd.command = CMD_OFFSET; cmd.param = eval; break;
#ifdef DMFLOG
default: Log("FX1: %02X.%02X\n", efx, eval);
#endif
}
}
// Effect 2
if (info & 0x04)
{
BYTE efx = lpStream[d++];
BYTE eval = lpStream[d++];
switch(efx)
{
// 1: Finetune
case 1: if (eval&0xf0) { cmd.command = CMD_S3MCMDEX; cmd.param = (eval>>4)|0x20; } break;
// 2: Note Delay
case 2: if (eval&0xe0) { cmd.command = CMD_S3MCMDEX; cmd.param = (eval>>5)|0xD0; } break;
// 3: Arpeggio
case 3: if (eval) { cmd.command = CMD_ARPEGGIO; cmd.param = eval; } break;
// 4: Portamento Up
case 4: cmd.command = CMD_PORTAMENTOUP; cmd.param = (eval >= 0xe0) ? 0xdf : eval; break;
// 5: Portamento Down
case 5: cmd.command = CMD_PORTAMENTODOWN; cmd.param = (eval >= 0xe0) ? 0xdf : eval; break;
// 6: Tone Portamento
case 6: cmd.command = CMD_TONEPORTAMENTO; cmd.param = eval; break;
// 8: Vibrato
case 8: cmd.command = CMD_VIBRATO; cmd.param = eval; break;
// 12: Note cut
case 12: if (eval & 0xe0) { cmd.command = CMD_S3MCMDEX; cmd.param = (eval>>5)|0xc0; }
else if (!cmd.note) { cmd.note = 0xfe; } break;
#ifdef DMFLOG
default: Log("FX2: %02X.%02X\n", efx, eval);
#endif
}
}
// Effect 3
if (info & 0x02)
{
BYTE efx = lpStream[d++];
BYTE eval = lpStream[d++];
switch(efx)
{
// 1: Vol Slide Up
case 1: if (eval == 0xff) break;
eval = (eval+3)>>2; if (eval > 0x0f) eval = 0x0f;
cmd.command = CMD_VOLUMESLIDE; cmd.param = eval<<4; break;
// 2: Vol Slide Down
case 2: if (eval == 0xff) break;
eval = (eval+3)>>2; if (eval > 0x0f) eval = 0x0f;
cmd.command = CMD_VOLUMESLIDE; cmd.param = eval; break;
// 7: Set Pan
case 7: if (!cmd.volcmd) { cmd.volcmd = VOLCMD_PANNING; cmd.vol = (eval+3)>>2; }
else { cmd.command = CMD_PANNING8; cmd.param = eval; } break;
// 8: Pan Slide Left
case 8: eval = (eval+3)>>2; if (eval > 0x0f) eval = 0x0f;
cmd.command = CMD_PANNINGSLIDE; cmd.param = eval<<4; break;
// 9: Pan Slide Right
case 9: eval = (eval+3)>>2; if (eval > 0x0f) eval = 0x0f;
cmd.command = CMD_PANNINGSLIDE; cmd.param = eval; break;
#ifdef DMFLOG
default: Log("FX3: %02X.%02X\n", efx, eval);
#endif
}
}
// Store effect
if (i < m_nChannels) p[i] = cmd;
if (d > dwPos)
{
#ifdef DMFLOG
Log("Unexpected EOP: row=%d\n", row);
#endif
break;
}
} else
{
infobyte[i]--;
}
// Find free channel for tempo change
if (tempochange)
{
tempochange = FALSE;
UINT speed=6, modtempo=tempo;
UINT rpm = ((ttype) && (pbeat)) ? tempo*pbeat : (tempo+1)*15;
for (speed=30; speed>1; speed--)
{
modtempo = rpm*speed/24;
if (modtempo <= 200) break;
if ((speed < 6) && (modtempo < 256)) break;
}
#ifdef DMFLOG
Log("Tempo change: ttype=%d pbeat=%d tempo=%3d -> speed=%d tempo=%d\n",
ttype, pbeat, tempo, speed, modtempo);
#endif
for (UINT ich=0; ich<m_nChannels; ich++) if (!p[ich].command)
{
if (speed)
{
p[ich].command = CMD_SPEED;
p[ich].param = (BYTE)speed;
speed = 0;
} else
if ((modtempo >= 32) && (modtempo < 256))
{
p[ich].command = CMD_TEMPO;
p[ich].param = (BYTE)modtempo;
modtempo = 0;
} else
{
break;
}
}
}
if (d >= dwPos) break;
}
#ifdef DMFLOG
Log(" %d/%d bytes remaining\n", dwPos-d, pt->jmpsize);
#endif
if (dwPos + 8 >= dwMemLength) break;
}
dwMemPos += patt->patsize + 8;
}
break;
// "SMPI": Sample Info
case 0x49504d53:
{
hasSMPI = 1;
DMFSMPI *pds = (DMFSMPI *)(lpStream+dwMemPos);
if (pds->size <= dwMemLength - dwMemPos)
{
DWORD dwPos = dwMemPos + 9;
m_nSamples = pds->samples;
if (m_nSamples >= MAX_SAMPLES) m_nSamples = MAX_SAMPLES-1;
for (UINT iSmp=1; iSmp<=m_nSamples; iSmp++)
{
UINT namelen = lpStream[dwPos];
smplflags[iSmp] = 0;
if (dwPos+namelen+1+sizeof(DMFSAMPLE) > dwMemPos+pds->size+8) break;
if (namelen)
{
UINT rlen = (namelen < 32) ? namelen : 31;
memcpy(m_szNames[iSmp], lpStream+dwPos+1, rlen);
m_szNames[iSmp][rlen] = 0;
}
dwPos += namelen + 1;
DMFSAMPLE *psh = (DMFSAMPLE *)(lpStream+dwPos);
MODINSTRUMENT *psmp = &Ins[iSmp];
psmp->nLength = psh->len;
psmp->nLoopStart = psh->loopstart;
psmp->nLoopEnd = psh->loopend;
psmp->nC4Speed = psh->c3speed;
psmp->nGlobalVol = 64;
psmp->nVolume = (psh->volume) ? ((WORD)psh->volume)+1 : (WORD)256;
psmp->uFlags = (psh->flags & 2) ? CHN_16BIT : 0;
if (psmp->uFlags & CHN_16BIT) psmp->nLength >>= 1;
if (psh->flags & 1) psmp->uFlags |= CHN_LOOP;
smplflags[iSmp] = psh->flags;
dwPos += (pfh->version < 8) ? 22 : 30;
#ifdef DMFLOG
Log("SMPI %d/%d: len=%d flags=0x%02X\n", iSmp, m_nSamples, psmp->nLength, psh->flags);
#endif
}
}
dwMemPos += pds->size + 8;
}
break;
// "SMPD": Sample Data
case 0x44504d53:
{
DWORD dwPos = dwMemPos + 8;
UINT ismpd = 0;
for (UINT iSmp=1; iSmp<=m_nSamples; iSmp++)
{
ismpd++;
DWORD pksize;
if (dwPos + 4 >= dwMemLength)
{
#ifdef DMFLOG
Log("Unexpected EOF at sample %d/%d! (pos=%d)\n", iSmp, m_nSamples, dwPos);
#endif
break;
}
pksize = *((LPDWORD)(lpStream+dwPos));
#ifdef DMFLOG
Log("sample %d: pos=0x%X pksize=%d ", iSmp, dwPos, pksize);
Log("len=%d flags=0x%X [%08X]\n", Ins[iSmp].nLength, smplflags[ismpd], *((LPDWORD)(lpStream+dwPos+4)));
#endif
dwPos += 4;
if (pksize > dwMemLength - dwPos)
{
#ifdef DMFLOG
Log("WARNING: pksize=%d, but only %d bytes left\n", pksize, dwMemLength-dwPos);
#endif
pksize = dwMemLength - dwPos;
}
if ((pksize) && (iSmp <= m_nSamples))
{
UINT flags = (Ins[iSmp].uFlags & CHN_16BIT) ? RS_PCM16S : RS_PCM8S;
if (hasSMPI && smplflags[ismpd] & 4)
flags = (Ins[iSmp].uFlags & CHN_16BIT) ? RS_DMF16 : RS_DMF8;
ReadSample(&Ins[iSmp], flags, (LPSTR)(lpStream+dwPos), pksize);
}
dwPos += pksize;
}
dwMemPos = dwPos;
}
break;
// "ENDE": end of file
case 0x45444e45:
goto dmfexit;
// Unrecognized id, or "ENDE" field
default:
dwMemPos += 4;
break;
}
}
dmfexit:
if (!m_nChannels)
{
if (!m_nSamples)
{
m_nType = MOD_TYPE_NONE;
return FALSE;
}
m_nChannels = 4;
}
return TRUE;
}
///////////////////////////////////////////////////////////////////////
// DMF Compression
#pragma pack(1)
typedef struct DMF_HNODE
{
short int left, right;
BYTE value;
} DMF_HNODE;
typedef struct DMF_HTREE
{
LPBYTE ibuf, ibufmax;
DWORD bitbuf;
UINT bitnum;
UINT lastnode, nodecount;
DMF_HNODE nodes[256];
} DMF_HTREE;
#pragma pack()
// DMF Huffman ReadBits
BYTE DMFReadBits(DMF_HTREE *tree, UINT nbits)
//-------------------------------------------
{
BYTE x = 0, bitv = 1;
while (nbits--)
{
if (tree->bitnum)
{
tree->bitnum--;
} else
{
tree->bitbuf = (tree->ibuf < tree->ibufmax) ? *(tree->ibuf++) : 0;
tree->bitnum = 7;
}
if (tree->bitbuf & 1) x |= bitv;
bitv <<= 1;
tree->bitbuf >>= 1;
}
return x;
}
//
// tree: [8-bit value][12-bit index][12-bit index] = 32-bit
//
void DMFNewNode(DMF_HTREE *tree)
//------------------------------
{
BYTE isleft, isright;
UINT actnode;
actnode = tree->nodecount;
if (actnode > 255) return;
tree->nodes[actnode].value = DMFReadBits(tree, 7);
isleft = DMFReadBits(tree, 1);
isright = DMFReadBits(tree, 1);
actnode = tree->lastnode;
if (actnode > 255) return;
tree->nodecount++;
tree->lastnode = tree->nodecount;
if (isleft)
{
tree->nodes[actnode].left = tree->lastnode;
DMFNewNode(tree);
} else
{
tree->nodes[actnode].left = -1;
}
tree->lastnode = tree->nodecount;
if (isright)
{
tree->nodes[actnode].right = tree->lastnode;
DMFNewNode(tree);
} else
{
tree->nodes[actnode].right = -1;
}
}
int DMFUnpack(LPBYTE psample, LPBYTE ibuf, LPBYTE ibufmax, UINT maxlen)
//----------------------------------------------------------------------
{
DMF_HTREE tree;
UINT actnode;
BYTE value, sign, delta = 0;
memset(&tree, 0, sizeof(tree));
tree.ibuf = ibuf;
tree.ibufmax = ibufmax;
DMFNewNode(&tree);
value = 0;
for (UINT i=0; i<maxlen; i++)
{
actnode = 0;
sign = DMFReadBits(&tree, 1);
do
{
if (DMFReadBits(&tree, 1))
actnode = tree.nodes[actnode].right;
else
actnode = tree.nodes[actnode].left;
if (actnode > 255) break;
delta = tree.nodes[actnode].value;
if ((tree.ibuf >= tree.ibufmax) && (!tree.bitnum)) break;
} while ((tree.nodes[actnode].left >= 0) && (tree.nodes[actnode].right >= 0));
if (sign) delta ^= 0xFF;
value += delta;
psample[i] = (i) ? value : 0;
}
#ifdef DMFLOG
// Log("DMFUnpack: %d remaining bytes\n", tree.ibufmax-tree.ibuf);
#endif
return tree.ibuf - ibuf;
}

@ -0,0 +1,236 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
//////////////////////////////////////////////
// DSIK Internal Format (DSM) module loader //
//////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
#pragma pack(1)
#define DSMID_RIFF 0x46464952 // "RIFF"
#define DSMID_DSMF 0x464d5344 // "DSMF"
#define DSMID_SONG 0x474e4f53 // "SONG"
#define DSMID_INST 0x54534e49 // "INST"
#define DSMID_PATT 0x54544150 // "PATT"
typedef struct DSMNOTE
{
BYTE note,ins,vol,cmd,inf;
} DSMNOTE;
typedef struct DSMINST
{
DWORD id_INST;
DWORD inst_len;
CHAR filename[13];
BYTE flags;
BYTE flags2;
BYTE volume;
DWORD length;
DWORD loopstart;
DWORD loopend;
DWORD reserved1;
WORD c2spd;
WORD reserved2;
CHAR samplename[28];
} DSMINST;
typedef struct DSMFILEHEADER
{
DWORD id_RIFF; // "RIFF"
DWORD riff_len;
DWORD id_DSMF; // "DSMF"
DWORD id_SONG; // "SONG"
DWORD song_len;
} DSMFILEHEADER;
typedef struct DSMSONG
{
CHAR songname[28];
WORD reserved1;
WORD flags;
DWORD reserved2;
WORD numord;
WORD numsmp;
WORD numpat;
WORD numtrk;
BYTE globalvol;
BYTE mastervol;
BYTE speed;
BYTE bpm;
BYTE panpos[16];
BYTE orders[128];
} DSMSONG;
typedef struct DSMPATT
{
DWORD id_PATT;
DWORD patt_len;
BYTE dummy1;
BYTE dummy2;
} DSMPATT;
#pragma pack()
BOOL CSoundFile::ReadDSM(LPCBYTE lpStream, DWORD dwMemLength)
//-----------------------------------------------------------
{
DSMFILEHEADER *pfh = (DSMFILEHEADER *)lpStream;
DSMSONG *psong;
DWORD dwMemPos;
UINT nPat, nSmp;
if ((!lpStream) || (dwMemLength < 1024) || (pfh->id_RIFF != DSMID_RIFF)
|| (pfh->riff_len + 8 > dwMemLength) || (pfh->riff_len < 1024)
|| (pfh->id_DSMF != DSMID_DSMF) || (pfh->id_SONG != DSMID_SONG)
|| (pfh->song_len > dwMemLength)) return FALSE;
psong = (DSMSONG *)(lpStream + sizeof(DSMFILEHEADER));
dwMemPos = sizeof(DSMFILEHEADER) + pfh->song_len;
m_nType = MOD_TYPE_DSM;
m_nChannels = psong->numtrk;
if (m_nChannels < 4) m_nChannels = 4;
if (m_nChannels > 16) m_nChannels = 16;
m_nSamples = psong->numsmp;
if (m_nSamples >= MAX_SAMPLES) m_nSamples = MAX_SAMPLES - 1;
m_nDefaultSpeed = psong->speed;
m_nDefaultTempo = psong->bpm;
m_nDefaultGlobalVolume = psong->globalvol << 2;
if ((!m_nDefaultGlobalVolume) || (m_nDefaultGlobalVolume > 256)) m_nDefaultGlobalVolume = 256;
m_nSongPreAmp = psong->mastervol & 0x7F;
for (UINT iOrd=0; iOrd<sizeof(psong->orders); iOrd++)
{
Order[iOrd] = (BYTE)((iOrd < psong->numord) ? psong->orders[iOrd] : 0xFF);
}
for (UINT iPan=0; iPan<16; iPan++)
{
ChnSettings[iPan].nPan = 0x80;
if (psong->panpos[iPan] <= 0x80)
{
ChnSettings[iPan].nPan = psong->panpos[iPan] << 1;
}
}
memcpy(m_szNames[0], psong->songname, 28);
nPat = 0;
nSmp = 1;
while (dwMemPos < dwMemLength - 8)
{
DSMPATT *ppatt = (DSMPATT *)(lpStream + dwMemPos);
DSMINST *pins = (DSMINST *)(lpStream+dwMemPos);
// Reading Patterns
if (ppatt->id_PATT == DSMID_PATT)
{
dwMemPos += 8;
if (dwMemPos + ppatt->patt_len >= dwMemLength) break;
DWORD dwPos = dwMemPos;
dwMemPos += ppatt->patt_len;
MODCOMMAND *m = AllocatePattern(64, m_nChannels);
if (!m) break;
PatternSize[nPat] = 64;
Patterns[nPat] = m;
UINT row = 0;
while ((row < 64) && (dwPos + 2 <= dwMemPos))
{
UINT flag = lpStream[dwPos++];
if (flag)
{
UINT ch = (flag & 0x0F) % m_nChannels;
if (flag & 0x80)
{
UINT note = lpStream[dwPos++];
if (note)
{
if (note <= 12*9) note += 12;
m[ch].note = (BYTE)note;
}
}
if (flag & 0x40)
{
m[ch].instr = lpStream[dwPos++];
}
if (flag & 0x20)
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = lpStream[dwPos++];
}
if (flag & 0x10)
{
UINT command = lpStream[dwPos++];
UINT param = lpStream[dwPos++];
switch(command)
{
// 4-bit Panning
case 0x08:
switch(param & 0xF0)
{
case 0x00: param <<= 4; break;
case 0x10: command = 0x0A; param = (param & 0x0F) << 4; break;
case 0x20: command = 0x0E; param = (param & 0x0F) | 0xA0; break;
case 0x30: command = 0x0E; param = (param & 0x0F) | 0x10; break;
case 0x40: command = 0x0E; param = (param & 0x0F) | 0x20; break;
default: command = 0;
}
break;
// Portamentos
case 0x11:
case 0x12:
command &= 0x0F;
break;
// 3D Sound (?)
case 0x13:
command = 'X' - 55;
param = 0x91;
break;
default:
// Volume + Offset (?)
command = ((command & 0xF0) == 0x20) ? 0x09 : 0;
}
m[ch].command = (BYTE)command;
m[ch].param = (BYTE)param;
if (command) ConvertModCommand(&m[ch]);
}
} else
{
m += m_nChannels;
row++;
}
}
nPat++;
} else
// Reading Samples
if ((nSmp <= m_nSamples) && (pins->id_INST == DSMID_INST))
{
if (dwMemPos + pins->inst_len >= dwMemLength - 8) break;
DWORD dwPos = dwMemPos + sizeof(DSMINST);
dwMemPos += 8 + pins->inst_len;
memcpy(m_szNames[nSmp], pins->samplename, 28);
MODINSTRUMENT *psmp = &Ins[nSmp];
memcpy(psmp->name, pins->filename, 13);
psmp->nGlobalVol = 64;
psmp->nC4Speed = pins->c2spd;
psmp->uFlags = (WORD)((pins->flags & 1) ? CHN_LOOP : 0);
psmp->nLength = pins->length;
psmp->nLoopStart = pins->loopstart;
psmp->nLoopEnd = pins->loopend;
psmp->nVolume = (WORD)(pins->volume << 2);
if (psmp->nVolume > 256) psmp->nVolume = 256;
UINT smptype = (pins->flags & 2) ? RS_PCM8S : RS_PCM8U;
ReadSample(psmp, smptype, (LPCSTR)(lpStream+dwPos), dwMemLength - dwPos);
nSmp++;
} else
{
break;
}
}
return TRUE;
}

@ -0,0 +1,265 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
////////////////////////////////////////
// Farandole (FAR) module loader //
////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#define FARFILEMAGIC 0xFE524146 // "FAR"
#pragma pack(1)
typedef struct FARHEADER1
{
DWORD id; // file magic FAR=
CHAR songname[40]; // songname
CHAR magic2[3]; // 13,10,26
WORD headerlen; // remaining length of header in bytes
BYTE version; // 0xD1
BYTE onoff[16];
BYTE edit1[9];
BYTE speed;
BYTE panning[16];
BYTE edit2[4];
WORD stlen;
} FARHEADER1;
typedef struct FARHEADER2
{
BYTE orders[256];
BYTE numpat;
BYTE snglen;
BYTE loopto;
WORD patsiz[256];
} FARHEADER2;
typedef struct FARSAMPLE
{
CHAR samplename[32];
DWORD length;
BYTE finetune;
BYTE volume;
DWORD reppos;
DWORD repend;
BYTE type;
BYTE loop;
} FARSAMPLE;
#pragma pack()
BOOL CSoundFile::ReadFAR(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const FARHEADER1 *pmh1 = (const FARHEADER1 *)lpStream;
const FARHEADER2 *pmh2;
DWORD dwMemPos = sizeof(FARHEADER1);
UINT headerlen, stlen;
BYTE samplemap[8];
if ((!lpStream) || (dwMemLength < 1024) || (bswapLE32(pmh1->id) != FARFILEMAGIC)
|| (pmh1->magic2[0] != 13) || (pmh1->magic2[1] != 10) || (pmh1->magic2[2] != 26)) return FALSE;
headerlen = bswapLE16(pmh1->headerlen);
stlen = bswapLE16( pmh1->stlen );
if ((headerlen >= dwMemLength) || (dwMemPos + stlen + sizeof(FARHEADER2) >= dwMemLength)) return FALSE;
// Globals
m_nType = MOD_TYPE_FAR;
m_nChannels = 16;
m_nInstruments = 0;
m_nSamples = 0;
m_nSongPreAmp = 0x20;
m_nDefaultSpeed = pmh1->speed;
m_nDefaultTempo = 80;
m_nDefaultGlobalVolume = 256;
memcpy(m_szNames[0], pmh1->songname, 32);
// Channel Setting
for (UINT nchpan=0; nchpan<16; nchpan++)
{
ChnSettings[nchpan].dwFlags = 0;
ChnSettings[nchpan].nPan = ((pmh1->panning[nchpan] & 0x0F) << 4) + 8;
ChnSettings[nchpan].nVolume = 64;
}
// Reading comment
if (stlen)
{
UINT szLen = stlen;
if (szLen > dwMemLength - dwMemPos) szLen = dwMemLength - dwMemPos;
if ((m_lpszSongComments = new char[szLen + 1]) != NULL)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos, szLen);
m_lpszSongComments[szLen] = 0;
}
dwMemPos += stlen;
}
// Reading orders
if (sizeof(FARHEADER2) > dwMemLength - dwMemPos) return TRUE;
pmh2 = (const FARHEADER2 *)(lpStream + dwMemPos);
dwMemPos += sizeof(FARHEADER2);
if (dwMemPos >= dwMemLength) return TRUE;
for (UINT iorder=0; iorder<MAX_ORDERS; iorder++)
{
Order[iorder] = (iorder <= pmh2->snglen) ? pmh2->orders[iorder] : 0xFF;
}
m_nRestartPos = pmh2->loopto;
// Reading Patterns
dwMemPos += headerlen - (869 + stlen);
if (dwMemPos >= dwMemLength) return TRUE;
// end byteswap of pattern data
WORD *patsiz = (WORD *)pmh2->patsiz;
for (UINT ipat=0; ipat<256; ipat++) if (patsiz[ipat])
{
UINT patlen = bswapLE16(patsiz[ipat]);
if ((ipat >= MAX_PATTERNS) || (patlen < 2))
{
dwMemPos += patlen;
continue;
}
if (dwMemPos + patlen >= dwMemLength) return TRUE;
UINT rows = (patlen - 2) >> 6;
if (!rows)
{
dwMemPos += patlen;
continue;
}
if (rows > 256) rows = 256;
if (rows < 16) rows = 16;
PatternSize[ipat] = rows;
if ((Patterns[ipat] = AllocatePattern(rows, m_nChannels)) == NULL) return TRUE;
MODCOMMAND *m = Patterns[ipat];
UINT patbrk = lpStream[dwMemPos];
const BYTE *p = lpStream + dwMemPos + 2;
UINT max = rows*16*4;
if (max > patlen-2) max = patlen-2;
for (UINT len=0; len<max; len += 4, m++)
{
BYTE note = p[len];
BYTE ins = p[len+1];
BYTE vol = p[len+2];
BYTE eff = p[len+3];
if (note)
{
m->instr = ins + 1;
m->note = note + 36;
}
if (vol & 0x0F)
{
m->volcmd = VOLCMD_VOLUME;
m->vol = (vol & 0x0F) << 2;
if (m->vol <= 4) m->vol = 0;
}
switch(eff & 0xF0)
{
// 1.x: Portamento Up
case 0x10:
m->command = CMD_PORTAMENTOUP;
m->param = eff & 0x0F;
break;
// 2.x: Portamento Down
case 0x20:
m->command = CMD_PORTAMENTODOWN;
m->param = eff & 0x0F;
break;
// 3.x: Tone-Portamento
case 0x30:
m->command = CMD_TONEPORTAMENTO;
m->param = (eff & 0x0F) << 2;
break;
// 4.x: Retrigger
case 0x40:
m->command = CMD_RETRIG;
m->param = 6 / (1+(eff&0x0F)) + 1;
break;
// 5.x: Set Vibrato Depth
case 0x50:
m->command = CMD_VIBRATO;
m->param = (eff & 0x0F);
break;
// 6.x: Set Vibrato Speed
case 0x60:
m->command = CMD_VIBRATO;
m->param = (eff & 0x0F) << 4;
break;
// 7.x: Vol Slide Up
case 0x70:
m->command = CMD_VOLUMESLIDE;
m->param = (eff & 0x0F) << 4;
break;
// 8.x: Vol Slide Down
case 0x80:
m->command = CMD_VOLUMESLIDE;
m->param = (eff & 0x0F);
break;
// A.x: Port to vol
case 0xA0:
m->volcmd = VOLCMD_VOLUME;
m->vol = ((eff & 0x0F) << 2) + 4;
break;
// B.x: Set Balance
case 0xB0:
m->command = CMD_PANNING8;
m->param = (eff & 0x0F) << 4;
break;
// F.x: Set Speed
case 0xF0:
m->command = CMD_SPEED;
m->param = eff & 0x0F;
break;
default:
if ((patbrk) && (patbrk+1 == (len >> 6)) && (patbrk+1 != rows-1))
{
m->command = CMD_PATTERNBREAK;
patbrk = 0;
}
}
}
dwMemPos += patlen;
}
// Reading samples
if (dwMemPos + 8 >= dwMemLength) return TRUE;
memcpy(samplemap, lpStream+dwMemPos, 8);
dwMemPos += 8;
MODINSTRUMENT *pins = &Ins[1];
for (UINT ismp=0; ismp<64; ismp++, pins++) if (samplemap[ismp >> 3] & (1 << (ismp & 7)))
{
if (dwMemPos + sizeof(FARSAMPLE) > dwMemLength) return TRUE;
const FARSAMPLE *pfs = reinterpret_cast<const FARSAMPLE*>(lpStream + dwMemPos);
dwMemPos += sizeof(FARSAMPLE);
m_nSamples = ismp + 1;
memcpy(m_szNames[ismp+1], pfs->samplename, 32);
const DWORD length = bswapLE32( pfs->length ) ; /* endian fix - Toad */
pins->nLength = length ;
pins->nLoopStart = bswapLE32(pfs->reppos) ;
pins->nLoopEnd = bswapLE32(pfs->repend) ;
pins->nFineTune = 0;
pins->nC4Speed = 8363*2;
pins->nGlobalVol = 64;
pins->nVolume = pfs->volume << 4;
pins->uFlags = 0;
if ((pins->nLength > 3) && (dwMemPos + 4 < dwMemLength))
{
if (pfs->type & 1)
{
pins->uFlags |= CHN_16BIT;
pins->nLength >>= 1;
pins->nLoopStart >>= 1;
pins->nLoopEnd >>= 1;
}
if ((pfs->loop & 8) && (pins->nLoopEnd > 4)) pins->uFlags |= CHN_LOOP;
ReadSample(pins, (pins->uFlags & CHN_16BIT) ? RS_PCM16S : RS_PCM8S,
(LPSTR)(lpStream+dwMemPos), dwMemLength - dwMemPos);
}
dwMemPos += length;
}
return TRUE;
}

File diff suppressed because it is too large Load Diff

@ -0,0 +1,15 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
///////////////////////////////////////////////////
//
// J2B module loader
//
///////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"

@ -0,0 +1,503 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
//////////////////////////////////////////////
// DigiTracker (MDL) module loader //
//////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
typedef struct MDLSONGHEADER
{
DWORD id; // "DMDL" = 0x4C444D44
BYTE version;
} MDLSONGHEADER;
typedef struct MDLINFOBLOCK
{
CHAR songname[32];
CHAR composer[20];
WORD norders;
WORD repeatpos;
BYTE globalvol;
BYTE speed;
BYTE tempo;
BYTE channelinfo[32];
BYTE seq[256];
} MDLINFOBLOCK;
typedef struct MDLPATTERNDATA
{
BYTE channels;
BYTE lastrow; // nrows = lastrow+1
CHAR name[16];
WORD data[1];
} MDLPATTERNDATA;
void ConvertMDLCommand(MODCOMMAND *m, UINT eff, UINT data)
//--------------------------------------------------------
{
UINT command = 0, param = data;
switch(eff)
{
case 0x01: command = CMD_PORTAMENTOUP; break;
case 0x02: command = CMD_PORTAMENTODOWN; break;
case 0x03: command = CMD_TONEPORTAMENTO; break;
case 0x04: command = CMD_VIBRATO; break;
case 0x05: command = CMD_ARPEGGIO; break;
case 0x07: command = (param < 0x20) ? CMD_SPEED : CMD_TEMPO; break;
case 0x08: command = CMD_PANNING8; param <<= 1; break;
case 0x0B: command = CMD_POSITIONJUMP; break;
case 0x0C: command = CMD_GLOBALVOLUME; break;
case 0x0D: command = CMD_PATTERNBREAK; param = (data & 0x0F) + (data>>4)*10; break;
case 0x0E:
command = CMD_S3MCMDEX;
switch(data & 0xF0)
{
case 0x00: command = 0; break; // What is E0x in MDL (there is a bunch) ?
case 0x10: if (param & 0x0F) { param |= 0xF0; command = CMD_PANNINGSLIDE; } else command = 0; break;
case 0x20: if (param & 0x0F) { param = (param << 4) | 0x0F; command = CMD_PANNINGSLIDE; } else command = 0; break;
case 0x30: param = (data & 0x0F) | 0x10; break; // glissando
case 0x40: param = (data & 0x0F) | 0x30; break; // vibrato waveform
case 0x60: param = (data & 0x0F) | 0xB0; break;
case 0x70: param = (data & 0x0F) | 0x40; break; // tremolo waveform
case 0x90: command = CMD_RETRIG; param &= 0x0F; break;
case 0xA0: param = (data & 0x0F) << 4; command = CMD_GLOBALVOLSLIDE; break;
case 0xB0: param = data & 0x0F; command = CMD_GLOBALVOLSLIDE; break;
case 0xF0: param = ((data >> 8) & 0x0F) | 0xA0; break;
}
break;
case 0x0F: command = CMD_SPEED; break;
case 0x10: if ((param & 0xF0) != 0xE0) { command = CMD_VOLUMESLIDE; if ((param & 0xF0) == 0xF0) param = ((param << 4) | 0x0F); else param >>= 2; } break;
case 0x20: if ((param & 0xF0) != 0xE0) { command = CMD_VOLUMESLIDE; if ((param & 0xF0) != 0xF0) param >>= 2; } break;
case 0x30: command = CMD_RETRIG; break;
case 0x40: command = CMD_TREMOLO; break;
case 0x50: command = CMD_TREMOR; break;
case 0xEF: if (param > 0xFF) param = 0xFF; command = CMD_OFFSET; break;
}
if (command)
{
m->command = command;
m->param = param;
}
}
void UnpackMDLTrack(MODCOMMAND *pat, UINT nChannels, UINT nRows, UINT nTrack, const BYTE *lpTracks)
//-------------------------------------------------------------------------------------------------
{
MODCOMMAND cmd, *m = pat;
UINT len = *((WORD *)lpTracks);
UINT pos = 0, row = 0, i;
lpTracks += 2;
for (UINT ntrk=1; ntrk<nTrack; ntrk++)
{
lpTracks += len;
len = *((WORD *)lpTracks);
lpTracks += 2;
}
cmd.note = cmd.instr = 0;
cmd.volcmd = cmd.vol = 0;
cmd.command = cmd.param = 0;
while ((row < nRows) && (pos < len))
{
UINT xx;
BYTE b = lpTracks[pos++];
xx = b >> 2;
switch(b & 0x03)
{
case 0x01:
for (i=0; i<=xx; i++)
{
if (row) *m = *(m-nChannels);
m += nChannels;
row++;
if (row >= nRows) break;
}
break;
case 0x02:
if (xx < row) *m = pat[nChannels*xx];
m += nChannels;
row++;
break;
case 0x03:
{
cmd.note = (xx & 0x01) ? lpTracks[pos++] : 0;
cmd.instr = (xx & 0x02) ? lpTracks[pos++] : 0;
cmd.volcmd = cmd.vol = 0;
cmd.command = cmd.param = 0;
if ((cmd.note < NOTE_MAX-12) && (cmd.note)) cmd.note += 12;
UINT volume = (xx & 0x04) ? lpTracks[pos++] : 0;
UINT commands = (xx & 0x08) ? lpTracks[pos++] : 0;
UINT command1 = commands & 0x0F;
UINT command2 = commands & 0xF0;
UINT param1 = (xx & 0x10) ? lpTracks[pos++] : 0;
UINT param2 = (xx & 0x20) ? lpTracks[pos++] : 0;
if ((command1 == 0x0E) && ((param1 & 0xF0) == 0xF0) && (!command2))
{
param1 = ((param1 & 0x0F) << 8) | param2;
command1 = 0xEF;
command2 = param2 = 0;
}
if (volume)
{
cmd.volcmd = VOLCMD_VOLUME;
cmd.vol = (volume+1) >> 2;
}
ConvertMDLCommand(&cmd, command1, param1);
if ((cmd.command != CMD_SPEED)
&& (cmd.command != CMD_TEMPO)
&& (cmd.command != CMD_PATTERNBREAK))
ConvertMDLCommand(&cmd, command2, param2);
*m = cmd;
m += nChannels;
row++;
}
break;
// Empty Slots
default:
row += xx+1;
m += (xx+1)*nChannels;
if (row >= nRows) break;
}
}
}
BOOL CSoundFile::ReadMDL(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
DWORD dwMemPos, dwPos, blocklen, dwTrackPos;
const MDLSONGHEADER *pmsh = (const MDLSONGHEADER *)lpStream;
const MDLINFOBLOCK *pmib;
const MDLPATTERNDATA *pmpd;
UINT i,j, norders = 0, npatterns = 0, ntracks = 0;
UINT ninstruments = 0, nsamples = 0;
WORD block;
WORD patterntracks[MAX_PATTERNS*32];
BYTE smpinfo[MAX_SAMPLES];
BYTE insvolenv[MAX_INSTRUMENTS];
BYTE inspanenv[MAX_INSTRUMENTS];
LPCBYTE pvolenv, ppanenv, ppitchenv;
UINT nvolenv, npanenv, npitchenv;
if ((!lpStream) || (dwMemLength < 1024)) return FALSE;
if ((pmsh->id != 0x4C444D44) || ((pmsh->version & 0xF0) > 0x10)) return FALSE;
memset(patterntracks, 0, sizeof(patterntracks));
memset(smpinfo, 0, sizeof(smpinfo));
memset(insvolenv, 0, sizeof(insvolenv));
memset(inspanenv, 0, sizeof(inspanenv));
dwMemPos = 5;
dwTrackPos = 0;
pvolenv = ppanenv = ppitchenv = NULL;
nvolenv = npanenv = npitchenv = 0;
m_nSamples = m_nInstruments = 0;
while (dwMemPos+6 < dwMemLength)
{
block = *((WORD *)(lpStream+dwMemPos));
blocklen = *((DWORD *)(lpStream+dwMemPos+2));
dwMemPos += 6;
if (dwMemPos + blocklen > dwMemLength)
{
if (dwMemPos == 11) return FALSE;
break;
}
switch(block)
{
// IN: infoblock
case 0x4E49:
pmib = (MDLINFOBLOCK *)(lpStream+dwMemPos);
memcpy(m_szNames[0], pmib->songname, 32);
norders = pmib->norders;
if (norders > MAX_ORDERS) norders = MAX_ORDERS;
m_nRestartPos = pmib->repeatpos;
m_nDefaultGlobalVolume = pmib->globalvol;
m_nDefaultTempo = pmib->tempo;
m_nDefaultSpeed = pmib->speed;
m_nChannels = 4;
for (i=0; i<32; i++)
{
ChnSettings[i].nVolume = 64;
ChnSettings[i].nPan = (pmib->channelinfo[i] & 0x7F) << 1;
if (pmib->channelinfo[i] & 0x80)
ChnSettings[i].dwFlags |= CHN_MUTE;
else
m_nChannels = i+1;
}
for (j=0; j<norders; j++) Order[j] = pmib->seq[j];
break;
// ME: song message
case 0x454D:
if (blocklen)
{
if (m_lpszSongComments) delete [] m_lpszSongComments;
m_lpszSongComments = new char[blocklen];
if (m_lpszSongComments)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos, blocklen);
m_lpszSongComments[blocklen-1] = 0;
}
}
break;
// PA: Pattern Data
case 0x4150:
npatterns = lpStream[dwMemPos];
if (npatterns > MAX_PATTERNS) npatterns = MAX_PATTERNS;
dwPos = dwMemPos + 1;
for (i=0; i<npatterns; i++)
{
if (dwPos+18 >= dwMemLength) break;
pmpd = (MDLPATTERNDATA *)(lpStream + dwPos);
if (pmpd->channels > 32) break;
PatternSize[i] = pmpd->lastrow+1;
if (m_nChannels < pmpd->channels) m_nChannels = pmpd->channels;
dwPos += 18 + 2*pmpd->channels;
for (j=0; j<pmpd->channels; j++)
{
patterntracks[i*32+j] = pmpd->data[j];
}
}
break;
// TR: Track Data
case 0x5254:
if (dwTrackPos) break;
ntracks = *((WORD *)(lpStream+dwMemPos));
dwTrackPos = dwMemPos+2;
break;
// II: Instruments
case 0x4949:
ninstruments = lpStream[dwMemPos];
dwPos = dwMemPos+1;
for (i=0; i<ninstruments; i++)
{
UINT nins = lpStream[dwPos];
if ((nins >= MAX_INSTRUMENTS) || (!nins)) break;
if (m_nInstruments < nins) m_nInstruments = nins;
if (!Headers[nins])
{
UINT note = 12;
if ((Headers[nins] = new INSTRUMENTHEADER) == NULL) break;
INSTRUMENTHEADER *penv = Headers[nins];
memset(penv, 0, sizeof(INSTRUMENTHEADER));
memcpy(penv->name, lpStream+dwPos+2, 32);
penv->nGlobalVol = 64;
penv->nPPC = 5*12;
for (j=0; j<lpStream[dwPos+1]; j++)
{
const BYTE *ps = lpStream+dwPos+34+14*j;
while ((note < (UINT)(ps[1]+12)) && (note < NOTE_MAX))
{
penv->NoteMap[note] = note+1;
if (ps[0] < MAX_SAMPLES)
{
int ismp = ps[0];
penv->Keyboard[note] = ps[0];
Ins[ismp].nVolume = ps[2];
Ins[ismp].nPan = ps[4] << 1;
Ins[ismp].nVibType = ps[11];
Ins[ismp].nVibSweep = ps[10];
Ins[ismp].nVibDepth = ps[9];
Ins[ismp].nVibRate = ps[8];
}
penv->nFadeOut = (ps[7] << 8) | ps[6];
if (penv->nFadeOut == 0xFFFF) penv->nFadeOut = 0;
note++;
}
// Use volume envelope ?
if (ps[3] & 0x80)
{
penv->dwFlags |= ENV_VOLUME;
insvolenv[nins] = (ps[3] & 0x3F) + 1;
}
// Use panning envelope ?
if (ps[5] & 0x80)
{
penv->dwFlags |= ENV_PANNING;
inspanenv[nins] = (ps[5] & 0x3F) + 1;
}
}
}
dwPos += 34 + 14*lpStream[dwPos+1];
}
for (j=1; j<=m_nInstruments; j++) if (!Headers[j])
{
Headers[j] = new INSTRUMENTHEADER;
if (Headers[j]) memset(Headers[j], 0, sizeof(INSTRUMENTHEADER));
}
break;
// VE: Volume Envelope
case 0x4556:
if ((nvolenv = lpStream[dwMemPos]) == 0) break;
if (dwMemPos + nvolenv*32 + 1 <= dwMemLength) pvolenv = lpStream + dwMemPos + 1;
break;
// PE: Panning Envelope
case 0x4550:
if ((npanenv = lpStream[dwMemPos]) == 0) break;
if (dwMemPos + npanenv*32 + 1 <= dwMemLength) ppanenv = lpStream + dwMemPos + 1;
break;
// FE: Pitch Envelope
case 0x4546:
if ((npitchenv = lpStream[dwMemPos]) == 0) break;
if (dwMemPos + npitchenv*32 + 1 <= dwMemLength) ppitchenv = lpStream + dwMemPos + 1;
break;
// IS: Sample Infoblock
case 0x5349:
nsamples = lpStream[dwMemPos];
dwPos = dwMemPos+1;
for (i=0; i<nsamples; i++, dwPos += 59)
{
UINT nins = lpStream[dwPos];
if ((nins >= MAX_SAMPLES) || (!nins)) continue;
if (m_nSamples < nins) m_nSamples = nins;
MODINSTRUMENT *pins = &Ins[nins];
memcpy(m_szNames[nins], lpStream+dwPos+1, 32);
memcpy(pins->name, lpStream+dwPos+33, 8);
pins->nC4Speed = *((DWORD *)(lpStream+dwPos+41));
pins->nLength = *((DWORD *)(lpStream+dwPos+45));
pins->nLoopStart = *((DWORD *)(lpStream+dwPos+49));
pins->nLoopEnd = pins->nLoopStart + *((DWORD *)(lpStream+dwPos+53));
if (pins->nLoopEnd > pins->nLoopStart) pins->uFlags |= CHN_LOOP;
pins->nGlobalVol = 64;
if (lpStream[dwPos+58] & 0x01)
{
pins->uFlags |= CHN_16BIT;
pins->nLength >>= 1;
pins->nLoopStart >>= 1;
pins->nLoopEnd >>= 1;
}
if (lpStream[dwPos+58] & 0x02) pins->uFlags |= CHN_PINGPONGLOOP;
smpinfo[nins] = (lpStream[dwPos+58] >> 2) & 3;
}
break;
// SA: Sample Data
case 0x4153:
dwPos = dwMemPos;
for (i=1; i<=m_nSamples; i++) if ((Ins[i].nLength) && (!Ins[i].pSample) && (smpinfo[i] != 3) && (dwPos < dwMemLength))
{
MODINSTRUMENT *pins = &Ins[i];
UINT flags = (pins->uFlags & CHN_16BIT) ? RS_PCM16S : RS_PCM8S;
if (!smpinfo[i])
{
dwPos += ReadSample(pins, flags, (LPSTR)(lpStream+dwPos), dwMemLength - dwPos);
} else
{
DWORD dwLen = *((DWORD *)(lpStream+dwPos));
dwPos += 4;
if ((dwLen < dwMemLength) && (dwLen <= dwMemLength - dwPos) && (dwLen > 4))
{
flags = (pins->uFlags & CHN_16BIT) ? RS_MDL16 : RS_MDL8;
ReadSample(pins, flags, (LPSTR)(lpStream+dwPos), dwLen);
}
dwPos += dwLen;
}
}
break;
}
dwMemPos += blocklen;
}
// Unpack Patterns
if ((dwTrackPos) && (npatterns) && (m_nChannels) && (ntracks))
{
for (UINT ipat=0; ipat<npatterns; ipat++)
{
if ((Patterns[ipat] = AllocatePattern(PatternSize[ipat], m_nChannels)) == NULL) break;
for (UINT chn=0; chn<m_nChannels; chn++) if ((patterntracks[ipat*32+chn]) && (patterntracks[ipat*32+chn] <= ntracks))
{
MODCOMMAND *m = Patterns[ipat] + chn;
UnpackMDLTrack(m, m_nChannels, PatternSize[ipat], patterntracks[ipat*32+chn], lpStream+dwTrackPos);
}
}
}
// Set up envelopes
for (UINT iIns=1; iIns<=m_nInstruments; iIns++) if (Headers[iIns])
{
INSTRUMENTHEADER *penv = Headers[iIns];
// Setup volume envelope
if ((nvolenv) && (pvolenv) && (insvolenv[iIns]))
{
LPCBYTE pve = pvolenv;
for (UINT nve=0; nve<nvolenv; nve++, pve+=33) if (pve[0]+1 == insvolenv[iIns])
{
WORD vtick = 1;
penv->nVolEnv = 15;
for (UINT iv=0; iv<15; iv++)
{
if (iv) vtick += pve[iv*2+1];
penv->VolPoints[iv] = vtick;
penv->VolEnv[iv] = pve[iv*2+2];
if (!pve[iv*2+1])
{
penv->nVolEnv = iv+1;
break;
}
}
penv->nVolSustainBegin = penv->nVolSustainEnd = pve[31] & 0x0F;
if (pve[31] & 0x10) penv->dwFlags |= ENV_VOLSUSTAIN;
if (pve[31] & 0x20) penv->dwFlags |= ENV_VOLLOOP;
penv->nVolLoopStart = pve[32] & 0x0F;
penv->nVolLoopEnd = pve[32] >> 4;
}
}
// Setup panning envelope
if ((npanenv) && (ppanenv) && (inspanenv[iIns]))
{
LPCBYTE ppe = ppanenv;
for (UINT npe=0; npe<npanenv; npe++, ppe+=33) if (ppe[0]+1 == inspanenv[iIns])
{
WORD vtick = 1;
penv->nPanEnv = 15;
for (UINT iv=0; iv<15; iv++)
{
if (iv) vtick += ppe[iv*2+1];
penv->PanPoints[iv] = vtick;
penv->PanEnv[iv] = ppe[iv*2+2];
if (!ppe[iv*2+1])
{
penv->nPanEnv = iv+1;
break;
}
}
if (ppe[31] & 0x10) penv->dwFlags |= ENV_PANSUSTAIN;
if (ppe[31] & 0x20) penv->dwFlags |= ENV_PANLOOP;
penv->nPanLoopStart = ppe[32] & 0x0F;
penv->nPanLoopEnd = ppe[32] >> 4;
}
}
}
m_dwSongFlags |= SONG_LINEARSLIDES;
m_nType = MOD_TYPE_MDL;
return TRUE;
}
/////////////////////////////////////////////////////////////////////////
// MDL Sample Unpacking
// MDL Huffman ReadBits compression
WORD MDLReadBits(DWORD &bitbuf, UINT &bitnum, LPBYTE &ibuf, CHAR n)
//-----------------------------------------------------------------
{
WORD v = (WORD)(bitbuf & ((1 << n) - 1) );
bitbuf >>= n;
bitnum -= n;
if (bitnum <= 24)
{
bitbuf |= (((DWORD)(*ibuf++)) << bitnum);
bitnum += 8;
}
return v;
}

@ -0,0 +1,931 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
#include "stdafx.h"
#include "sndfile.h"
//#define MED_LOG
#ifdef MED_LOG
extern void Log(LPCSTR s, ...);
#endif
//////////////////////////////////////////////////////////
// OctaMed MED file support (import only)
//
// Lookup table for bpm values.
static const BYTE bpmvals[10] = { 179,164,152,141,131,123,116,110,104,99 };
// flags
#define MMD_FLAG_FILTERON 0x1
#define MMD_FLAG_JUMPINGON 0x2
#define MMD_FLAG_JUMP8TH 0x4
#define MMD_FLAG_INSTRSATT 0x8 // instruments are attached (this is a module)
#define MMD_FLAG_VOLHEX 0x10
#define MMD_FLAG_STSLIDE 0x20 // SoundTracker mode for slides
#define MMD_FLAG_8CHANNEL 0x40 // OctaMED 8 channel song
#define MMD_FLAG_SLOWHQ 0x80 // HQ slows playing speed (V2-V4 compatibility)
// flags2
#define MMD_FLAG2_BMASK 0x1F
#define MMD_FLAG2_BPM 0x20
#define MMD_FLAG2_MIX 0x80 // uses Mixing (V7+)
// flags3:
#define MMD_FLAG3_STEREO 0x1 // mixing in Stereo mode
#define MMD_FLAG3_FREEPAN 0x2 // free panning
#define MMD_FLAG3_GM 0x4 // module designed for GM/XG compatibility
// generic MMD tags
#define MMDTAG_END 0
#define MMDTAG_PTR 0x80000000 // data needs relocation
#define MMDTAG_MUSTKNOW 0x40000000 // loader must fail if this isn't recognized
#define MMDTAG_MUSTWARN 0x20000000 // loader must warn if this isn't recognized
// ExpData tags
// # of effect groups, including the global group (will
// override settings in MMDSong struct), default = 1
#define MMDTAG_EXP_NUMFXGROUPS 1
#define MMDTAG_TRK_NAME (MMDTAG_PTR|1) // trackinfo tags
#define MMDTAG_TRK_NAMELEN 2 // namelen includes zero term.
#define MMDTAG_TRK_FXGROUP 3
// effectinfo tags
#define MMDTAG_FX_ECHOTYPE 1
#define MMDTAG_FX_ECHOLEN 2
#define MMDTAG_FX_ECHODEPTH 3
#define MMDTAG_FX_STEREOSEP 4
#define MMDTAG_FX_GROUPNAME (MMDTAG_PTR|5) // the Global Effects group shouldn't have name saved!
#define MMDTAG_FX_GRPNAMELEN 6 // namelen includes zero term.
#pragma pack(1)
typedef struct tagMEDMODULEHEADER
{
DWORD id; // MMD1-MMD3
DWORD modlen; // Size of file
DWORD song; // Position in file for this song
WORD psecnum;
WORD pseq;
DWORD blockarr; // Position in file for blocks
DWORD mmdflags;
DWORD smplarr; // Position in file for samples
DWORD reserved;
DWORD expdata; // Absolute offset in file for ExpData (0 if not present)
DWORD reserved2;
WORD pstate;
WORD pblock;
WORD pline;
WORD pseqnum;
WORD actplayline;
BYTE counter;
BYTE extra_songs; // # of songs - 1
} MEDMODULEHEADER;
typedef struct tagMMD0SAMPLE
{
WORD rep, replen;
BYTE midich;
BYTE midipreset;
BYTE svol;
signed char strans;
} MMD0SAMPLE;
// Sample header is immediately followed by sample data...
typedef struct tagMMDSAMPLEHEADER
{
DWORD length; // length of *one* *unpacked* channel in *bytes*
WORD type;
// if non-negative
// bits 0-3 reserved for multi-octave instruments, not supported on the PC
// 0x10: 16 bit (otherwise 8 bit)
// 0x20: Stereo (otherwise mono)
// 0x40: Uses DeltaCode
// 0x80: Packed data
// -1: Synth
// -2: Hybrid
// if type indicates packed data, these fields follow, otherwise we go right to the data
WORD packtype; // Only 1 = ADPCM is supported
WORD subtype; // Packing subtype
// ADPCM subtype
// 1: g723_40
// 2: g721
// 3: g723_24
BYTE commonflags; // flags common to all packtypes (none defined so far)
BYTE packerflags; // flags for the specific packtype
ULONG leftchlen; // packed length of left channel in bytes
ULONG rightchlen; // packed length of right channel in bytes (ONLY PRESENT IN STEREO SAMPLES)
BYTE SampleData[1]; // Sample Data
} MMDSAMPLEHEADER;
// MMD0/MMD1 song header
typedef struct tagMMD0SONGHEADER
{
MMD0SAMPLE sample[63];
WORD numblocks; // # of blocks
WORD songlen; // # of entries used in playseq
BYTE playseq[256]; // Play sequence
WORD deftempo; // BPM tempo
signed char playtransp; // Play transpose
BYTE flags; // 0x10: Hex Volumes | 0x20: ST/NT/PT Slides | 0x40: 8 Channels song
BYTE flags2; // [b4-b0]+1: Tempo LPB, 0x20: tempo mode, 0x80: mix_conv=on
BYTE tempo2; // tempo TPL
BYTE trkvol[16]; // track volumes
BYTE mastervol; // master volume
BYTE numsamples; // # of samples (max=63)
} MMD0SONGHEADER;
// MMD2/MMD3 song header
typedef struct tagMMD2SONGHEADER
{
MMD0SAMPLE sample[63];
WORD numblocks; // # of blocks
WORD numsections; // # of sections
DWORD playseqtable; // filepos of play sequence
DWORD sectiontable; // filepos of sections table (WORD array)
DWORD trackvols; // filepos of tracks volume (BYTE array)
WORD numtracks; // # of tracks (max 64)
WORD numpseqs; // # of play sequences
DWORD trackpans; // filepos of tracks pan values (BYTE array)
LONG flags3; // 0x1:stereo_mix, 0x2:free_panning, 0x4:GM/XG compatibility
WORD voladj; // vol_adjust (set to 100 if 0)
WORD channels; // # of channels (4 if =0)
BYTE mix_echotype; // 1:normal,2:xecho
BYTE mix_echodepth; // 1..6
WORD mix_echolen; // > 0
signed char mix_stereosep; // -4..4
BYTE pad0[223];
WORD deftempo; // BPM tempo
signed char playtransp; // play transpose
BYTE flags; // 0x1:filteron, 0x2:jumpingon, 0x4:jump8th, 0x8:instr_attached, 0x10:hex_vol, 0x20:PT_slides, 0x40:8ch_conv,0x80:hq slows playing speed
BYTE flags2; // 0x80:mix_conv=on, [b4-b0]+1:tempo LPB, 0x20:tempo_mode
BYTE tempo2; // tempo TPL
BYTE pad1[16];
BYTE mastervol; // master volume
BYTE numsamples; // # of samples (max 63)
} MMD2SONGHEADER;
// For MMD0 the note information is held in 3 bytes, byte0, byte1, byte2. For reference we
// number the bits in each byte 0..7, where 0 is the low bit.
// The note is held as bits 5..0 of byte0
// The instrument is encoded in 6 bits, bits 7 and 6 of byte0 and bits 7,6,5,4 of byte1
// The command number is bits 3,2,1,0 of byte1, command data is in byte2:
// For command 0, byte2 represents the second data byte, otherwise byte2
// represents the first data byte.
typedef struct tagMMD0BLOCK
{
BYTE numtracks;
BYTE lines; // File value is 1 less than actual, so 0 -> 1 line
} MMD0BLOCK; // BYTE data[lines+1][tracks][3];
// For MMD1,MMD2,MMD3 the note information is carried in 4 bytes, byte0, byte1,
// byte2 and byte3
// The note is held as byte0 (values above 0x84 are ignored)
// The instrument is held as byte1
// The command number is held as byte2, command data is in byte3
// For commands 0 and 0x19 byte3 represents the second data byte,
// otherwise byte2 represents the first data byte.
typedef struct tagMMD1BLOCK
{
WORD numtracks; // Number of tracks, may be > 64, but then that data is skipped.
WORD lines; // Stored value is 1 less than actual, so 0 -> 1 line
DWORD info; // Offset of BlockInfo (if 0, no block_info is present)
} MMD1BLOCK;
typedef struct tagMMD1BLOCKINFO
{
DWORD hlmask; // Unimplemented - ignore
DWORD blockname; // file offset of block name
DWORD blocknamelen; // length of block name (including term. 0)
DWORD pagetable; // file offset of command page table
DWORD cmdexttable; // file offset of command extension table
DWORD reserved[4]; // future expansion
} MMD1BLOCKINFO;
// A set of play sequences is stored as an array of ULONG files offsets
// Each offset points to the play sequence itself.
typedef struct tagMMD2PLAYSEQ
{
CHAR name[32];
DWORD command_offs; // filepos of command table
DWORD reserved;
WORD length;
WORD seq[512]; // skip if > 0x8000
} MMD2PLAYSEQ;
// A command table contains commands that effect a particular play sequence
// entry. The only commands read in are STOP or POSJUMP, all others are ignored
// POSJUMP is presumed to have extra bytes containing a WORD for the position
typedef struct tagMMDCOMMAND
{
WORD offset; // Offset within current sequence entry
BYTE cmdnumber; // STOP (537) or POSJUMP (538) (others skipped)
BYTE extra_count;
BYTE extra_bytes[4];// [extra_count];
} MMDCOMMAND; // Last entry has offset == 0xFFFF, cmd_number == 0 and 0 extrabytes
typedef struct tagMMD0EXP
{
DWORD nextmod; // File offset of next Hdr
DWORD exp_smp; // Pointer to extra instrument data
WORD s_ext_entries; // Number of extra instrument entries
WORD s_ext_entrsz; // Size of extra instrument data
DWORD annotxt;
DWORD annolen;
DWORD iinfo; // Instrument names
WORD i_ext_entries;
WORD i_ext_entrsz;
DWORD jumpmask;
DWORD rgbtable;
BYTE channelsplit[4]; // Only used if 8ch_conv (extra channel for every nonzero entry)
DWORD n_info;
DWORD songname; // Song name
DWORD songnamelen;
DWORD dumps;
DWORD mmdinfo;
DWORD mmdrexx;
DWORD mmdcmd3x;
DWORD trackinfo_ofs; // ptr to song->numtracks ptrs to tag lists
DWORD effectinfo_ofs; // ptr to group ptrs
DWORD tag_end;
} MMD0EXP;
#pragma pack()
static void MedConvert(MODCOMMAND *p, const MMD0SONGHEADER *pmsh)
//---------------------------------------------------------------
{
UINT command = p->command;
UINT param = p->param;
switch(command)
{
case 0x00: if (param) command = CMD_ARPEGGIO; else command = 0; break;
case 0x01: command = CMD_PORTAMENTOUP; break;
case 0x02: command = CMD_PORTAMENTODOWN; break;
case 0x03: command = CMD_TONEPORTAMENTO; break;
case 0x04: command = CMD_VIBRATO; break;
case 0x05: command = CMD_TONEPORTAVOL; break;
case 0x06: command = CMD_VIBRATOVOL; break;
case 0x07: command = CMD_TREMOLO; break;
case 0x0A: if (param & 0xF0) param &= 0xF0; command = CMD_VOLUMESLIDE; if (!param) command = 0; break;
case 0x0B: command = CMD_POSITIONJUMP; break;
case 0x0C: command = CMD_VOLUME;
if (pmsh->flags & MMD_FLAG_VOLHEX)
{
if (param < 0x80)
{
param = (param+1) / 2;
} else command = 0;
} else
{
if (param <= 0x99)
{
param = (param >> 4)*10+((param & 0x0F) % 10);
if (param > 64) param = 64;
} else command = 0;
}
break;
case 0x09: command = (param < 0x20) ? CMD_SPEED : CMD_TEMPO; break;
case 0x0D: if (param & 0xF0) param &= 0xF0; command = CMD_VOLUMESLIDE; if (!param) command = 0; break;
case 0x0F: // Set Tempo / Special
// F.00 = Pattern Break
if (!param) command = CMD_PATTERNBREAK; else
// F.01 - F.F0: Set tempo/speed
if (param <= 0xF0)
{
if (pmsh->flags & MMD_FLAG_8CHANNEL)
{
param = (param > 10) ? 99 : bpmvals[param-1];
} else
// F.01 - F.0A: Set Speed
if (param <= 0x0A)
{
command = CMD_SPEED;
} else
// Old tempo
if (!(pmsh->flags2 & MMD_FLAG2_BPM))
{
param = _muldiv(param, 5*715909, 2*474326);
}
// F.0B - F.F0: Set Tempo (assumes LPB=4)
if (param > 0x0A)
{
command = CMD_TEMPO;
if (param < 0x21) param = 0x21;
if (param > 240) param = 240;
}
} else
switch(param)
{
// F.F1: Retrig 2x
case 0xF1:
command = CMD_MODCMDEX;
param = 0x93;
break;
// F.F2: Note Delay 2x
case 0xF2:
command = CMD_MODCMDEX;
param = 0xD3;
break;
// F.F3: Retrig 3x
case 0xF3:
command = CMD_MODCMDEX;
param = 0x92;
break;
// F.F4: Note Delay 1/3
case 0xF4:
command = CMD_MODCMDEX;
param = 0xD2;
break;
// F.F5: Note Delay 2/3
case 0xF5:
command = CMD_MODCMDEX;
param = 0xD4;
break;
// F.F8: Filter Off
case 0xF8:
command = CMD_MODCMDEX;
param = 0x00;
break;
// F.F9: Filter On
case 0xF9:
command = CMD_MODCMDEX;
param = 0x01;
break;
// F.FD: Very fast tone-portamento
case 0xFD:
command = CMD_TONEPORTAMENTO;
param = 0xFF;
break;
// F.FE: End Song
case 0xFE:
command = CMD_SPEED;
param = 0;
break;
// F.FF: Note Cut
case 0xFF:
command = CMD_MODCMDEX;
param = 0xC0;
break;
default:
#ifdef MED_LOG
Log("Unknown Fxx command: cmd=0x%02X param=0x%02X\n", command, param);
#endif
param = command = 0;
}
break;
// 11.0x: Fine Slide Up
case 0x11:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0x10;
break;
// 12.0x: Fine Slide Down
case 0x12:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0x20;
break;
// 14.xx: Vibrato
case 0x14:
command = CMD_VIBRATO;
break;
// 15.xx: FineTune
case 0x15:
command = CMD_MODCMDEX;
param &= 0x0F;
param |= 0x50;
break;
// 16.xx: Pattern Loop
case 0x16:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0x60;
break;
// 18.xx: Note Cut
case 0x18:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0xC0;
break;
// 19.xx: Sample Offset
case 0x19:
command = CMD_OFFSET;
break;
// 1A.0x: Fine Volume Up
case 0x1A:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0xA0;
break;
// 1B.0x: Fine Volume Down
case 0x1B:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0xB0;
break;
// 1D.xx: Pattern Break
case 0x1D:
command = CMD_PATTERNBREAK;
break;
// 1E.0x: Pattern Delay
case 0x1E:
command = CMD_MODCMDEX;
if (param > 0x0F) param = 0x0F;
param |= 0xE0;
break;
// 1F.xy: Retrig
case 0x1F:
command = CMD_RETRIG;
param &= 0x0F;
break;
// 2E.xx: set panning
case 0x2E:
command = CMD_MODCMDEX;
param = ((param + 0x10) & 0xFF) >> 1;
if (param > 0x0F) param = 0x0F;
param |= 0x80;
break;
default:
#ifdef MED_LOG
// 0x2E ?
Log("Unknown command: cmd=0x%02X param=0x%02X\n", command, param);
#endif
command = param = 0;
}
p->command = command;
p->param = param;
}
BOOL CSoundFile::ReadMed(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const MEDMODULEHEADER *pmmh;
const MMD0SONGHEADER *pmsh;
const MMD2SONGHEADER *pmsh2;
const MMD0EXP *pmex;
DWORD dwBlockArr, dwSmplArr, dwExpData, wNumBlocks;
LPDWORD pdwTable;
CHAR version;
UINT deftempo;
int playtransp = 0;
if ((!lpStream) || (dwMemLength < 0x200)) return FALSE;
pmmh = (MEDMODULEHEADER *)lpStream;
if (((pmmh->id & 0x00FFFFFF) != 0x444D4D) || (!pmmh->song)) return FALSE;
// Check for 'MMDx'
DWORD dwSong = bswapBE32(pmmh->song);
if ((dwSong >= dwMemLength) || (dwSong + sizeof(MMD0SONGHEADER) >= dwMemLength)) return FALSE;
version = (signed char)((pmmh->id >> 24) & 0xFF);
if ((version < '0') || (version > '3')) return FALSE;
#ifdef MED_LOG
Log("\nLoading MMD%c module (flags=0x%02X)...\n", version, bswapBE32(pmmh->mmdflags));
Log(" modlen = %d\n", bswapBE32(pmmh->modlen));
Log(" song = 0x%08X\n", bswapBE32(pmmh->song));
Log(" psecnum = %d\n", bswapBE16(pmmh->psecnum));
Log(" pseq = %d\n", bswapBE16(pmmh->pseq));
Log(" blockarr = 0x%08X\n", bswapBE32(pmmh->blockarr));
Log(" mmdflags = 0x%08X\n", bswapBE32(pmmh->mmdflags));
Log(" smplarr = 0x%08X\n", bswapBE32(pmmh->smplarr));
Log(" reserved = 0x%08X\n", bswapBE32(pmmh->reserved));
Log(" expdata = 0x%08X\n", bswapBE32(pmmh->expdata));
Log(" reserved2= 0x%08X\n", bswapBE32(pmmh->reserved2));
Log(" pstate = %d\n", bswapBE16(pmmh->pstate));
Log(" pblock = %d\n", bswapBE16(pmmh->pblock));
Log(" pline = %d\n", bswapBE16(pmmh->pline));
Log(" pseqnum = %d\n", bswapBE16(pmmh->pseqnum));
Log(" actplayline=%d\n", bswapBE16(pmmh->actplayline));
Log(" counter = %d\n", pmmh->counter);
Log(" extra_songs = %d\n", pmmh->extra_songs);
Log("\n");
#endif
m_nType = MOD_TYPE_MED;
m_nSongPreAmp = 0x20;
dwBlockArr = bswapBE32(pmmh->blockarr);
dwSmplArr = bswapBE32(pmmh->smplarr);
dwExpData = bswapBE32(pmmh->expdata);
if ((dwExpData) && (dwExpData < dwMemLength - sizeof(MMD0EXP)))
pmex = (MMD0EXP *)(lpStream+dwExpData);
else
pmex = NULL;
pmsh = (MMD0SONGHEADER *)(lpStream + dwSong);
pmsh2 = (MMD2SONGHEADER *)pmsh;
#ifdef MED_LOG
if (version < '2')
{
Log("MMD0 Header:\n");
Log(" numblocks = %d\n", bswapBE16(pmsh->numblocks));
Log(" songlen = %d\n", bswapBE16(pmsh->songlen));
Log(" playseq = ");
for (UINT idbg1=0; idbg1<16; idbg1++) Log("%2d, ", pmsh->playseq[idbg1]);
Log("...\n");
Log(" deftempo = 0x%04X\n", bswapBE16(pmsh->deftempo));
Log(" playtransp = %d\n", (signed char)pmsh->playtransp);
Log(" flags(1,2) = 0x%02X, 0x%02X\n", pmsh->flags, pmsh->flags2);
Log(" tempo2 = %d\n", pmsh->tempo2);
Log(" trkvol = ");
for (UINT idbg2=0; idbg2<16; idbg2++) Log("0x%02X, ", pmsh->trkvol[idbg2]);
Log("...\n");
Log(" mastervol = 0x%02X\n", pmsh->mastervol);
Log(" numsamples = %d\n", pmsh->numsamples);
} else
{
Log("MMD2 Header:\n");
Log(" numblocks = %d\n", bswapBE16(pmsh2->numblocks));
Log(" numsections= %d\n", bswapBE16(pmsh2->numsections));
Log(" playseqptr = 0x%04X\n", bswapBE32(pmsh2->playseqtable));
Log(" sectionptr = 0x%04X\n", bswapBE32(pmsh2->sectiontable));
Log(" trackvols = 0x%04X\n", bswapBE32(pmsh2->trackvols));
Log(" numtracks = %d\n", bswapBE16(pmsh2->numtracks));
Log(" numpseqs = %d\n", bswapBE16(pmsh2->numpseqs));
Log(" trackpans = 0x%04X\n", bswapBE32(pmsh2->trackpans));
Log(" flags3 = 0x%08X\n", bswapBE32(pmsh2->flags3));
Log(" voladj = %d\n", bswapBE16(pmsh2->voladj));
Log(" channels = %d\n", bswapBE16(pmsh2->channels));
Log(" echotype = %d\n", pmsh2->mix_echotype);
Log(" echodepth = %d\n", pmsh2->mix_echodepth);
Log(" echolen = %d\n", bswapBE16(pmsh2->mix_echolen));
Log(" stereosep = %d\n", (signed char)pmsh2->mix_stereosep);
Log(" deftempo = 0x%04X\n", bswapBE16(pmsh2->deftempo));
Log(" playtransp = %d\n", (signed char)pmsh2->playtransp);
Log(" flags(1,2) = 0x%02X, 0x%02X\n", pmsh2->flags, pmsh2->flags2);
Log(" tempo2 = %d\n", pmsh2->tempo2);
Log(" mastervol = 0x%02X\n", pmsh2->mastervol);
Log(" numsamples = %d\n", pmsh->numsamples);
}
Log("\n");
#endif
wNumBlocks = bswapBE16(pmsh->numblocks);
m_nChannels = 4;
m_nSamples = pmsh->numsamples;
if (m_nSamples > 63) m_nSamples = 63;
// Tempo
m_nDefaultTempo = 125;
deftempo = bswapBE16(pmsh->deftempo);
if (!deftempo) deftempo = 125;
if (pmsh->flags2 & MMD_FLAG2_BPM)
{
UINT tempo_tpl = (pmsh->flags2 & MMD_FLAG2_BMASK) + 1;
if (!tempo_tpl) tempo_tpl = 4;
deftempo *= tempo_tpl;
deftempo /= 4;
#ifdef MED_LOG
Log("newtempo: %3d bpm (bpm=%3d lpb=%2d)\n", deftempo, bswapBE16(pmsh->deftempo), (pmsh->flags2 & MMD_FLAG2_BMASK)+1);
#endif
} else
{
if (pmsh->flags & MMD_FLAG_8CHANNEL && deftempo > 0 && deftempo <= 10)
{
deftempo = bpmvals[deftempo-1];
} else {
deftempo = _muldiv(deftempo, 5*715909, 2*474326);
}
#ifdef MED_LOG
Log("oldtempo: %3d bpm (bpm=%3d)\n", deftempo, bswapBE16(pmsh->deftempo));
#endif
}
// Speed
m_nDefaultSpeed = pmsh->tempo2;
if (!m_nDefaultSpeed) m_nDefaultSpeed = 6;
if (deftempo < 0x21) deftempo = 0x21;
if (deftempo > 255)
{
while ((m_nDefaultSpeed > 3) && (deftempo > 260))
{
deftempo = (deftempo * (m_nDefaultSpeed - 1)) / m_nDefaultSpeed;
m_nDefaultSpeed--;
}
if (deftempo > 255) deftempo = 255;
}
m_nDefaultTempo = deftempo;
// Reading Samples
for (UINT iSHdr=0; iSHdr<m_nSamples; iSHdr++)
{
MODINSTRUMENT *pins = &Ins[iSHdr+1];
pins->nLoopStart = bswapBE16(pmsh->sample[iSHdr].rep) << 1;
pins->nLoopEnd = pins->nLoopStart + (bswapBE16(pmsh->sample[iSHdr].replen) << 1);
pins->nVolume = (pmsh->sample[iSHdr].svol << 2);
pins->nGlobalVol = 64;
if (pins->nVolume > 256) pins->nVolume = 256;
pins->RelativeTone = -12 * pmsh->sample[iSHdr].strans;
pins->nPan = 128;
if (pins->nLoopEnd) pins->uFlags |= CHN_LOOP;
}
// Common Flags
if (!(pmsh->flags & 0x20)) m_dwSongFlags |= SONG_FASTVOLSLIDES;
// Reading play sequence
if (version < '2')
{
UINT nbo = pmsh->songlen >> 8;
if (nbo >= MAX_ORDERS) nbo = MAX_ORDERS-1;
if (!nbo) nbo = 1;
memcpy(Order, pmsh->playseq, nbo);
playtransp = pmsh->playtransp;
} else
{
UINT nOrders, nSections;
UINT nTrks = bswapBE16(pmsh2->numtracks);
if ((nTrks >= 4) && (nTrks <= 32)) m_nChannels = nTrks;
DWORD playseqtable = bswapBE32(pmsh2->playseqtable);
UINT numplayseqs = bswapBE16(pmsh2->numpseqs);
if (!numplayseqs) numplayseqs = 1;
nOrders = 0;
nSections = bswapBE16(pmsh2->numsections);
DWORD sectiontable = bswapBE32(pmsh2->sectiontable);
if ((!nSections) || (!sectiontable) || (sectiontable >= dwMemLength-2)) nSections = 1;
nOrders = 0;
for (UINT iSection=0; iSection<nSections; iSection++)
{
UINT nplayseq = 0;
if ((sectiontable) && (sectiontable < dwMemLength-2))
{
nplayseq = lpStream[sectiontable+1];
sectiontable += 2; // WORDs
} else
{
nSections = 0;
}
UINT pseq = 0;
if ((playseqtable) && (playseqtable < dwMemLength) && (nplayseq*4 < dwMemLength - playseqtable))
{
pseq = bswapBE32(((LPDWORD)(lpStream+playseqtable))[nplayseq]);
}
if ((pseq) && (pseq < dwMemLength - sizeof(MMD2PLAYSEQ)))
{
const MMD2PLAYSEQ *pmps = (MMD2PLAYSEQ *)(lpStream + pseq);
if (!m_szNames[0][0]) memcpy(m_szNames[0], pmps->name, 31);
UINT n = bswapBE16(pmps->length);
if (n < (dwMemLength - (pseq + sizeof(*pmps)) + sizeof(pmps->seq)) / sizeof(pmps->seq[0]))
{
for (UINT i=0; i<n; i++)
{
UINT seqval = pmps->seq[i] >> 8;
if ((seqval < wNumBlocks) && (nOrders < MAX_ORDERS-1))
{
Order[nOrders++] = seqval;
}
}
}
}
}
playtransp = pmsh2->playtransp;
while (nOrders < MAX_ORDERS) Order[nOrders++] = 0xFF;
}
// Reading Expansion structure
if (pmex)
{
// Channel Split
if ((m_nChannels == 4) && (pmsh->flags & MMD_FLAG_8CHANNEL))
{
for (UINT i8ch=0; i8ch<4; i8ch++)
{
if (pmex->channelsplit[i8ch]) m_nChannels++;
}
}
// Song Comments
uint32_t annotxt = bswapBE32(pmex->annotxt);
uint32_t annolen = bswapBE32(pmex->annolen);
if ((annotxt) && (annolen) && (annotxt + annolen > annotxt) // overflow checks.
&& (annotxt+annolen <= dwMemLength))
{
m_lpszSongComments = new char[annolen+1];
memcpy(m_lpszSongComments, lpStream+annotxt, annolen);
m_lpszSongComments[annolen] = 0;
}
// Song Name
uint32_t songname = bswapBE32(pmex->songname);
uint32_t songnamelen = bswapBE32(pmex->songnamelen);
if ((songname) && (songnamelen) && (songname+songnamelen > songname)
&& (songname+songnamelen <= dwMemLength))
{
if (songnamelen > 31) songnamelen = 31;
memcpy(m_szNames[0], lpStream+songname, songnamelen);
m_szNames[0][31] = '\0';
}
// Sample Names
DWORD smpinfoex = bswapBE32(pmex->iinfo);
if (smpinfoex)
{
DWORD iinfoptr = bswapBE32(pmex->iinfo);
UINT ientries = bswapBE16(pmex->i_ext_entries);
UINT ientrysz = bswapBE16(pmex->i_ext_entrsz);
if ((iinfoptr) && (ientrysz < 256) &&
(ientries*ientrysz < dwMemLength) &&
(iinfoptr < dwMemLength - (ientries*ientrysz)))
{
LPCSTR psznames = (LPCSTR)(lpStream + iinfoptr);
UINT maxnamelen = ientrysz;
// copy a max of 32 bytes.
if (maxnamelen > 32) maxnamelen = 32;
for (UINT i=0; i<ientries; i++) if (i < m_nSamples)
{
lstrcpyn(m_szNames[i+1], psznames + i*ientrysz, maxnamelen);
m_szNames[i+1][31] = '\0';
}
}
}
// Track Names
DWORD trackinfo_ofs = bswapBE32(pmex->trackinfo_ofs);
if ((trackinfo_ofs) && (trackinfo_ofs < dwMemLength) && (m_nChannels * 4 < dwMemLength - trackinfo_ofs))
{
DWORD *ptrktags = (DWORD *)(lpStream + trackinfo_ofs);
for (UINT i=0; i<m_nChannels; i++)
{
DWORD trknameofs = 0, trknamelen = 0;
DWORD trktagofs = bswapBE32(ptrktags[i]);
if (trktagofs)
{
while (trktagofs < dwMemLength - 8)
{
DWORD ntag = bswapBE32(*(DWORD *)(lpStream + trktagofs));
if (ntag == MMDTAG_END) break;
DWORD tagdata = bswapBE32(*(DWORD *)(lpStream + trktagofs + 4));
switch(ntag)
{
case MMDTAG_TRK_NAMELEN: trknamelen = tagdata; break;
case MMDTAG_TRK_NAME: trknameofs = tagdata; break;
}
trktagofs += 8;
}
if (trknamelen > MAX_CHANNELNAME) trknamelen = MAX_CHANNELNAME;
if ((trknameofs) && (trknamelen < dwMemLength) && (trknameofs < dwMemLength - trknamelen))
{
lstrcpyn(ChnSettings[i].szName, (LPCSTR)(lpStream+trknameofs), MAX_CHANNELNAME);
ChnSettings[i].szName[MAX_CHANNELNAME-1] = '\0';
}
}
}
}
}
// Reading samples
if (dwSmplArr > dwMemLength - 4*m_nSamples) return TRUE;
pdwTable = (LPDWORD)(lpStream + dwSmplArr);
for (UINT iSmp=0; iSmp<m_nSamples; iSmp++) if (pdwTable[iSmp])
{
UINT dwPos = bswapBE32(pdwTable[iSmp]);
if ((dwPos >= dwMemLength) || (dwPos + sizeof(MMDSAMPLEHEADER) >= dwMemLength)) continue;
MMDSAMPLEHEADER *psdh = (MMDSAMPLEHEADER *)(lpStream + dwPos);
UINT len = bswapBE32(psdh->length);
#ifdef MED_LOG
Log("SampleData %d: stype=0x%02X len=%d\n", iSmp, bswapBE16(psdh->type), len);
#endif
if ((len > MAX_SAMPLE_LENGTH) || (dwPos + len + 6 > dwMemLength)) len = 0;
UINT flags = RS_PCM8S, stype = bswapBE16(psdh->type);
LPSTR psdata = (LPSTR)(lpStream + dwPos + 6);
if (stype & 0x80)
{
psdata += (stype & 0x20) ? 14 : 6;
} else
{
if (stype & 0x10)
{
Ins[iSmp+1].uFlags |= CHN_16BIT;
len /= 2;
flags = (stype & 0x20) ? RS_STPCM16M : RS_PCM16M;
} else
{
flags = (stype & 0x20) ? RS_STPCM8S : RS_PCM8S;
}
if (stype & 0x20) len /= 2;
}
Ins[iSmp+1].nLength = len;
ReadSample(&Ins[iSmp+1], flags, psdata, dwMemLength - dwPos - 6);
}
// Reading patterns (blocks)
if (wNumBlocks > MAX_PATTERNS) wNumBlocks = MAX_PATTERNS;
if ((!dwBlockArr) || (dwBlockArr > dwMemLength - 4*wNumBlocks)) return TRUE;
pdwTable = (LPDWORD)(lpStream + dwBlockArr);
playtransp += (version == '3') ? 24 : 48;
for (UINT iBlk=0; iBlk<wNumBlocks; iBlk++)
{
UINT dwPos = bswapBE32(pdwTable[iBlk]);
if ((!dwPos) || (dwPos >= dwMemLength) || (dwPos >= dwMemLength - 8)) continue;
UINT lines = 64, tracks = 4;
if (version == '0')
{
const MMD0BLOCK *pmb = (const MMD0BLOCK *)(lpStream + dwPos);
lines = pmb->lines + 1;
tracks = pmb->numtracks;
if (!tracks) tracks = m_nChannels;
if ((Patterns[iBlk] = AllocatePattern(lines, m_nChannels)) == NULL) continue;
PatternSize[iBlk] = lines;
MODCOMMAND *p = Patterns[iBlk];
LPBYTE s = (LPBYTE)(lpStream + dwPos + 2);
UINT maxlen = tracks*lines*3;
if (maxlen + dwPos > dwMemLength - 2) break;
for (UINT y=0; y<lines; y++)
{
for (UINT x=0; x<tracks; x++, s+=3) if (x < m_nChannels)
{
BYTE note = s[0] & 0x3F;
BYTE instr = s[1] >> 4;
if (s[0] & 0x80) instr |= 0x10;
if (s[0] & 0x40) instr |= 0x20;
if ((note) && (note <= 132)) p->note = note + playtransp;
p->instr = instr;
p->command = s[1] & 0x0F;
p->param = s[2];
// if (!iBlk) Log("%02X.%02X.%02X | ", s[0], s[1], s[2]);
MedConvert(p, pmsh);
p++;
}
//if (!iBlk) Log("\n");
}
} else
{
const MMD1BLOCK *pmb = (MMD1BLOCK *)(lpStream + dwPos);
#ifdef MED_LOG
Log("MMD1BLOCK: lines=%2d, tracks=%2d, offset=0x%04X\n",
bswapBE16(pmb->lines), bswapBE16(pmb->numtracks), bswapBE32(pmb->info));
#endif
const MMD1BLOCKINFO *pbi = NULL;
BYTE *pcmdext = NULL;
lines = (pmb->lines >> 8) + 1;
tracks = pmb->numtracks >> 8;
if (!tracks) tracks = m_nChannels;
if ((Patterns[iBlk] = AllocatePattern(lines, m_nChannels)) == NULL) continue;
PatternSize[iBlk] = (WORD)lines;
DWORD dwBlockInfo = bswapBE32(pmb->info);
if ((dwBlockInfo) && (dwBlockInfo < dwMemLength - sizeof(MMD1BLOCKINFO)))
{
pbi = (MMD1BLOCKINFO *)(lpStream + dwBlockInfo);
#ifdef MED_LOG
Log(" BLOCKINFO: blockname=0x%04X namelen=%d pagetable=0x%04X &cmdexttable=0x%04X\n",
bswapBE32(pbi->blockname), bswapBE32(pbi->blocknamelen), bswapBE32(pbi->pagetable), bswapBE32(pbi->cmdexttable));
#endif
if ((pbi->blockname) && (pbi->blocknamelen))
{
DWORD nameofs = bswapBE32(pbi->blockname);
UINT namelen = bswapBE32(pbi->blocknamelen);
if ((nameofs < dwMemLength) && (namelen < dwMemLength + nameofs))
{
SetPatternName(iBlk, (LPCSTR)(lpStream+nameofs));
}
}
if (pbi->cmdexttable)
{
DWORD cmdexttable = bswapBE32(pbi->cmdexttable);
if (cmdexttable < dwMemLength - 4)
{
cmdexttable = bswapBE32(*(DWORD *)(lpStream + cmdexttable));
if ((cmdexttable) && (cmdexttable <= dwMemLength - lines*tracks))
{
pcmdext = (BYTE *)(lpStream + cmdexttable);
}
}
}
}
MODCOMMAND *p = Patterns[iBlk];
LPBYTE s = (LPBYTE)(lpStream + dwPos + 8);
UINT maxlen = tracks*lines*4;
if (maxlen + dwPos > dwMemLength - 8) break;
for (UINT y=0; y<lines; y++)
{
for (UINT x=0; x<tracks; x++, s+=4) if (x < m_nChannels)
{
BYTE note = s[0];
if ((note) && (note <= 132))
{
int rnote = note + playtransp;
if (rnote < 1) rnote = 1;
if (rnote > NOTE_MAX) rnote = NOTE_MAX;
p->note = (BYTE)rnote;
}
p->instr = s[1];
p->command = s[2];
p->param = s[3];
if (pcmdext) p->vol = pcmdext[x];
MedConvert(p, pmsh);
p++;
}
if (pcmdext) pcmdext += tracks;
}
}
}
// Setup channel pan positions
for (UINT iCh=0; iCh<m_nChannels; iCh++)
{
ChnSettings[iCh].nPan = (((iCh&3) == 1) || ((iCh&3) == 2)) ? 0xC0 : 0x40;
ChnSettings[iCh].nVolume = 64;
}
return TRUE;
}

File diff suppressed because it is too large Load Diff

@ -0,0 +1,536 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
#include "stdafx.h"
#include "sndfile.h"
#include "tables.h"
#ifdef _MSC_VER
//#pragma warning(disable:4244)
#endif
//////////////////////////////////////////////////////////
// ProTracker / NoiseTracker MOD/NST file support
void CSoundFile::ConvertModCommand(MODCOMMAND *m) const
//-----------------------------------------------------
{
UINT command = m->command, param = m->param;
switch(command)
{
case 0x00: if (param) command = CMD_ARPEGGIO; break;
case 0x01: command = CMD_PORTAMENTOUP; break;
case 0x02: command = CMD_PORTAMENTODOWN; break;
case 0x03: command = CMD_TONEPORTAMENTO; break;
case 0x04: command = CMD_VIBRATO; break;
case 0x05: command = CMD_TONEPORTAVOL; if (param & 0xF0) param &= 0xF0; break;
case 0x06: command = CMD_VIBRATOVOL; if (param & 0xF0) param &= 0xF0; break;
case 0x07: command = CMD_TREMOLO; break;
case 0x08: command = CMD_PANNING8; break;
case 0x09: command = CMD_OFFSET; break;
case 0x0A: command = CMD_VOLUMESLIDE; if (param & 0xF0) param &= 0xF0; break;
case 0x0B: command = CMD_POSITIONJUMP; break;
case 0x0C: command = CMD_VOLUME; break;
case 0x0D: command = CMD_PATTERNBREAK; param = ((param >> 4) * 10) + (param & 0x0F); break;
case 0x0E: command = CMD_MODCMDEX; break;
case 0x0F: command = (param <= (UINT)((m_nType & (MOD_TYPE_XM|MOD_TYPE_MT2)) ? 0x1F : 0x20)) ? CMD_SPEED : CMD_TEMPO;
if ((param == 0xFF) && (m_nSamples == 15)) command = 0; break;
// Extension for XM extended effects
case 'G' - 55: command = CMD_GLOBALVOLUME; break;
case 'H' - 55: command = CMD_GLOBALVOLSLIDE; if (param & 0xF0) param &= 0xF0; break;
case 'K' - 55: command = CMD_KEYOFF; break;
case 'L' - 55: command = CMD_SETENVPOSITION; break;
case 'M' - 55: command = CMD_CHANNELVOLUME; break;
case 'N' - 55: command = CMD_CHANNELVOLSLIDE; break;
case 'P' - 55: command = CMD_PANNINGSLIDE; if (param & 0xF0) param &= 0xF0; break;
case 'R' - 55: command = CMD_RETRIG; break;
case 'T' - 55: command = CMD_TREMOR; break;
case 'X' - 55: command = CMD_XFINEPORTAUPDOWN; break;
case 'Y' - 55: command = CMD_PANBRELLO; break;
case 'Z' - 55: command = CMD_MIDI; break;
default: command = 0;
}
m->command = command;
m->param = param;
}
WORD CSoundFile::ModSaveCommand(const MODCOMMAND *m, BOOL bXM) const
//------------------------------------------------------------------
{
UINT command = m->command & 0x3F, param = m->param;
switch(command)
{
case 0: command = param = 0; break;
case CMD_ARPEGGIO: command = 0; break;
case CMD_PORTAMENTOUP:
if (m_nType & (MOD_TYPE_S3M|MOD_TYPE_IT|MOD_TYPE_STM))
{
if ((param & 0xF0) == 0xE0) { command=0x0E; param=((param & 0x0F) >> 2)|0x10; break; }
else if ((param & 0xF0) == 0xF0) { command=0x0E; param &= 0x0F; param|=0x10; break; }
}
command = 0x01;
break;
case CMD_PORTAMENTODOWN:
if (m_nType & (MOD_TYPE_S3M|MOD_TYPE_IT|MOD_TYPE_STM))
{
if ((param & 0xF0) == 0xE0) { command=0x0E; param=((param & 0x0F) >> 2)|0x20; break; }
else if ((param & 0xF0) == 0xF0) { command=0x0E; param &= 0x0F; param|=0x20; break; }
}
command = 0x02;
break;
case CMD_TONEPORTAMENTO: command = 0x03; break;
case CMD_VIBRATO: command = 0x04; break;
case CMD_TONEPORTAVOL: command = 0x05; break;
case CMD_VIBRATOVOL: command = 0x06; break;
case CMD_TREMOLO: command = 0x07; break;
case CMD_PANNING8:
command = 0x08;
if (bXM)
{
if ((m_nType != MOD_TYPE_IT) && (m_nType != MOD_TYPE_XM) && (param <= 0x80))
{
param <<= 1;
if (param > 255) param = 255;
}
} else
{
if ((m_nType == MOD_TYPE_IT) || (m_nType == MOD_TYPE_XM)) param >>= 1;
}
break;
case CMD_OFFSET: command = 0x09; break;
case CMD_VOLUMESLIDE: command = 0x0A; break;
case CMD_POSITIONJUMP: command = 0x0B; break;
case CMD_VOLUME: command = 0x0C; break;
case CMD_PATTERNBREAK: command = 0x0D; param = ((param / 10) << 4) | (param % 10); break;
case CMD_MODCMDEX: command = 0x0E; break;
case CMD_SPEED: command = 0x0F; if (param > 0x20) param = 0x20; break;
case CMD_TEMPO: if (param > 0x20) { command = 0x0F; break; }
case CMD_GLOBALVOLUME: command = 'G' - 55; break;
case CMD_GLOBALVOLSLIDE: command = 'H' - 55; break;
case CMD_KEYOFF: command = 'K' - 55; break;
case CMD_SETENVPOSITION: command = 'L' - 55; break;
case CMD_CHANNELVOLUME: command = 'M' - 55; break;
case CMD_CHANNELVOLSLIDE: command = 'N' - 55; break;
case CMD_PANNINGSLIDE: command = 'P' - 55; break;
case CMD_RETRIG: command = 'R' - 55; break;
case CMD_TREMOR: command = 'T' - 55; break;
case CMD_XFINEPORTAUPDOWN: command = 'X' - 55; break;
case CMD_PANBRELLO: command = 'Y' - 55; break;
case CMD_MIDI: command = 'Z' - 55; break;
case CMD_S3MCMDEX:
switch(param & 0xF0)
{
case 0x10: command = 0x0E; param = (param & 0x0F) | 0x30; break;
case 0x20: command = 0x0E; param = (param & 0x0F) | 0x50; break;
case 0x30: command = 0x0E; param = (param & 0x0F) | 0x40; break;
case 0x40: command = 0x0E; param = (param & 0x0F) | 0x70; break;
case 0x90: command = 'X' - 55; break;
case 0xB0: command = 0x0E; param = (param & 0x0F) | 0x60; break;
case 0xA0:
case 0x50:
case 0x70:
case 0x60: command = param = 0; break;
default: command = 0x0E; break;
}
break;
default: command = param = 0;
}
return (WORD)((command << 8) | (param));
}
#pragma pack(1)
typedef struct _MODSAMPLE
{
CHAR name[22];
WORD length;
BYTE finetune;
BYTE volume;
WORD loopstart;
WORD looplen;
} MODSAMPLE, *PMODSAMPLE;
typedef struct _MODMAGIC
{
BYTE nOrders;
BYTE nRestartPos;
BYTE Orders[128];
char Magic[4]; // changed from CHAR
} MODMAGIC, *PMODMAGIC;
#pragma pack()
static BOOL IsValidName(LPCSTR s, int length, CHAR minChar)
//-----------------------------------------------------------------
{
int i, nt;
for (i = 0, nt = 0; i < length; i++)
{
if(s[i])
{
if (nt) return FALSE;// garbage after null
if (s[i] < minChar) return FALSE;// caller says it's garbage
}
else if (!nt) nt = i;// found null terminator
}
return TRUE;
}
BOOL IsMagic(LPCSTR s1, LPCSTR s2)
{
return ((*(DWORD *)s1) == (*(DWORD *)s2)) ? TRUE : FALSE;
}
BOOL CSoundFile::ReadMod(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
char s[1024]; // changed from CHAR
DWORD dwMemPos, dwTotalSampleLen;
PMODMAGIC pMagic;
UINT nErr;
if ((!lpStream) || (dwMemLength < 0x600)) return FALSE;
dwMemPos = 20;
m_nSamples = 31;
m_nChannels = 4;
pMagic = (PMODMAGIC)(lpStream+dwMemPos+sizeof(MODSAMPLE)*31);
// Check Mod Magic
memcpy(s, pMagic->Magic, 4);
if ((IsMagic(s, "M.K.")) || (IsMagic(s, "M!K!"))
|| (IsMagic(s, "M&K!")) || (IsMagic(s, "N.T."))) m_nChannels = 4; else
if ((IsMagic(s, "CD81")) || (IsMagic(s, "OKTA"))) m_nChannels = 8; else
if ((s[0]=='F') && (s[1]=='L') && (s[2]=='T') && (s[3]>='4') && (s[3]<='9')) m_nChannels = s[3] - '0'; else
if ((s[0]>='2') && (s[0]<='9') && (s[1]=='C') && (s[2]=='H') && (s[3]=='N')) m_nChannels = s[0] - '0'; else
if ((s[0]=='1') && (s[1]>='0') && (s[1]<='9') && (s[2]=='C') && (s[3]=='H')) m_nChannels = s[1] - '0' + 10; else
if ((s[0]=='2') && (s[1]>='0') && (s[1]<='9') && (s[2]=='C') && (s[3]=='H')) m_nChannels = s[1] - '0' + 20; else
if ((s[0]=='3') && (s[1]>='0') && (s[1]<='2') && (s[2]=='C') && (s[3]=='H')) m_nChannels = s[1] - '0' + 30; else
if ((s[0]=='T') && (s[1]=='D') && (s[2]=='Z') && (s[3]>='4') && (s[3]<='9')) m_nChannels = s[3] - '0'; else
if (IsMagic(s,"16CN")) m_nChannels = 16; else
if (IsMagic(s,"32CN")) m_nChannels = 32;
else {
if (!IsValidName((LPCSTR)lpStream, 20, ' '))
return FALSE;
m_nSamples = 15;
}
// Load Samples
nErr = 0;
dwTotalSampleLen = 0;
for (UINT i=1; i<=m_nSamples; i++)
{
PMODSAMPLE pms = (PMODSAMPLE)(lpStream+dwMemPos);
MODINSTRUMENT *psmp = &Ins[i];
UINT loopstart, looplen;
if (m_nSamples == 15)
{
if (!IsValidName((LPCSTR)pms->name, 22, 14)) return FALSE;
if (pms->finetune>>4) return FALSE;
if (pms->volume > 64) return FALSE;
if (bswapBE16(pms->length) > 32768) return FALSE;
}
memcpy(m_szNames[i], pms->name, 22);
m_szNames[i][22] = 0;
psmp->uFlags = 0;
psmp->nLength = bswapBE16(pms->length)*2;
dwTotalSampleLen += psmp->nLength;
psmp->nFineTune = MOD2XMFineTune(pms->finetune & 0x0F);
psmp->nVolume = 4*pms->volume;
if (psmp->nVolume > 256) { psmp->nVolume = 256; nErr++; }
psmp->nGlobalVol = 64;
psmp->nPan = 128;
loopstart = bswapBE16(pms->loopstart)*2;
looplen = bswapBE16(pms->looplen)*2;
// Fix loops
if ((looplen > 2) && (loopstart+looplen > psmp->nLength)
&& (loopstart/2+looplen <= psmp->nLength))
{
loopstart /= 2;
}
psmp->nLoopStart = loopstart;
psmp->nLoopEnd = loopstart + looplen;
if (psmp->nLength < 4) psmp->nLength = 0;
if (psmp->nLength)
{
if (psmp->nLoopStart >= psmp->nLength) { psmp->nLoopStart = psmp->nLength-1; }
if (psmp->nLoopEnd > psmp->nLength) { psmp->nLoopEnd = psmp->nLength; }
if ((psmp->nLoopStart > psmp->nLoopEnd) || (psmp->nLoopEnd <= 8)
|| (psmp->nLoopEnd - psmp->nLoopStart <= 4))
{
psmp->nLoopStart = 0;
psmp->nLoopEnd = 0;
}
if (psmp->nLoopEnd > psmp->nLoopStart)
{
psmp->uFlags |= CHN_LOOP;
}
}
dwMemPos += sizeof(MODSAMPLE);
}
if ((m_nSamples == 15) && (dwTotalSampleLen > dwMemLength * 4)) return FALSE;
pMagic = (PMODMAGIC)(lpStream+dwMemPos);
dwMemPos += sizeof(MODMAGIC);
if (m_nSamples == 15) {
dwMemPos -= 4;
if (pMagic->nOrders > 128) return FALSE;
}
memset(Order, 0,sizeof(Order));
memcpy(Order, pMagic->Orders, 128);
UINT nbp, nbpbuggy, nbpbuggy2, norders;
norders = pMagic->nOrders;
if ((!norders) || (norders > 0x80))
{
norders = 0x80;
while ((norders > 1) && (!Order[norders-1])) norders--;
}
nbpbuggy = 0;
nbpbuggy2 = 0;
nbp = 0;
for (UINT iord=0; iord<128; iord++)
{
UINT i = Order[iord];
if ((i < 0x80) && (nbp <= i))
{
nbp = i+1;
if (iord<norders) nbpbuggy = nbp;
}
if (i >= nbpbuggy2) nbpbuggy2 = i+1;
}
for (UINT iend=norders; iend<MAX_ORDERS; iend++) Order[iend] = 0xFF;
norders--;
m_nRestartPos = pMagic->nRestartPos;
if (m_nRestartPos >= 0x78) m_nRestartPos = 0;
if (m_nRestartPos + 1 >= (UINT)norders) m_nRestartPos = 0;
if (!nbp) return FALSE;
DWORD dwWowTest = dwTotalSampleLen+dwMemPos;
if ((IsMagic(pMagic->Magic, "M.K.")) && (dwWowTest + nbp*8*256 == dwMemLength)) m_nChannels = 8;
if ((nbp != nbpbuggy) && (dwWowTest + nbp*m_nChannels*256 != dwMemLength))
{
if (dwWowTest + nbpbuggy*m_nChannels*256 == dwMemLength) nbp = nbpbuggy;
else nErr += 8;
} else
if ((nbpbuggy2 > nbp) && (dwWowTest + nbpbuggy2*m_nChannels*256 == dwMemLength))
{
nbp = nbpbuggy2;
}
if ((dwWowTest < 0x600) || (dwWowTest > dwMemLength)) nErr += 8;
if ((m_nSamples == 15) && (nErr >= 16)) return FALSE;
// Default settings
m_nType = MOD_TYPE_MOD;
m_nDefaultSpeed = 6;
m_nDefaultTempo = 125;
m_nMinPeriod = 14 << 2;
m_nMaxPeriod = 3424 << 2;
memcpy(m_szNames, lpStream, 20);
// Setting channels pan
for (UINT ich=0; ich<m_nChannels; ich++)
{
ChnSettings[ich].nVolume = 64;
if (gdwSoundSetup & SNDMIX_MAXDEFAULTPAN)
ChnSettings[ich].nPan = (((ich&3)==1) || ((ich&3)==2)) ? 256 : 0;
else
ChnSettings[ich].nPan = (((ich&3)==1) || ((ich&3)==2)) ? 0xC0 : 0x40;
}
// Reading channels
for (UINT ipat=0; ipat<nbp; ipat++)
{
if (ipat < MAX_PATTERNS)
{
if ((Patterns[ipat] = AllocatePattern(64, m_nChannels)) == NULL) break;
PatternSize[ipat] = 64;
if (dwMemPos + m_nChannels*256 >= dwMemLength) break;
MODCOMMAND *m = Patterns[ipat];
LPCBYTE p = lpStream + dwMemPos;
for (UINT j=m_nChannels*64; j; m++,p+=4,j--)
{
BYTE A0=p[0], A1=p[1], A2=p[2], A3=p[3];
UINT n = ((((UINT)A0 & 0x0F) << 8) | (A1));
if ((n) && (n != 0xFFF)) m->note = GetNoteFromPeriod(n << 2);
m->instr = ((UINT)A2 >> 4) | (A0 & 0x10);
m->command = A2 & 0x0F;
m->param = A3;
if ((m->command) || (m->param)) ConvertModCommand(m);
}
}
dwMemPos += m_nChannels*256;
}
// Reading instruments
DWORD dwErrCheck = 0;
for (UINT ismp=1; ismp<=m_nSamples; ismp++) if (Ins[ismp].nLength)
{
LPSTR p = (LPSTR)(lpStream+dwMemPos);
UINT flags = 0;
if (dwMemPos + 5 >= dwMemLength) break;
if (! strncmp(p, "ADPCM", 5))
{
flags = 3;
p += 5;
dwMemPos += 5;
}
DWORD dwSize = ReadSample(&Ins[ismp], flags, p, dwMemLength - dwMemPos);
if (dwSize)
{
dwMemPos += dwSize;
dwErrCheck++;
}
}
#ifdef MODPLUG_TRACKER
return TRUE;
#else
return (dwErrCheck) ? TRUE : FALSE;
#endif
}
#ifndef MODPLUG_NO_FILESAVE
#ifdef _MSC_VER
#pragma warning(disable:4100)
#endif
BOOL CSoundFile::SaveMod(LPCSTR lpszFileName, UINT nPacking)
//----------------------------------------------------------
{
BYTE insmap[32];
UINT inslen[32];
BYTE bTab[32];
BYTE ord[128];
FILE *f;
if ((!m_nChannels) || (!lpszFileName)) return FALSE;
if ((f = fopen(lpszFileName, "wb")) == NULL) return FALSE;
memset(ord, 0, sizeof(ord));
memset(inslen, 0, sizeof(inslen));
if (m_nInstruments)
{
memset(insmap, 0, sizeof(insmap));
for (UINT i=1; i<32; i++) if (Headers[i])
{
for (UINT j=0; j<128; j++) if (Headers[i]->Keyboard[j])
{
insmap[i] = Headers[i]->Keyboard[j];
break;
}
}
} else
{
for (UINT i=0; i<32; i++) insmap[i] = (BYTE)i;
}
// Writing song name
fwrite(m_szNames, 20, 1, f);
// Writing instrument definition
for (UINT iins=1; iins<=31; iins++)
{
MODINSTRUMENT *pins = &Ins[insmap[iins]];
memcpy(bTab, m_szNames[iins],22);
inslen[iins] = pins->nLength;
if (inslen[iins] > 0x1fff0) inslen[iins] = 0x1fff0;
bTab[22] = inslen[iins] >> 9;
bTab[23] = inslen[iins] >> 1;
if (pins->RelativeTone < 0) bTab[24] = 0x08; else
if (pins->RelativeTone > 0) bTab[24] = 0x07; else
bTab[24] = (BYTE)XM2MODFineTune(pins->nFineTune);
bTab[25] = pins->nVolume >> 2;
bTab[26] = pins->nLoopStart >> 9;
bTab[27] = pins->nLoopStart >> 1;
bTab[28] = (pins->nLoopEnd - pins->nLoopStart) >> 9;
bTab[29] = (pins->nLoopEnd - pins->nLoopStart) >> 1;
fwrite(bTab, 30, 1, f);
}
// Writing number of patterns
UINT nbp=0, norders=128;
for (UINT iord=0; iord<128; iord++)
{
if (Order[iord] == 0xFF)
{
norders = iord;
break;
}
if ((Order[iord] < 0x80) && (nbp<=Order[iord])) nbp = Order[iord]+1;
}
bTab[0] = norders;
bTab[1] = m_nRestartPos;
fwrite(bTab, 2, 1, f);
// Writing pattern list
if (norders) memcpy(ord, Order, norders);
fwrite(ord, 128, 1, f);
// Writing signature
if (m_nChannels == 4)
lstrcpy((LPSTR)&bTab, "M.K.");
else
wsprintf((LPSTR)&bTab, "%luCHN", m_nChannels);
fwrite(bTab, 4, 1, f);
// Writing patterns
for (UINT ipat=0; ipat<nbp; ipat++) if (Patterns[ipat])
{
BYTE s[64*4];
MODCOMMAND *m = Patterns[ipat];
for (UINT i=0; i<64; i++) if (i < PatternSize[ipat])
{
LPBYTE p=s;
for (UINT c=0; c<m_nChannels; c++,p+=4,m++)
{
UINT param = ModSaveCommand(m, FALSE);
UINT command = param >> 8;
param &= 0xFF;
if (command > 0x0F) command = param = 0;
if ((m->vol >= 0x10) && (m->vol <= 0x50) && (!command) && (!param)) { command = 0x0C; param = m->vol - 0x10; }
UINT period = m->note;
if (period)
{
if (period < 37) period = 37;
period -= 37;
if (period >= 6*12) period = 6*12-1;
period = ProTrackerPeriodTable[period];
}
UINT instr = (m->instr > 31) ? 0 : m->instr;
p[0] = ((period >> 8) & 0x0F) | (instr & 0x10);
p[1] = period & 0xFF;
p[2] = ((instr & 0x0F) << 4) | (command & 0x0F);
p[3] = param;
}
fwrite(s, m_nChannels, 4, f);
} else
{
memset(s, 0, m_nChannels*4);
fwrite(s, m_nChannels, 4, f);
}
}
// Writing instruments
for (UINT ismpd=1; ismpd<=31; ismpd++) if (inslen[ismpd])
{
MODINSTRUMENT *pins = &Ins[insmap[ismpd]];
UINT flags = RS_PCM8S;
#ifndef NO_PACKING
if (!(pins->uFlags & (CHN_16BIT|CHN_STEREO)))
{
if ((nPacking) && (CanPackSample((char *)pins->pSample, inslen[ismpd], nPacking)))
{
fwrite("ADPCM", 1, 5, f);
flags = RS_ADPCM4;
}
}
#endif
WriteSample(f, pins, flags, inslen[ismpd]);
}
fclose(f);
return TRUE;
}
#ifdef _MSC_VER
#pragma warning(default:4100)
#endif
#endif // MODPLUG_NO_FILESAVE

@ -0,0 +1,635 @@
#include "stdafx.h"
#include "sndfile.h"
//#define MT2DEBUG
#pragma pack(1)
typedef struct _MT2FILEHEADER
{
DWORD dwMT20; // 0x3032544D "MT20"
DWORD dwSpecial;
WORD wVersion;
CHAR szTrackerName[32]; // "MadTracker 2.0"
CHAR szSongName[64];
WORD nOrders;
WORD wRestart;
WORD wPatterns;
WORD wChannels;
WORD wSamplesPerTick;
BYTE bTicksPerLine;
BYTE bLinesPerBeat;
DWORD fulFlags; // b0=packed patterns
WORD wInstruments;
WORD wSamples;
BYTE Orders[256];
} MT2FILEHEADER;
typedef struct _MT2PATTERN
{
WORD wLines;
DWORD wDataLen;
} MT2PATTERN;
typedef struct _MT2COMMAND
{
BYTE note; // 0=nothing, 97=note off
BYTE instr;
BYTE vol;
BYTE pan;
BYTE fxcmd;
BYTE fxparam1;
BYTE fxparam2;
} MT2COMMAND;
typedef struct _MT2DRUMSDATA
{
WORD wDrumPatterns;
WORD wDrumSamples[8];
BYTE DrumPatternOrder[256];
} MT2DRUMSDATA;
typedef struct _MT2AUTOMATION
{
DWORD dwFlags;
DWORD dwEffectId;
DWORD nEnvPoints;
} MT2AUTOMATION;
typedef struct _MT2INSTRUMENT
{
CHAR szName[32];
DWORD dwDataLen;
WORD wSamples;
BYTE GroupsMapping[96];
BYTE bVibType;
BYTE bVibSweep;
BYTE bVibDepth;
BYTE bVibRate;
WORD wFadeOut;
WORD wNNA;
WORD wInstrFlags;
WORD wEnvFlags1;
WORD wEnvFlags2;
} MT2INSTRUMENT;
typedef struct _MT2ENVELOPE
{
BYTE nFlags;
BYTE nPoints;
BYTE nSustainPos;
BYTE nLoopStart;
BYTE nLoopEnd;
BYTE bReserved[3];
BYTE EnvData[64];
} MT2ENVELOPE;
typedef struct _MT2SYNTH
{
BYTE nSynthId;
BYTE nFxId;
WORD wCutOff;
BYTE nResonance;
BYTE nAttack;
BYTE nDecay;
BYTE bReserved[25];
} MT2SYNTH;
typedef struct _MT2SAMPLE
{
CHAR szName[32];
DWORD dwDataLen;
DWORD dwLength;
DWORD dwFrequency;
BYTE nQuality;
BYTE nChannels;
BYTE nFlags;
BYTE nLoop;
DWORD dwLoopStart;
DWORD dwLoopEnd;
WORD wVolume;
BYTE nPan;
BYTE nBaseNote;
WORD wSamplesPerBeat;
} MT2SAMPLE;
typedef struct _MT2GROUP
{
BYTE nSmpNo;
BYTE nVolume; // 0-128
BYTE nFinePitch;
BYTE Reserved[5];
} MT2GROUP;
#pragma pack()
static VOID ConvertMT2Command(CSoundFile *that, MODCOMMAND *m, const MT2COMMAND *p)
//---------------------------------------------------------------------------
{
// Note
m->note = 0;
if (p->note) m->note = (p->note > 96) ? 0xFF : p->note+12;
// Instrument
m->instr = p->instr;
// Volume Column
if ((p->vol >= 0x10) && (p->vol <= 0x90))
{
m->volcmd = VOLCMD_VOLUME;
m->vol = (p->vol - 0x10) >> 1;
} else
if ((p->vol >= 0xA0) && (p->vol <= 0xAF))
{
m->volcmd = VOLCMD_VOLSLIDEDOWN;
m->vol = (p->vol & 0x0f);
} else
if ((p->vol >= 0xB0) && (p->vol <= 0xBF))
{
m->volcmd = VOLCMD_VOLSLIDEUP;
m->vol = (p->vol & 0x0f);
} else
if ((p->vol >= 0xC0) && (p->vol <= 0xCF))
{
m->volcmd = VOLCMD_FINEVOLDOWN;
m->vol = (p->vol & 0x0f);
} else
if ((p->vol >= 0xD0) && (p->vol <= 0xDF))
{
m->volcmd = VOLCMD_FINEVOLUP;
m->vol = (p->vol & 0x0f);
} else
{
m->volcmd = 0;
m->vol = 0;
}
// Effects
m->command = 0;
m->param = 0;
if ((p->fxcmd) || (p->fxparam1) || (p->fxparam2))
{
if (!p->fxcmd)
{
m->command = p->fxparam2;
m->param = p->fxparam1;
that->ConvertModCommand(m);
} else
{
// TODO: MT2 Effects
}
}
}
BOOL CSoundFile::ReadMT2(LPCBYTE lpStream, DWORD dwMemLength)
//-----------------------------------------------------------
{
const MT2FILEHEADER *pfh = (MT2FILEHEADER *)lpStream;
DWORD dwMemPos, dwDrumDataPos, dwExtraDataPos;
UINT nDrumDataLen, nExtraDataLen;
const MT2DRUMSDATA *pdd;
const MT2INSTRUMENT *InstrMap[255];
const MT2SAMPLE *SampleMap[256];
if ((!lpStream) || (dwMemLength < sizeof(MT2FILEHEADER))
|| (pfh->dwMT20 != 0x3032544D)
|| (pfh->wVersion < 0x0200) || (pfh->wVersion >= 0x0300)
|| (pfh->wChannels < 4) || (pfh->wChannels > 64)) return FALSE;
pdd = NULL;
m_nType = MOD_TYPE_MT2;
m_nChannels = pfh->wChannels;
m_nRestartPos = pfh->wRestart;
m_nDefaultSpeed = pfh->bTicksPerLine;
m_nDefaultTempo = 125;
if ((pfh->wSamplesPerTick > 100) && (pfh->wSamplesPerTick < 5000))
{
m_nDefaultTempo = 110250 / pfh->wSamplesPerTick;
}
for (UINT iOrd=0; iOrd<MAX_ORDERS; iOrd++)
{
Order[iOrd] = (BYTE)((iOrd < pfh->nOrders) ? pfh->Orders[iOrd] : 0xFF);
}
memcpy(m_szNames[0], pfh->szSongName, 32);
m_szNames[0][31] = 0;
dwMemPos = sizeof(MT2FILEHEADER);
nDrumDataLen = *(WORD *)(lpStream + dwMemPos);
dwDrumDataPos = dwMemPos + 2;
if (nDrumDataLen >= 2) pdd = (MT2DRUMSDATA *)(lpStream+dwDrumDataPos);
dwMemPos += 2 + nDrumDataLen;
#ifdef MT2DEBUG
Log("MT2 v%03X: \"%s\" (flags=%04X)\n", pfh->wVersion, m_szNames[0], pfh->fulFlags);
Log("%d Channels, %d Patterns, %d Instruments, %d Samples\n", pfh->wChannels, pfh->wPatterns, pfh->wInstruments, pfh->wSamples);
Log("Drum Data: %d bytes @%04X\n", nDrumDataLen, dwDrumDataPos);
#endif
if (dwMemPos >= dwMemLength-12) return TRUE;
if (!*(DWORD *)(lpStream+dwMemPos)) dwMemPos += 4;
if (!*(DWORD *)(lpStream+dwMemPos)) dwMemPos += 4;
nExtraDataLen = *(DWORD *)(lpStream+dwMemPos);
dwExtraDataPos = dwMemPos + 4;
dwMemPos += 4;
#ifdef MT2DEBUG
Log("Extra Data: %d bytes @%04X\n", nExtraDataLen, dwExtraDataPos);
#endif
if (dwMemPos + nExtraDataLen >= dwMemLength) return TRUE;
while (dwMemPos+8 < dwExtraDataPos + nExtraDataLen)
{
DWORD dwId = *(DWORD *)(lpStream+dwMemPos);
DWORD dwLen = *(DWORD *)(lpStream+dwMemPos+4);
dwMemPos += 8;
if (dwMemPos + dwLen > dwMemLength) return TRUE;
#ifdef MT2DEBUG
CHAR s[5];
memcpy(s, &dwId, 4);
s[4] = 0;
Log("pos=0x%04X: %s: %d bytes\n", dwMemPos-8, s, dwLen);
#endif
switch(dwId)
{
// MSG
case 0x0047534D:
if ((dwLen > 3) && (!m_lpszSongComments))
{
DWORD nTxtLen = dwLen;
if (nTxtLen > 32000) nTxtLen = 32000;
m_lpszSongComments = new char[nTxtLen]; // changed from CHAR
if (m_lpszSongComments)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos+1, nTxtLen-1);
m_lpszSongComments[nTxtLen-1] = 0;
}
}
break;
// SUM -> author name (or "Unregistered")
// TMAP
// TRKS
case 0x534b5254:
break;
}
dwMemPos += dwLen;
}
// Load Patterns
dwMemPos = dwExtraDataPos + nExtraDataLen;
for (UINT iPat=0; iPat<pfh->wPatterns; iPat++) if (dwMemPos < dwMemLength-6)
{
const MT2PATTERN *pmp = (MT2PATTERN *)(lpStream+dwMemPos);
UINT wDataLen = (pmp->wDataLen + 1) & ~1;
dwMemPos += 6;
if (dwMemPos + wDataLen > dwMemLength) break;
UINT nLines = pmp->wLines;
if ((iPat < MAX_PATTERNS) && (nLines > 0) && (nLines <= 256))
{
#ifdef MT2DEBUG
Log("Pattern #%d @%04X: %d lines, %d bytes\n", iPat, dwMemPos-6, nLines, pmp->wDataLen);
#endif
PatternSize[iPat] = nLines;
Patterns[iPat] = AllocatePattern(nLines, m_nChannels);
if (!Patterns[iPat]) return TRUE;
MODCOMMAND *m = Patterns[iPat];
UINT len = wDataLen;
if (pfh->fulFlags & 1) // Packed Patterns
{
BYTE *p = (BYTE *)(lpStream+dwMemPos);
UINT pos = 0, row=0, ch=0;
while (pos < len)
{
MT2COMMAND cmd;
UINT infobyte = p[pos++];
UINT rptcount = 0;
if (infobyte == 0xff)
{
rptcount = p[pos++];
infobyte = p[pos++];
#if 0
Log("(%d.%d) FF(%02X).%02X\n", row, ch, rptcount, infobyte);
} else
{
Log("(%d.%d) %02X\n", row, ch, infobyte);
#endif
}
if (infobyte & 0x7f)
{
UINT patpos = row*m_nChannels+ch;
cmd.note = cmd.instr = cmd.vol = cmd.pan = cmd.fxcmd = cmd.fxparam1 = cmd.fxparam2 = 0;
if (infobyte & 1) cmd.note = p[pos++];
if (infobyte & 2) cmd.instr = p[pos++];
if (infobyte & 4) cmd.vol = p[pos++];
if (infobyte & 8) cmd.pan = p[pos++];
if (infobyte & 16) cmd.fxcmd = p[pos++];
if (infobyte & 32) cmd.fxparam1 = p[pos++];
if (infobyte & 64) cmd.fxparam2 = p[pos++];
#ifdef MT2DEBUG
if (cmd.fxcmd)
{
Log("(%d.%d) MT2 FX=%02X.%02X.%02X\n", row, ch, cmd.fxcmd, cmd.fxparam1, cmd.fxparam2);
}
#endif
ConvertMT2Command(this, &m[patpos], &cmd);
}
row += rptcount+1;
while (row >= nLines) { row-=nLines; ch++; }
if (ch >= m_nChannels) break;
}
} else
{
const MT2COMMAND *p = (MT2COMMAND *)(lpStream+dwMemPos);
UINT n = 0;
while ((len > sizeof(MT2COMMAND)) && (n < m_nChannels*nLines))
{
ConvertMT2Command(this, m, p);
len -= sizeof(MT2COMMAND);
n++;
p++;
m++;
}
}
}
dwMemPos += wDataLen;
}
// Skip Drum Patterns
if (pdd)
{
#ifdef MT2DEBUG
Log("%d Drum Patterns at offset 0x%08X\n", pdd->wDrumPatterns, dwMemPos);
#endif
for (UINT iDrm=0; iDrm<pdd->wDrumPatterns; iDrm++)
{
if (dwMemPos > dwMemLength-2) return TRUE;
UINT nLines = *(WORD *)(lpStream+dwMemPos);
#ifdef MT2DEBUG
if (nLines != 64) Log("Drum Pattern %d: %d Lines @%04X\n", iDrm, nLines, dwMemPos);
#endif
dwMemPos += 2 + nLines * 32;
}
}
// Automation
if (pfh->fulFlags & 2)
{
#ifdef MT2DEBUG
Log("Automation at offset 0x%08X\n", dwMemPos);
#endif
UINT nAutoCount = m_nChannels;
if (pfh->fulFlags & 0x10) nAutoCount++; // Master Automation
if ((pfh->fulFlags & 0x08) && (pdd)) nAutoCount += 8; // Drums Automation
nAutoCount *= pfh->wPatterns;
for (UINT iAuto=0; iAuto<nAutoCount; iAuto++)
{
if (dwMemPos+12 >= dwMemLength) return TRUE;
const MT2AUTOMATION *pma = (MT2AUTOMATION *)(lpStream+dwMemPos);
dwMemPos += (pfh->wVersion <= 0x201) ? 4 : 8;
for (UINT iEnv=0; iEnv<14; iEnv++)
{
if (pma->dwFlags & (1 << iEnv))
{
#ifdef MT2DEBUG
UINT nPoints = *(DWORD *)(lpStream+dwMemPos);
Log(" Env[%d/%d] %04X @%04X: %d points\n", iAuto, nAutoCount, 1 << iEnv, dwMemPos-8, nPoints);
#endif
dwMemPos += 260;
}
}
}
}
// Load Instruments
#ifdef MT2DEBUG
Log("Loading instruments at offset 0x%08X\n", dwMemPos);
#endif
memset(InstrMap, 0, sizeof(InstrMap));
m_nInstruments = (pfh->wInstruments < MAX_INSTRUMENTS) ? pfh->wInstruments : MAX_INSTRUMENTS-1;
for (UINT iIns=1; iIns<=255; iIns++)
{
if (dwMemPos+36 > dwMemLength) return TRUE;
const MT2INSTRUMENT *pmi = (MT2INSTRUMENT *)(lpStream+dwMemPos);
INSTRUMENTHEADER *penv = NULL;
if (iIns <= m_nInstruments)
{
penv = new INSTRUMENTHEADER;
Headers[iIns] = penv;
if (penv)
{
memset(penv, 0, sizeof(INSTRUMENTHEADER));
memcpy(penv->name, pmi->szName, 32);
penv->nGlobalVol = 64;
penv->nPan = 128;
for (UINT i=0; i<NOTE_MAX; i++)
{
penv->NoteMap[i] = i+1;
}
}
}
#ifdef MT2DEBUG
if (iIns <= pfh->wInstruments) Log(" Instrument #%d at offset %04X: %d bytes\n", iIns, dwMemPos, pmi->dwDataLen);
#endif
if (((LONG)pmi->dwDataLen > 0) && (dwMemPos <= dwMemLength - 40) && (pmi->dwDataLen <= dwMemLength - (dwMemPos + 40)))
{
InstrMap[iIns-1] = pmi;
if (penv)
{
penv->nFadeOut = pmi->wFadeOut;
penv->nNNA = pmi->wNNA & 3;
penv->nDCT = (pmi->wNNA>>8) & 3;
penv->nDNA = (pmi->wNNA>>12) & 3;
MT2ENVELOPE *pehdr[4];
WORD *pedata[4];
if (pfh->wVersion <= 0x201)
{
DWORD dwEnvPos = dwMemPos + sizeof(MT2INSTRUMENT) - 4;
pehdr[0] = (MT2ENVELOPE *)(lpStream+dwEnvPos);
pehdr[1] = (MT2ENVELOPE *)(lpStream+dwEnvPos+8);
pehdr[2] = pehdr[3] = NULL;
pedata[0] = (WORD *)(lpStream+dwEnvPos+16);
pedata[1] = (WORD *)(lpStream+dwEnvPos+16+64);
pedata[2] = pedata[3] = NULL;
} else
{
DWORD dwEnvPos = dwMemPos + sizeof(MT2INSTRUMENT);
for (UINT i=0; i<4; i++)
{
if (pmi->wEnvFlags1 & (1<<i))
{
pehdr[i] = (MT2ENVELOPE *)(lpStream+dwEnvPos);
pedata[i] = (WORD *)pehdr[i]->EnvData;
dwEnvPos += sizeof(MT2ENVELOPE);
} else
{
pehdr[i] = NULL;
pedata[i] = NULL;
}
}
}
// Load envelopes
for (UINT iEnv=0; iEnv<4; iEnv++) if (pehdr[iEnv])
{
const MT2ENVELOPE *pme = pehdr[iEnv];
WORD *pEnvPoints = NULL;
BYTE *pEnvData = NULL;
#ifdef MT2DEBUG
Log(" Env %d.%d @%04X: %d points\n", iIns, iEnv, (UINT)(((BYTE *)pme)-lpStream), pme->nPoints);
#endif
switch(iEnv)
{
// Volume Envelope
case 0:
if (pme->nFlags & 1) penv->dwFlags |= ENV_VOLUME;
if (pme->nFlags & 2) penv->dwFlags |= ENV_VOLSUSTAIN;
if (pme->nFlags & 4) penv->dwFlags |= ENV_VOLLOOP;
penv->nVolEnv = (pme->nPoints > 16) ? 16 : pme->nPoints;
penv->nVolSustainBegin = penv->nVolSustainEnd = pme->nSustainPos;
penv->nVolLoopStart = pme->nLoopStart;
penv->nVolLoopEnd = pme->nLoopEnd;
pEnvPoints = penv->VolPoints;
pEnvData = penv->VolEnv;
break;
// Panning Envelope
case 1:
if (pme->nFlags & 1) penv->dwFlags |= ENV_PANNING;
if (pme->nFlags & 2) penv->dwFlags |= ENV_PANSUSTAIN;
if (pme->nFlags & 4) penv->dwFlags |= ENV_PANLOOP;
penv->nPanEnv = (pme->nPoints > 16) ? 16 : pme->nPoints;
penv->nPanSustainBegin = penv->nPanSustainEnd = pme->nSustainPos;
penv->nPanLoopStart = pme->nLoopStart;
penv->nPanLoopEnd = pme->nLoopEnd;
pEnvPoints = penv->PanPoints;
pEnvData = penv->PanEnv;
break;
// Pitch/Filter envelope
default:
if (pme->nFlags & 1) penv->dwFlags |= (iEnv==3) ? (ENV_PITCH|ENV_FILTER) : ENV_PITCH;
if (pme->nFlags & 2) penv->dwFlags |= ENV_PITCHSUSTAIN;
if (pme->nFlags & 4) penv->dwFlags |= ENV_PITCHLOOP;
penv->nPitchEnv = (pme->nPoints > 16) ? 16 : pme->nPoints;
penv->nPitchSustainBegin = penv->nPitchSustainEnd = pme->nSustainPos;
penv->nPitchLoopStart = pme->nLoopStart;
penv->nPitchLoopEnd = pme->nLoopEnd;
pEnvPoints = penv->PitchPoints;
pEnvData = penv->PitchEnv;
}
// Envelope data
if ((pEnvPoints) && (pEnvData) && (pedata[iEnv]))
{
WORD *psrc = pedata[iEnv];
for (UINT i=0; i<16; i++)
{
pEnvPoints[i] = psrc[i*2];
pEnvData[i] = (BYTE)psrc[i*2+1];
}
}
}
}
dwMemPos += pmi->dwDataLen + 36;
if (pfh->wVersion > 0x201) dwMemPos += 4; // ?
} else
{
dwMemPos += 36;
}
}
#ifdef MT2DEBUG
Log("Loading samples at offset 0x%08X\n", dwMemPos);
#endif
memset(SampleMap, 0, sizeof(SampleMap));
m_nSamples = (pfh->wSamples < MAX_SAMPLES) ? pfh->wSamples : MAX_SAMPLES-1;
for (UINT iSmp=1; iSmp<=256; iSmp++)
{
if (dwMemPos+36 > dwMemLength) return TRUE;
const MT2SAMPLE *pms = (MT2SAMPLE *)(lpStream+dwMemPos);
#ifdef MT2DEBUG
if (iSmp <= m_nSamples) Log(" Sample #%d at offset %04X: %d bytes\n", iSmp, dwMemPos, pms->dwDataLen);
#endif
if (iSmp < MAX_SAMPLES)
{
memcpy(m_szNames[iSmp], pms->szName, 32);
}
if (pms->dwDataLen > 0)
{
SampleMap[iSmp-1] = pms;
if (iSmp < MAX_SAMPLES)
{
MODINSTRUMENT *psmp = &Ins[iSmp];
psmp->nGlobalVol = 64;
psmp->nVolume = (pms->wVolume >> 7);
psmp->nPan = (pms->nPan == 0x80) ? 128 : (pms->nPan^0x80);
psmp->nLength = pms->dwLength;
psmp->nC4Speed = pms->dwFrequency;
psmp->nLoopStart = pms->dwLoopStart;
psmp->nLoopEnd = pms->dwLoopEnd;
FrequencyToTranspose(psmp);
psmp->RelativeTone -= pms->nBaseNote - 49;
psmp->nC4Speed = TransposeToFrequency(psmp->RelativeTone, psmp->nFineTune);
if (pms->nQuality == 2) { psmp->uFlags |= CHN_16BIT; psmp->nLength >>= 1; }
if (pms->nChannels == 2) { psmp->nLength >>= 1; }
if (pms->nLoop == 1) psmp->uFlags |= CHN_LOOP;
if (pms->nLoop == 2) psmp->uFlags |= CHN_LOOP|CHN_PINGPONGLOOP;
}
dwMemPos += pms->dwDataLen + 36;
} else
{
dwMemPos += 36;
}
}
#ifdef MT2DEBUG
Log("Loading groups at offset 0x%08X\n", dwMemPos);
#endif
for (UINT iMap=0; iMap<255; iMap++) if (InstrMap[iMap])
{
if (dwMemPos+8 > dwMemLength) return TRUE;
const MT2INSTRUMENT *pmi = InstrMap[iMap];
INSTRUMENTHEADER *penv = NULL;
if (iMap<m_nInstruments) penv = Headers[iMap+1];
for (UINT iGrp=0; iGrp<pmi->wSamples; iGrp++)
{
if (penv)
{
const MT2GROUP *pmg = (MT2GROUP *)(lpStream+dwMemPos);
for (UINT i=0; i<96; i++)
{
if (pmi->GroupsMapping[i] == iGrp)
{
UINT nSmp = pmg->nSmpNo+1;
penv->Keyboard[i+12] = (BYTE)nSmp;
if (nSmp <= m_nSamples)
{
Ins[nSmp].nVibType = pmi->bVibType;
Ins[nSmp].nVibSweep = pmi->bVibSweep;
Ins[nSmp].nVibDepth = pmi->bVibDepth;
Ins[nSmp].nVibRate = pmi->bVibRate;
}
}
}
}
dwMemPos += 8;
}
}
#ifdef MT2DEBUG
Log("Loading sample data at offset 0x%08X\n", dwMemPos);
#endif
for (UINT iData=0; iData<256; iData++) if ((iData < m_nSamples) && (SampleMap[iData]))
{
const MT2SAMPLE *pms = SampleMap[iData];
MODINSTRUMENT *psmp = &Ins[iData+1];
if (!(pms->nFlags & 5))
{
if (psmp->nLength > 0)
{
#ifdef MT2DEBUG
Log(" Reading sample #%d at offset 0x%04X (len=%d)\n", iData+1, dwMemPos, psmp->nLength);
#endif
UINT rsflags;
if (pms->nChannels == 2)
rsflags = (psmp->uFlags & CHN_16BIT) ? RS_STPCM16D : RS_STPCM8D;
else
rsflags = (psmp->uFlags & CHN_16BIT) ? RS_PCM16D : RS_PCM8D;
dwMemPos += ReadSample(psmp, rsflags, (LPCSTR)(lpStream+dwMemPos), dwMemLength-dwMemPos);
}
} else
if (dwMemPos+4 < dwMemLength)
{
UINT nNameLen = *(DWORD *)(lpStream+dwMemPos);
dwMemPos += nNameLen + 16;
}
if (dwMemPos+4 >= dwMemLength) break;
}
return TRUE;
}

@ -0,0 +1,168 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
//////////////////////////////////////////////////////////
// MTM file support (import only)
#pragma pack(1)
typedef struct tagMTMSAMPLE
{
char samplename[22]; // changed from CHAR
DWORD length;
DWORD reppos;
DWORD repend;
CHAR finetune;
BYTE volume;
BYTE attribute;
} MTMSAMPLE;
typedef struct tagMTMHEADER
{
char id[4]; // MTM file marker + version // changed from CHAR
char songname[20]; // ASCIIZ songname // changed from CHAR
WORD numtracks; // number of tracks saved
BYTE lastpattern; // last pattern number saved
BYTE lastorder; // last order number to play (songlength-1)
WORD commentsize; // length of comment field
BYTE numsamples; // number of samples saved
BYTE attribute; // attribute byte (unused)
BYTE beatspertrack;
BYTE numchannels; // number of channels used
BYTE panpos[32]; // voice pan positions
} MTMHEADER;
#pragma pack()
BOOL CSoundFile::ReadMTM(LPCBYTE lpStream, DWORD dwMemLength)
//-----------------------------------------------------------
{
MTMHEADER *pmh = (MTMHEADER *)lpStream;
DWORD dwMemPos = 66;
if ((!lpStream) || (dwMemLength < 0x100)) return FALSE;
if ((strncmp(pmh->id, "MTM", 3)) || (pmh->numchannels > 32)
|| (pmh->numsamples >= MAX_SAMPLES) || (!pmh->numsamples)
|| (!pmh->numtracks) || (!pmh->numchannels)
|| (!pmh->lastpattern) || (pmh->lastpattern >= MAX_PATTERNS))
return FALSE;
strncpy(m_szNames[0], pmh->songname, 20);
m_szNames[0][20] = 0;
if (dwMemPos + 37*pmh->numsamples + 128 + 192*pmh->numtracks
+ 64 * (pmh->lastpattern+1) + pmh->commentsize >= dwMemLength)
return FALSE;
m_nType = MOD_TYPE_MTM;
m_nSamples = pmh->numsamples;
m_nChannels = pmh->numchannels;
// Reading instruments
for (UINT i=1; i<=m_nSamples; i++)
{
MTMSAMPLE *pms = (MTMSAMPLE *)(lpStream + dwMemPos);
strncpy(m_szNames[i], pms->samplename, 22);
m_szNames[i][22] = 0;
Ins[i].nVolume = pms->volume << 2;
Ins[i].nGlobalVol = 64;
DWORD len = pms->length;
if ((len > 4) && (len <= MAX_SAMPLE_LENGTH))
{
Ins[i].nLength = len;
Ins[i].nLoopStart = pms->reppos;
Ins[i].nLoopEnd = pms->repend;
if (Ins[i].nLoopEnd > Ins[i].nLength)
Ins[i].nLoopEnd = Ins[i].nLength;
if (Ins[i].nLoopStart + 4 >= Ins[i].nLoopEnd)
Ins[i].nLoopStart = Ins[i].nLoopEnd = 0;
if (Ins[i].nLoopEnd) Ins[i].uFlags |= CHN_LOOP;
Ins[i].nFineTune = MOD2XMFineTune(pms->finetune);
if (pms->attribute & 0x01)
{
Ins[i].uFlags |= CHN_16BIT;
Ins[i].nLength >>= 1;
Ins[i].nLoopStart >>= 1;
Ins[i].nLoopEnd >>= 1;
}
Ins[i].nPan = 128;
}
dwMemPos += 37;
}
// Setting Channel Pan Position
for (UINT ich=0; ich<m_nChannels; ich++)
{
ChnSettings[ich].nPan = ((pmh->panpos[ich] & 0x0F) << 4) + 8;
ChnSettings[ich].nVolume = 64;
}
// Reading pattern order
memcpy(Order, lpStream + dwMemPos, pmh->lastorder+1);
dwMemPos += 128;
// Reading Patterns
LPCBYTE pTracks = lpStream + dwMemPos;
dwMemPos += 192 * pmh->numtracks;
LPWORD pSeq = (LPWORD)(lpStream + dwMemPos);
for (UINT pat=0; pat<=pmh->lastpattern; pat++)
{
PatternSize[pat] = 64;
if ((Patterns[pat] = AllocatePattern(64, m_nChannels)) == NULL) break;
for (UINT n=0; n<32; n++) if ((pSeq[n]) && (pSeq[n] <= pmh->numtracks) && (n < m_nChannels))
{
LPCBYTE p = pTracks + 192 * (pSeq[n]-1);
MODCOMMAND *m = Patterns[pat] + n;
for (UINT i=0; i<64; i++, m+=m_nChannels, p+=3)
{
if (p[0] & 0xFC) m->note = (p[0] >> 2) + 37;
m->instr = ((p[0] & 0x03) << 4) | (p[1] >> 4);
UINT cmd = p[1] & 0x0F;
UINT param = p[2];
if (cmd == 0x0A)
{
if (param & 0xF0) param &= 0xF0; else param &= 0x0F;
}
m->command = cmd;
m->param = param;
if ((cmd) || (param)) ConvertModCommand(m);
}
}
pSeq += 32;
}
dwMemPos += 64*(pmh->lastpattern+1);
if ((pmh->commentsize) && (dwMemPos + pmh->commentsize < dwMemLength))
{
UINT n = pmh->commentsize;
m_lpszSongComments = new char[n+1];
if (m_lpszSongComments)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos, n);
m_lpszSongComments[n] = 0;
for (UINT i=0; i<n; i++)
{
if (!m_lpszSongComments[i])
{
m_lpszSongComments[i] = ((i+1) % 40) ? 0x20 : 0x0D;
}
}
}
}
dwMemPos += pmh->commentsize;
// Reading Samples
for (UINT ismp=1; ismp<=m_nSamples; ismp++)
{
if (dwMemPos >= dwMemLength) break;
dwMemPos += ReadSample(&Ins[ismp], (Ins[ismp].uFlags & CHN_16BIT) ? RS_PCM16U : RS_PCM8U,
(LPSTR)(lpStream + dwMemPos), dwMemLength - dwMemPos);
}
m_nMinPeriod = 64;
m_nMaxPeriod = 32767;
return TRUE;
}

@ -0,0 +1,197 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
//////////////////////////////////////////////
// Oktalyzer (OKT) module loader //
//////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
typedef struct OKTFILEHEADER
{
DWORD okta; // "OKTA"
DWORD song; // "SONG"
DWORD cmod; // "CMOD"
DWORD fixed8;
BYTE chnsetup[8];
DWORD samp; // "SAMP"
DWORD samplen;
} OKTFILEHEADER;
typedef struct OKTSAMPLE
{
CHAR name[20];
DWORD length;
WORD loopstart;
WORD looplen;
BYTE pad1;
BYTE volume;
BYTE pad2;
BYTE pad3;
} OKTSAMPLE;
BOOL CSoundFile::ReadOKT(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const OKTFILEHEADER *pfh = (OKTFILEHEADER *)lpStream;
DWORD dwMemPos = sizeof(OKTFILEHEADER);
UINT nsamples = 0, npatterns = 0, norders = 0;
if ((!lpStream) || (dwMemLength < 1024)) return FALSE;
if ((pfh->okta != 0x41544B4F) || (pfh->song != 0x474E4F53)
|| (pfh->cmod != 0x444F4D43) || (pfh->chnsetup[0]) || (pfh->chnsetup[2])
|| (pfh->chnsetup[4]) || (pfh->chnsetup[6]) || (pfh->fixed8 != 0x08000000)
|| (pfh->samp != 0x504D4153)) return FALSE;
m_nType = MOD_TYPE_OKT;
m_nChannels = 4 + pfh->chnsetup[1] + pfh->chnsetup[3] + pfh->chnsetup[5] + pfh->chnsetup[7];
if (m_nChannels > MAX_CHANNELS) m_nChannels = MAX_CHANNELS;
nsamples = bswapBE32(pfh->samplen) >> 5;
m_nSamples = nsamples;
if (m_nSamples >= MAX_SAMPLES) m_nSamples = MAX_SAMPLES-1;
// Reading samples
for (UINT smp=1; smp <= nsamples; smp++)
{
if (dwMemPos >= dwMemLength) return TRUE;
if (smp < MAX_SAMPLES)
{
OKTSAMPLE *psmp = (OKTSAMPLE *)(lpStream + dwMemPos);
MODINSTRUMENT *pins = &Ins[smp];
memcpy(m_szNames[smp], psmp->name, 20);
pins->uFlags = 0;
pins->nLength = bswapBE32(psmp->length) & ~1;
pins->nLoopStart = bswapBE16(psmp->loopstart);
pins->nLoopEnd = pins->nLoopStart + bswapBE16(psmp->looplen);
if (pins->nLoopStart + 2 < pins->nLoopEnd) pins->uFlags |= CHN_LOOP;
pins->nGlobalVol = 64;
pins->nVolume = psmp->volume << 2;
pins->nC4Speed = 8363;
}
dwMemPos += sizeof(OKTSAMPLE);
}
// SPEE
if (dwMemPos >= dwMemLength) return TRUE;
if (*((DWORD *)(lpStream + dwMemPos)) == 0x45455053)
{
m_nDefaultSpeed = lpStream[dwMemPos+9];
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
}
// SLEN
if (dwMemPos >= dwMemLength) return TRUE;
if (*((DWORD *)(lpStream + dwMemPos)) == 0x4E454C53)
{
npatterns = lpStream[dwMemPos+9];
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
}
// PLEN
if (dwMemPos >= dwMemLength) return TRUE;
if (*((DWORD *)(lpStream + dwMemPos)) == 0x4E454C50)
{
norders = lpStream[dwMemPos+9];
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
}
// PATT
if (dwMemPos >= dwMemLength) return TRUE;
if (*((DWORD *)(lpStream + dwMemPos)) == 0x54544150)
{
UINT orderlen = norders;
if (orderlen >= MAX_ORDERS) orderlen = MAX_ORDERS-1;
for (UINT i=0; i<orderlen; i++) Order[i] = lpStream[dwMemPos+10+i];
for (UINT j=orderlen; j>1; j--) { if (Order[j-1]) break; Order[j-1] = 0xFF; }
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
}
// PBOD
UINT npat = 0;
while ((dwMemPos+10 < dwMemLength) && (*((DWORD *)(lpStream + dwMemPos)) == 0x444F4250))
{
DWORD dwPos = dwMemPos + 10;
UINT rows = lpStream[dwMemPos+9];
if (!rows) rows = 64;
if (npat < MAX_PATTERNS)
{
if ((Patterns[npat] = AllocatePattern(rows, m_nChannels)) == NULL) return TRUE;
MODCOMMAND *m = Patterns[npat];
PatternSize[npat] = rows;
UINT imax = m_nChannels*rows;
for (UINT i=0; i<imax; i++, m++, dwPos+=4)
{
if (dwPos+4 > dwMemLength) break;
const BYTE *p = lpStream+dwPos;
UINT note = p[0];
if (note)
{
m->note = note + 48;
m->instr = p[1] + 1;
}
UINT command = p[2];
UINT param = p[3];
m->param = param;
switch(command)
{
// 0: no effect
case 0:
break;
// 1: Portamento Up
case 1:
case 17:
case 30:
if (param) m->command = CMD_PORTAMENTOUP;
break;
// 2: Portamento Down
case 2:
case 13:
case 21:
if (param) m->command = CMD_PORTAMENTODOWN;
break;
// 10: Arpeggio
case 10:
case 11:
case 12:
m->command = CMD_ARPEGGIO;
break;
// 15: Filter
case 15:
m->command = CMD_MODCMDEX;
m->param = param & 0x0F;
break;
// 25: Position Jump
case 25:
m->command = CMD_POSITIONJUMP;
break;
// 28: Set Speed
case 28:
m->command = CMD_SPEED;
break;
// 31: Volume Control
case 31:
if (param <= 0x40) m->command = CMD_VOLUME; else
if (param <= 0x50) { m->command = CMD_VOLUMESLIDE; m->param &= 0x0F; if (!m->param) m->param = 0x0F; } else
if (param <= 0x60) { m->command = CMD_VOLUMESLIDE; m->param = (param & 0x0F) << 4; if (!m->param) m->param = 0xF0; } else
if (param <= 0x70) { m->command = CMD_MODCMDEX; m->param = 0xB0 | (param & 0x0F); if (!(param & 0x0F)) m->param = 0xBF; } else
if (param <= 0x80) { m->command = CMD_MODCMDEX; m->param = 0xA0 | (param & 0x0F); if (!(param & 0x0F)) m->param = 0xAF; }
break;
}
}
}
npat++;
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
}
// SBOD
UINT nsmp = 1;
while ((dwMemPos+10 < dwMemLength) && (*((DWORD *)(lpStream + dwMemPos)) == 0x444F4253))
{
if (nsmp < MAX_SAMPLES) ReadSample(&Ins[nsmp], RS_PCM8S, (LPSTR)(lpStream+dwMemPos+8), dwMemLength-dwMemPos-8);
dwMemPos += bswapBE32(*((DWORD *)(lpStream + dwMemPos + 4))) + 8;
nsmp++;
}
return TRUE;
}

File diff suppressed because it is too large Load Diff

@ -0,0 +1,25 @@
#ifndef LOAD_PAT_H
#define LOAD_PAT_H
#ifdef __cplusplus
extern "C" {
#endif
void pat_init_patnames(void);
void pat_resetsmp(void);
int pat_numinstr(void);
int pat_numsmp(void);
int pat_smptogm(int smp);
int pat_gmtosmp(int gm);
int pat_gm_drumnr(int n);
int pat_gm_drumnote(int n);
const char *pat_gm_name(int gm);
int pat_modnote(int midinote);
int pat_smplooped(int smp);
BOOL PAT_Load_Instruments(void *c);
#ifdef __cplusplus
}
#endif
#endif

@ -0,0 +1,864 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
///////////////////////////////////////////////////
//
// PSM module loader
//
///////////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#define PSM_LOG
#define PSM_ID_NEW 0x204d5350
#define PSM_ID_OLD 0xfe4d5350
#define IFFID_FILE 0x454c4946
#define IFFID_TITL 0x4c544954
#define IFFID_SDFT 0x54464453
#define IFFID_PBOD 0x444f4250
#define IFFID_SONG 0x474e4f53
#define IFFID_PATT 0x54544150
#define IFFID_DSMP 0x504d5344
#define IFFID_OPLH 0x484c504f
#pragma pack(1)
typedef struct _PSMCHUNK
{
DWORD id;
DWORD len;
DWORD listid;
} PSMCHUNK;
void swap_PSMCHUNK(PSMCHUNK* p){
p->id = bswapLE32(p->id);
p->len = bswapLE32(p->len);
p->listid = bswapLE32(p->listid);
}
typedef struct _PSMSONGHDR
{
CHAR songname[8]; // "MAINSONG"
BYTE reserved1;
BYTE reserved2;
BYTE channels;
} PSMSONGHDR;
typedef struct _PSMPATTERN
{
DWORD size;
DWORD name;
WORD rows;
WORD reserved1;
BYTE data[4];
} PSMPATTERN;
void swap_PSMPATTERN(PSMPATTERN* p){
p->size = bswapLE32(p->size);
p->name = bswapLE32(p->name);
p->rows = bswapLE16(p->rows);
}
typedef struct _PSMSAMPLE
{
BYTE flags;
CHAR songname[8];
DWORD smpid;
CHAR samplename[34];
DWORD reserved1;
BYTE reserved2;
BYTE insno;
BYTE reserved3;
DWORD length;
DWORD loopstart;
DWORD loopend;
WORD reserved4;
BYTE defvol;
DWORD reserved5;
DWORD samplerate;
BYTE reserved6[19];
} PSMSAMPLE;
void swap_PSMSAMPLE(PSMSAMPLE* p){
p->smpid = bswapLE32(p->smpid);
p->length = bswapLE32(p->length);
p->loopstart = bswapLE32(p->loopstart);
p->loopend = bswapLE32(p->loopend);
p->samplerate = bswapLE32(p->samplerate);
}
#pragma pack()
BOOL CSoundFile::ReadPSM(LPCBYTE lpStream, DWORD dwMemLength)
//-----------------------------------------------------------
{
PSMCHUNK *pfh = (PSMCHUNK *)lpStream;
DWORD dwMemPos, dwSongPos;
DWORD smpnames[MAX_SAMPLES];
DWORD patptrs[MAX_PATTERNS];
BYTE samplemap[MAX_SAMPLES];
UINT nPatterns;
// Swap chunk
swap_PSMCHUNK(pfh);
// Chunk0: "PSM ",filesize,"FILE"
if (dwMemLength < 256) return FALSE;
if (pfh->id == PSM_ID_OLD)
{
#ifdef PSM_LOG
Log("Old PSM format not supported\n");
#endif
return FALSE;
}
if ((pfh->id != PSM_ID_NEW) || (pfh->len+12 > dwMemLength) || (pfh->listid != IFFID_FILE)) return FALSE;
m_nType = MOD_TYPE_PSM;
m_nChannels = 16;
m_nSamples = 0;
nPatterns = 0;
dwMemPos = 12;
dwSongPos = 0;
for (UINT iChPan=0; iChPan<16; iChPan++)
{
UINT pan = (((iChPan & 3) == 1) || ((iChPan&3)==2)) ? 0xC0 : 0x40;
ChnSettings[iChPan].nPan = pan;
}
while (dwMemPos+8 < dwMemLength)
{
PSMCHUNK *pchunk = (PSMCHUNK *)(lpStream+dwMemPos);
swap_PSMCHUNK(pchunk);
if ((pchunk->len >= dwMemLength - 8) || (dwMemPos + pchunk->len + 8 > dwMemLength)) break;
dwMemPos += 8;
PUCHAR pdata = (PUCHAR)(lpStream+dwMemPos);
ULONG len = pchunk->len;
if (len) switch(pchunk->id)
{
// "TITL": Song title
case IFFID_TITL:
if (!pdata[0]) { pdata++; len--; }
memcpy(m_szNames[0], pdata, (len>31) ? 31 : len);
m_szNames[0][31] = 0;
break;
// "PBOD": Pattern
case IFFID_PBOD:
if ((len >= 12) && (nPatterns < MAX_PATTERNS))
{
patptrs[nPatterns++] = dwMemPos-8;
}
break;
// "SONG": Song description
case IFFID_SONG:
if ((len >= sizeof(PSMSONGHDR)+8) && (!dwSongPos))
{
dwSongPos = dwMemPos - 8;
}
break;
// "DSMP": Sample Data
case IFFID_DSMP:
if ((len >= sizeof(PSMSAMPLE)) && (m_nSamples+1 < MAX_SAMPLES))
{
m_nSamples++;
MODINSTRUMENT *pins = &Ins[m_nSamples];
PSMSAMPLE *psmp = (PSMSAMPLE *)pdata;
swap_PSMSAMPLE(psmp);
smpnames[m_nSamples] = psmp->smpid;
memcpy(m_szNames[m_nSamples], psmp->samplename, 31);
m_szNames[m_nSamples][31] = 0;
samplemap[m_nSamples-1] = (BYTE)m_nSamples;
// Init sample
pins->nGlobalVol = 0x40;
pins->nC4Speed = psmp->samplerate;
pins->nLength = psmp->length;
pins->nLoopStart = psmp->loopstart;
pins->nLoopEnd = psmp->loopend;
pins->nPan = 128;
pins->nVolume = (psmp->defvol+1) * 2;
pins->uFlags = (psmp->flags & 0x80) ? CHN_LOOP : 0;
if (pins->nLoopStart > 0) pins->nLoopStart--;
// Point to sample data
pdata += 0x60;
len -= 0x60;
// Load sample data
if ((pins->nLength > 3) && (len > 3))
{
ReadSample(pins, RS_PCM8D, (LPCSTR)pdata, len);
} else
{
pins->nLength = 0;
}
}
break;
#if 0
default:
{
CHAR s[8], s2[64];
*(DWORD *)s = pchunk->id;
s[4] = 0;
wsprintf(s2, "%s: %4d bytes @ %4d\n", s, pchunk->len, dwMemPos);
OutputDebugString(s2);
}
#endif
}
dwMemPos += pchunk->len;
}
// Step #1: convert song structure
PSMSONGHDR *pSong = (PSMSONGHDR *)(lpStream+dwSongPos+8);
if ((!dwSongPos) || (pSong->channels < 2) || (pSong->channels > 32)) return TRUE;
m_nChannels = pSong->channels;
// Valid song header -> convert attached chunks
{
DWORD dwSongEnd = dwSongPos + 8 + *(DWORD *)(lpStream+dwSongPos+4);
dwMemPos = dwSongPos + 8 + 11; // sizeof(PSMCHUNK)+sizeof(PSMSONGHDR)
while (dwMemPos + 8 < dwSongEnd)
{
PSMCHUNK *pchunk = (PSMCHUNK *)(lpStream+dwMemPos);
swap_PSMCHUNK(pchunk);
dwMemPos += 8;
if ((pchunk->len > dwSongEnd) || (dwMemPos + pchunk->len > dwSongEnd)) break;
PUCHAR pdata = (PUCHAR)(lpStream+dwMemPos);
ULONG len = pchunk->len;
switch(pchunk->id)
{
case IFFID_OPLH:
if (len >= 0x20)
{
UINT pos = len - 3;
while (pos > 5)
{
BOOL bFound = FALSE;
pos -= 5;
DWORD dwName = *(DWORD *)(pdata+pos);
for (UINT i=0; i<nPatterns; i++)
{
DWORD dwPatName = ((PSMPATTERN *)(lpStream+patptrs[i]+8))->name;
if (dwName == dwPatName)
{
bFound = TRUE;
break;
}
}
if ((!bFound) && (pdata[pos+1] > 0) && (pdata[pos+1] <= 0x10)
&& (pdata[pos+3] > 0x40) && (pdata[pos+3] < 0xC0))
{
m_nDefaultSpeed = pdata[pos+1];
m_nDefaultTempo = pdata[pos+3];
break;
}
}
UINT iOrd = 0;
while ((pos+5<len) && (iOrd < MAX_ORDERS))
{
DWORD dwName = *(DWORD *)(pdata+pos);
for (UINT i=0; i<nPatterns; i++)
{
DWORD dwPatName = ((PSMPATTERN *)(lpStream+patptrs[i]+8))->name;
if (dwName == dwPatName)
{
Order[iOrd++] = i;
break;
}
}
pos += 5;
}
}
break;
}
dwMemPos += pchunk->len;
}
}
// Step #2: convert patterns
for (UINT nPat=0; nPat<nPatterns; nPat++)
{
PSMPATTERN *pPsmPat = (PSMPATTERN *)(lpStream+patptrs[nPat]+8);
swap_PSMPATTERN(pPsmPat);
ULONG len = *(DWORD *)(lpStream+patptrs[nPat]+4) - 12;
UINT nRows = pPsmPat->rows;
if (len > pPsmPat->size) len = pPsmPat->size;
if ((nRows < 64) || (nRows > 256)) nRows = 64;
PatternSize[nPat] = nRows;
if ((Patterns[nPat] = AllocatePattern(nRows, m_nChannels)) == NULL) break;
MODCOMMAND *m = Patterns[nPat];
BYTE *p = pPsmPat->data;
MODCOMMAND *sp, dummy;
UINT pos = 0;
UINT row = 0;
UINT rowlim;
#ifdef PSM_LOG
Log("Pattern %d at offset 0x%04X\n", nPat, (DWORD)(p - (BYTE *)lpStream));
#endif
UINT flags, ch;
rowlim = bswapLE16(pPsmPat->reserved1)-2;
while ((row < nRows) && (pos+1 < len))
{
if ((pos+1) >= rowlim) {
pos = rowlim;
rowlim = (((int)p[pos+1])<<8)
| ((int)p[pos+0]);
m += m_nChannels;
row++;
rowlim += pos;
pos += 2;
}
flags = p[pos++];
ch = p[pos++];
if (ch >= m_nChannels) {
sp = &dummy;
} else {
sp = &m[ch];
}
// Note + Instr
if ((flags & 0x80) && (pos+1 < len))
{
UINT note = p[pos++];
note = (note>>4)*12+(note&0x0f)+12+1;
if (note > 0x80) note = 0;
m[ch].note = note;
}
if ((flags & 0x40) && (pos+1 < len))
{
UINT nins = p[pos++];
#ifdef PSM_LOG
//if (!nPat) Log("note+ins: %02X.%02X\n", note, nins);
if ((!nPat) && (nins >= m_nSamples)) Log("WARNING: invalid instrument number (%d)\n", nins);
#endif
m[ch].instr = samplemap[nins];
}
// Volume
if ((flags & 0x20) && (pos < len))
{
m[ch].volcmd = VOLCMD_VOLUME;
m[ch].vol = p[pos++] / 2;
}
// Effect
if ((flags & 0x10) && (pos+1 < len))
{
UINT command = p[pos++];
UINT param = p[pos++];
// Convert effects
switch(command)
{
// 01: fine volslide up
case 0x01: command = CMD_VOLUMESLIDE; param |= 0x0f;
if (param == 15) param=31;
break;
// 02: volslide up
case 0x02: command = CMD_VOLUMESLIDE; param>>=1; param<<=4; break;
// 03: fine volslide down
case 0x03: command = CMD_VOLUMESLIDE; param>>=4; param |= 0xf0;
if (param == 240) param=241;
break;
// 04: fine volslide down
case 0x04: command = CMD_VOLUMESLIDE; param>>=4; param |= 0xf0; break;
// 0C: portamento up
case 0x0C: command = CMD_PORTAMENTOUP; param = (param+1)/2; break;
// 0E: portamento down
case 0x0E: command = CMD_PORTAMENTODOWN; param = (param+1)/2; break;
// 0F: tone portamento
case 0x0F: command = CMD_TONEPORTAMENTO; param = param/4; break;
// 15: vibrato
case 0x15: command = CMD_VIBRATO; break;
// 29: sample offset
case 0x29: pos += 2; break;
// 2A: retrigger note
case 0x2A: command = CMD_RETRIG; break;
// 33: Position Jump
case 0x33: command = CMD_POSITIONJUMP; break;
// 34: Pattern break
case 0x34: command = CMD_PATTERNBREAK; break;
// 3D: speed
case 0x3D: command = CMD_SPEED; break;
// 3E: tempo
case 0x3E: command = CMD_TEMPO; break;
// Unknown
default:
#ifdef PSM_LOG
Log("Unknown PSM effect pat=%d row=%d ch=%d: %02X.%02X\n", nPat, row, ch, command, param);
#endif
command = param = 0;
}
m[ch].command = (BYTE)command;
m[ch].param = (BYTE)param;
}
}
#ifdef PSM_LOG
if (pos < len)
{
Log("Pattern %d: %d/%d[%d] rows (%d bytes) -> %d bytes left\n", nPat, row, nRows, pPsmPat->rows, pPsmPat->size, len-pos);
}
#endif
}
// Done (finally!)
return TRUE;
}
//////////////////////////////////////////////////////////////
//
// PSM Old Format
//
/*
CONST
c_PSM_MaxOrder = $FF;
c_PSM_MaxSample = $FF;
c_PSM_MaxChannel = $0F;
TYPE
PPSM_Header = ^TPSM_Header;
TPSM_Header = RECORD
PSM_Sign : ARRAY[01..04] OF CHAR; { PSM + #254 }
PSM_SongName : ARRAY[01..58] OF CHAR;
PSM_Byte00 : BYTE;
PSM_Byte1A : BYTE;
PSM_Unknown00 : BYTE;
PSM_Unknown01 : BYTE;
PSM_Unknown02 : BYTE;
PSM_Speed : BYTE;
PSM_Tempo : BYTE;
PSM_Unknown03 : BYTE;
PSM_Unknown04 : WORD;
PSM_OrderLength : WORD;
PSM_PatternNumber : WORD;
PSM_SampleNumber : WORD;
PSM_ChannelNumber : WORD;
PSM_ChannelUsed : WORD;
PSM_OrderPosition : LONGINT;
PSM_ChannelSettingPosition : LONGINT;
PSM_PatternPosition : LONGINT;
PSM_SamplePosition : LONGINT;
{ *** perhaps there are some more infos in a larger header,
but i have not decoded it and so it apears here NOT }
END;
PPSM_Sample = ^TPSM_Sample;
TPSM_Sample = RECORD
PSM_SampleFileName : ARRAY[01..12] OF CHAR;
PSM_SampleByte00 : BYTE;
PSM_SampleName : ARRAY[01..22] OF CHAR;
PSM_SampleUnknown00 : ARRAY[01..02] OF BYTE;
PSM_SamplePosition : LONGINT;
PSM_SampleUnknown01 : ARRAY[01..04] OF BYTE;
PSM_SampleNumber : BYTE;
PSM_SampleFlags : WORD;
PSM_SampleLength : LONGINT;
PSM_SampleLoopBegin : LONGINT;
PSM_SampleLoopEnd : LONGINT;
PSM_Unknown03 : BYTE;
PSM_SampleVolume : BYTE;
PSM_SampleC5Speed : WORD;
END;
PPSM_SampleList = ^TPSM_SampleList;
TPSM_SampleList = ARRAY[01..c_PSM_MaxSample] OF TPSM_Sample;
PPSM_Order = ^TPSM_Order;
TPSM_Order = ARRAY[00..c_PSM_MaxOrder] OF BYTE;
PPSM_ChannelSettings = ^TPSM_ChannelSettings;
TPSM_ChannelSettings = ARRAY[00..c_PSM_MaxChannel] OF BYTE;
CONST
PSM_NotesInPattern : BYTE = $00;
PSM_ChannelInPattern : BYTE = $00;
CONST
c_PSM_SetSpeed = 60;
FUNCTION PSM_Size(FileName : STRING;FilePosition : LONGINT) : LONGINT;
BEGIN
END;
PROCEDURE PSM_UnpackPattern(VAR Source,Destination;PatternLength : WORD);
VAR
Witz : ARRAY[00..04] OF WORD;
I1,I2 : WORD;
I3,I4 : WORD;
TopicalByte : ^BYTE;
Pattern : PUnpackedPattern;
ChannelP : BYTE;
NoteP : BYTE;
InfoByte : BYTE;
CodeByte : BYTE;
InfoWord : WORD;
Effect : BYTE;
Opperand : BYTE;
Panning : BYTE;
Volume : BYTE;
PrevInfo : BYTE;
InfoIndex : BYTE;
BEGIN
Pattern := @Destination;
TopicalByte := @Source;
{ *** Initialize patttern }
FOR I2 := 0 TO c_Maximum_NoteIndex DO
FOR I3 := 0 TO c_Maximum_ChannelIndex DO
BEGIN
Pattern^[I2,I3,c_Pattern_NoteIndex] := $FF;
Pattern^[I2,I3,c_Pattern_SampleIndex] := $00;
Pattern^[I2,I3,c_Pattern_VolumeIndex] := $FF;
Pattern^[I2,I3,c_Pattern_PanningIndex] := $FF;
Pattern^[I2,I3,c_Pattern_EffectIndex] := $00;
Pattern^[I2,I3,c_Pattern_OpperandIndex] := $00;
END;
{ *** Byte-pointer on first pattern-entry }
ChannelP := $00;
NoteP := $00;
InfoByte := $00;
PrevInfo := $00;
InfoIndex := $02;
{ *** read notes in pattern }
PSM_NotesInPattern := TopicalByte^; INC(TopicalByte); DEC(PatternLength); INC(InfoIndex);
PSM_ChannelInPattern := TopicalByte^; INC(TopicalByte); DEC(PatternLength); INC(InfoIndex);
{ *** unpack pattern }
WHILE (INTEGER(PatternLength) > 0) AND (NoteP < c_Maximum_NoteIndex) DO
BEGIN
{ *** Read info-byte }
InfoByte := TopicalByte^; INC(TopicalByte); DEC(PatternLength); INC(InfoIndex);
IF InfoByte <> $00 THEN
BEGIN
ChannelP := InfoByte AND $0F;
IF InfoByte AND 128 = 128 THEN { note and sample }
BEGIN
{ *** read note }
CodeByte := TopicalByte^; INC(TopicalByte); DEC(PatternLength);
DEC(CodeByte);
CodeByte := CodeByte MOD 12 * 16 + CodeByte DIV 12 + 2;
Pattern^[NoteP,ChannelP,c_Pattern_NoteIndex] := CodeByte;
{ *** read sample }
CodeByte := TopicalByte^; INC(TopicalByte); DEC(PatternLength);
Pattern^[NoteP,ChannelP,c_Pattern_SampleIndex] := CodeByte;
END;
IF InfoByte AND 64 = 64 THEN { Volume }
BEGIN
CodeByte := TopicalByte^; INC(TopicalByte); DEC(PatternLength);
Pattern^[NoteP,ChannelP,c_Pattern_VolumeIndex] := CodeByte;
END;
IF InfoByte AND 32 = 32 THEN { effect AND opperand }
BEGIN
Effect := TopicalByte^; INC(TopicalByte); DEC(PatternLength);
Opperand := TopicalByte^; INC(TopicalByte); DEC(PatternLength);
CASE Effect OF
c_PSM_SetSpeed:
BEGIN
Effect := c_I_Set_Speed;
END;
ELSE
BEGIN
Effect := c_I_NoEffect;
Opperand := $00;
END;
END;
Pattern^[NoteP,ChannelP,c_Pattern_EffectIndex] := Effect;
Pattern^[NoteP,ChannelP,c_Pattern_OpperandIndex] := Opperand;
END;
END ELSE INC(NoteP);
END;
END;
PROCEDURE PSM_Load(FileName : STRING;FilePosition : LONGINT;VAR Module : PModule;VAR ErrorCode : WORD);
{ *** caution : Module has to be inited before!!!! }
VAR
Header : PPSM_Header;
Sample : PPSM_SampleList;
Order : PPSM_Order;
ChannelSettings : PPSM_ChannelSettings;
MultiPurposeBuffer : PByteArray;
PatternBuffer : PUnpackedPattern;
TopicalParaPointer : WORD;
InFile : FILE;
I1,I2 : WORD;
I3,I4 : WORD;
TempW : WORD;
TempB : BYTE;
TempP : PByteArray;
TempI : INTEGER;
{ *** copy-vars for loop-extension }
CopySource : LONGINT;
CopyDestination : LONGINT;
CopyLength : LONGINT;
BEGIN
{ *** try to open file }
ASSIGN(InFile,FileName);
{$I-}
RESET(InFile,1);
{$I+}
IF IORESULT <> $00 THEN
BEGIN
EXIT;
END;
{$I-}
{ *** seek start of module }
IF FILESIZE(InFile) < FilePosition THEN
BEGIN
EXIT;
END;
SEEK(InFile,FilePosition);
{ *** look for enough memory for temporary variables }
IF MEMAVAIL < SIZEOF(TPSM_Header) + SIZEOF(TPSM_SampleList) +
SIZEOF(TPSM_Order) + SIZEOF(TPSM_ChannelSettings) +
SIZEOF(TByteArray) + SIZEOF(TUnpackedPattern)
THEN
BEGIN
EXIT;
END;
{ *** init dynamic variables }
NEW(Header);
NEW(Sample);
NEW(Order);
NEW(ChannelSettings);
NEW(MultiPurposeBuffer);
NEW(PatternBuffer);
{ *** read header }
BLOCKREAD(InFile,Header^,SIZEOF(TPSM_Header));
{ *** test if this is a DSM-file }
IF NOT ((Header^.PSM_Sign[1] = 'P') AND (Header^.PSM_Sign[2] = 'S') AND
(Header^.PSM_Sign[3] = 'M') AND (Header^.PSM_Sign[4] = #254)) THEN
BEGIN
ErrorCode := c_NoValidFileFormat;
CLOSE(InFile);
EXIT;
END;
{ *** read order }
SEEK(InFile,FilePosition + Header^.PSM_OrderPosition);
BLOCKREAD(InFile,Order^,Header^.PSM_OrderLength);
{ *** read channelsettings }
SEEK(InFile,FilePosition + Header^.PSM_ChannelSettingPosition);
BLOCKREAD(InFile,ChannelSettings^,SIZEOF(TPSM_ChannelSettings));
{ *** read samplelist }
SEEK(InFile,FilePosition + Header^.PSM_SamplePosition);
BLOCKREAD(InFile,Sample^,Header^.PSM_SampleNumber * SIZEOF(TPSM_Sample));
{ *** copy header to intern NTMIK-structure }
Module^.Module_Sign := 'MF';
Module^.Module_FileFormatVersion := $0100;
Module^.Module_SampleNumber := Header^.PSM_SampleNumber;
Module^.Module_PatternNumber := Header^.PSM_PatternNumber;
Module^.Module_OrderLength := Header^.PSM_OrderLength;
Module^.Module_ChannelNumber := Header^.PSM_ChannelNumber+1;
Module^.Module_Initial_GlobalVolume := 64;
Module^.Module_Initial_MasterVolume := $C0;
Module^.Module_Initial_Speed := Header^.PSM_Speed;
Module^.Module_Initial_Tempo := Header^.PSM_Tempo;
{ *** paragraph 01 start }
Module^.Module_Flags := c_Module_Flags_ZeroVolume * BYTE(1) +
c_Module_Flags_Stereo * BYTE(1) +
c_Module_Flags_ForceAmigaLimits * BYTE(0) +
c_Module_Flags_Panning * BYTE(1) +
c_Module_Flags_Surround * BYTE(1) +
c_Module_Flags_QualityMixing * BYTE(1) +
c_Module_Flags_FastVolumeSlides * BYTE(0) +
c_Module_Flags_SpecialCustomData * BYTE(0) +
c_Module_Flags_SongName * BYTE(1);
I1 := $01;
WHILE (Header^.PSM_SongName[I1] > #00) AND (I1 < c_Module_SongNameLength) DO
BEGIN
Module^.Module_Name[I1] := Header^.PSM_SongName[I1];
INC(I1);
END;
Module^.Module_Name[c_Module_SongNameLength] := #00;
{ *** Init channelsettings }
FOR I1 := 0 TO c_Maximum_ChannelIndex DO
BEGIN
IF I1 < Header^.PSM_ChannelUsed THEN
BEGIN
{ *** channel enabled }
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_GlobalVolume := 64;
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Panning := (ChannelSettings^[I1]) * $08;
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Code := I1 + $10 * BYTE(ChannelSettings^[I1] > $08) +
c_ChannelSettings_Code_ChannelEnabled * BYTE(1) +
c_ChannelSettings_Code_ChannelDigital * BYTE(1);
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Controls :=
c_ChannelSettings_Controls_EnhancedMode * BYTE(1) +
c_ChannelSettings_Controls_SurroundMode * BYTE(0);
END
ELSE
BEGIN
{ *** channel disabled }
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_GlobalVolume := $00;
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Panning := $00;
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Code := $00;
Module^.Module_ChannelSettingPointer^[I1].ChannelSettings_Controls := $00;
END;
END;
{ *** init and copy order }
FILLCHAR(Module^.Module_OrderPointer^,c_Maximum_OrderIndex+1,$FF);
MOVE(Order^,Module^.Module_OrderPointer^,Header^.PSM_OrderLength);
{ *** read pattern }
SEEK(InFile,FilePosition + Header^.PSM_PatternPosition);
NTMIK_LoaderPatternNumber := Header^.PSM_PatternNumber-1;
FOR I1 := 0 TO Header^.PSM_PatternNumber-1 DO
BEGIN
NTMIK_LoadPatternProcedure;
{ *** read length }
BLOCKREAD(InFile,TempW,2);
{ *** read pattern }
BLOCKREAD(InFile,MultiPurposeBuffer^,TempW-2);
{ *** unpack pattern and set notes per channel to 64 }
PSM_UnpackPattern(MultiPurposeBuffer^,PatternBuffer^,TempW);
NTMIK_PackPattern(MultiPurposeBuffer^,PatternBuffer^,PSM_NotesInPattern);
TempW := WORD(256) * MultiPurposeBuffer^[01] + MultiPurposeBuffer^[00];
GETMEM(Module^.Module_PatternPointer^[I1],TempW);
MOVE(MultiPurposeBuffer^,Module^.Module_PatternPointer^[I1]^,TempW);
{ *** next pattern }
END;
{ *** read samples }
NTMIK_LoaderSampleNumber := Header^.PSM_SampleNumber;
FOR I1 := 1 TO Header^.PSM_SampleNumber DO
BEGIN
NTMIK_LoadSampleProcedure;
{ *** get index for sample }
I3 := Sample^[I1].PSM_SampleNumber;
{ *** clip PSM-sample }
IF Sample^[I1].PSM_SampleLoopEnd > Sample^[I1].PSM_SampleLength
THEN Sample^[I1].PSM_SampleLoopEnd := Sample^[I1].PSM_SampleLength;
{ *** init intern sample }
NEW(Module^.Module_SamplePointer^[I3]);
FILLCHAR(Module^.Module_SamplePointer^[I3]^,SIZEOF(TSample),$00);
FILLCHAR(Module^.Module_SamplePointer^[I3]^.Sample_SampleName,c_Sample_SampleNameLength,#32);
FILLCHAR(Module^.Module_SamplePointer^[I3]^.Sample_FileName,c_Sample_FileNameLength,#32);
{ *** copy informations to intern sample }
I2 := $01;
WHILE (Sample^[I1].PSM_SampleName[I2] > #00) AND (I2 < c_Sample_SampleNameLength) DO
BEGIN
Module^.Module_SamplePointer^[I3]^.Sample_SampleName[I2] := Sample^[I1].PSM_SampleName[I2];
INC(I2);
END;
Module^.Module_SamplePointer^[I3]^.Sample_Sign := 'DF';
Module^.Module_SamplePointer^[I3]^.Sample_FileFormatVersion := $00100;
Module^.Module_SamplePointer^[I3]^.Sample_Position := $00000000;
Module^.Module_SamplePointer^[I3]^.Sample_Selector := $0000;
Module^.Module_SamplePointer^[I3]^.Sample_Volume := Sample^[I1].PSM_SampleVolume;
Module^.Module_SamplePointer^[I3]^.Sample_LoopCounter := $00;
Module^.Module_SamplePointer^[I3]^.Sample_C5Speed := Sample^[I1].PSM_SampleC5Speed;
Module^.Module_SamplePointer^[I3]^.Sample_Length := Sample^[I1].PSM_SampleLength;
Module^.Module_SamplePointer^[I3]^.Sample_LoopBegin := Sample^[I1].PSM_SampleLoopBegin;
Module^.Module_SamplePointer^[I3]^.Sample_LoopEnd := Sample^[I1].PSM_SampleLoopEnd;
{ *** now it's time for the flags }
Module^.Module_SamplePointer^[I3]^.Sample_Flags :=
c_Sample_Flags_DigitalSample * BYTE(1) +
c_Sample_Flags_8BitSample * BYTE(1) +
c_Sample_Flags_UnsignedSampleData * BYTE(1) +
c_Sample_Flags_Packed * BYTE(0) +
c_Sample_Flags_LoopCounter * BYTE(0) +
c_Sample_Flags_SampleName * BYTE(1) +
c_Sample_Flags_LoopActive *
BYTE(Sample^[I1].PSM_SampleFlags AND (LONGINT(1) SHL 15) = (LONGINT(1) SHL 15));
{ *** alloc memory for sample-data }
E_Getmem(Module^.Module_SamplePointer^[I3]^.Sample_Selector,
Module^.Module_SamplePointer^[I3]^.Sample_Position,
Module^.Module_SamplePointer^[I3]^.Sample_Length + c_LoopExtensionSize);
{ *** read out data }
EPT(TempP).p_Selector := Module^.Module_SamplePointer^[I3]^.Sample_Selector;
EPT(TempP).p_Offset := $0000;
SEEK(InFile,Sample^[I1].PSM_SamplePosition);
E_BLOCKREAD(InFile,TempP^,Module^.Module_SamplePointer^[I3]^.Sample_Length);
{ *** 'coz the samples are signed in a DSM-file -> PC-fy them }
IF Module^.Module_SamplePointer^[I3]^.Sample_Length > 4 THEN
BEGIN
CopyLength := Module^.Module_SamplePointer^[I3]^.Sample_Length;
{ *** decode sample }
ASM
DB 066h; MOV CX,WORD PTR CopyLength
{ *** load sample selector }
MOV ES,WORD PTR TempP[00002h]
DB 066h; XOR SI,SI
DB 066h; XOR DI,DI
XOR AH,AH
{ *** conert all bytes }
@@MainLoop:
DB 026h; DB 067h; LODSB
ADD AL,AH
MOV AH,AL
DB 067h; STOSB
DB 066h; LOOP @@MainLoop
END;
{ *** make samples unsigned }
ASM
DB 066h; MOV CX,WORD PTR CopyLength
{ *** load sample selector }
MOV ES,WORD PTR TempP[00002h]
DB 066h; XOR SI,SI
DB 066h; XOR DI,DI
{ *** conert all bytes }
@@MainLoop:
DB 026h; DB 067h; LODSB
SUB AL,080h
DB 067h; STOSB
DB 066h; LOOP @@MainLoop
END;
{ *** Create Loop-Extension }
IF Module^.Module_SamplePointer^[I3]^.Sample_Flags AND c_Sample_Flags_LoopActive = c_Sample_Flags_LoopActive THEN
BEGIN
CopySource := Module^.Module_SamplePointer^[I3]^.Sample_LoopBegin;
CopyDestination := Module^.Module_SamplePointer^[I3]^.Sample_LoopEnd;
CopyLength := CopyDestination - CopySource;
ASM
{ *** load sample-selector }
MOV ES,WORD PTR TempP[00002h]
DB 066h; MOV DI,WORD PTR CopyDestination
{ *** calculate number of full sample-loops to copy }
XOR DX,DX
MOV AX,c_LoopExtensionSize
MOV BX,WORD PTR CopyLength
DIV BX
OR AX,AX
JE @@NoFullLoop
{ *** copy some full-loops (size=bx) }
MOV CX,AX
@@InnerLoop:
PUSH CX
DB 066h; MOV SI,WORD PTR CopySource
MOV CX,BX
DB 0F3h; DB 026h,067h,0A4h { REP MOVS BYTE PTR ES:[EDI],ES:[ESI] }
POP CX
LOOP @@InnerLoop
@@NoFullLoop:
{ *** calculate number of rest-bytes to copy }
DB 066h; MOV SI,WORD PTR CopySource
MOV CX,DX
DB 0F3h; DB 026h,067h,0A4h { REP MOVS BYTE PTR ES:[EDI],ES:[ESI] }
END;
END
ELSE
BEGIN
CopyDestination := Module^.Module_SamplePointer^[I3]^.Sample_Length;
ASM
{ *** load sample-selector }
MOV ES,WORD PTR TempP[00002h]
DB 066h; MOV DI,WORD PTR CopyDestination
{ *** clear extension }
MOV CX,c_LoopExtensionSize
MOV AL,080h
DB 0F3h; DB 067h,0AAh { REP STOS BYTE PTR ES:[EDI] }
END;
END;
END;
{ *** next sample }
END;
{ *** init period-ranges }
NTMIK_MaximumPeriod := $0000D600 SHR 1;
NTMIK_MinimumPeriod := $0000D600 SHR 8;
{ *** close file }
CLOSE(InFile);
{ *** dispose all dynamic variables }
DISPOSE(Header);
DISPOSE(Sample);
DISPOSE(Order);
DISPOSE(ChannelSettings);
DISPOSE(MultiPurposeBuffer);
DISPOSE(PatternBuffer);
{ *** set errorcode to noerror }
ErrorCode := c_NoError;
END;
*/

@ -0,0 +1,212 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
//////////////////////////////////////////////
// PTM PolyTracker module loader //
//////////////////////////////////////////////
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#pragma pack(1)
typedef struct PTMFILEHEADER
{
CHAR songname[28]; // name of song, asciiz string
CHAR eof; // 26
BYTE version_lo; // 03 version of file, currently 0203h
BYTE version_hi; // 02
BYTE reserved1; // reserved, set to 0
WORD norders; // number of orders (0..256)
WORD nsamples; // number of instruments (1..255)
WORD npatterns; // number of patterns (1..128)
WORD nchannels; // number of channels (voices) used (1..32)
WORD fileflags; // set to 0
WORD reserved2; // reserved, set to 0
DWORD ptmf_id; // song identification, 'PTMF' or 0x464d5450
BYTE reserved3[16]; // reserved, set to 0
BYTE chnpan[32]; // channel panning settings, 0..15, 0 = left, 7 = middle, 15 = right
BYTE orders[256]; // order list, valid entries 0..nOrders-1
WORD patseg[128]; // pattern offsets (*16)
} PTMFILEHEADER, *LPPTMFILEHEADER;
#define SIZEOF_PTMFILEHEADER 608
typedef struct PTMSAMPLE
{
BYTE sampletype; // sample type (bit array)
CHAR filename[12]; // name of external sample file
BYTE volume; // default volume
WORD nC4Spd; // C4 speed
WORD sampleseg; // sample segment (used internally)
WORD fileofs[2]; // offset of sample data
WORD length[2]; // sample size (in bytes)
WORD loopbeg[2]; // start of loop
WORD loopend[2]; // end of loop
WORD gusdata[8];
char samplename[28]; // name of sample, asciiz // changed from CHAR
DWORD ptms_id; // sample identification, 'PTMS' or 0x534d5450
} PTMSAMPLE;
#define SIZEOF_PTMSAMPLE 80
#pragma pack()
static uint32_t BS2WORD(uint16_t w[2]) {
uint32_t u32 = (w[1] << 16) + w[0];
return(bswapLE32(u32));
}
BOOL CSoundFile::ReadPTM(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
DWORD dwMemPos;
UINT nOrders;
if ((!lpStream) || (dwMemLength < sizeof(PTMFILEHEADER))) return FALSE;
PTMFILEHEADER pfh = *(LPPTMFILEHEADER)lpStream;
pfh.norders = bswapLE16(pfh.norders);
pfh.nsamples = bswapLE16(pfh.nsamples);
pfh.npatterns = bswapLE16(pfh.npatterns);
pfh.nchannels = bswapLE16(pfh.nchannels);
pfh.fileflags = bswapLE16(pfh.fileflags);
pfh.reserved2 = bswapLE16(pfh.reserved2);
pfh.ptmf_id = bswapLE32(pfh.ptmf_id);
for (UINT j=0; j<128; j++)
{
pfh.patseg[j] = bswapLE16(pfh.patseg[j]);
}
if ((pfh.ptmf_id != 0x464d5450) || (!pfh.nchannels)
|| (pfh.nchannels > 32)
|| (pfh.norders > 256) || (!pfh.norders)
|| (!pfh.nsamples) || (pfh.nsamples > 255)
|| (!pfh.npatterns) || (pfh.npatterns > 128)
|| (SIZEOF_PTMFILEHEADER+pfh.nsamples*SIZEOF_PTMSAMPLE >= (int)dwMemLength)) return FALSE;
memcpy(m_szNames[0], pfh.songname, 28);
m_szNames[0][28] = 0;
m_nType = MOD_TYPE_PTM;
m_nChannels = pfh.nchannels;
m_nSamples = (pfh.nsamples < MAX_SAMPLES) ? pfh.nsamples : MAX_SAMPLES-1;
dwMemPos = SIZEOF_PTMFILEHEADER;
nOrders = (pfh.norders < MAX_ORDERS) ? pfh.norders : MAX_ORDERS-1;
memcpy(Order, pfh.orders, nOrders);
for (UINT ipan=0; ipan<m_nChannels; ipan++)
{
ChnSettings[ipan].nVolume = 64;
ChnSettings[ipan].nPan = ((pfh.chnpan[ipan] & 0x0F) << 4) + 4;
}
for (UINT ismp=0; ismp<m_nSamples; ismp++, dwMemPos += SIZEOF_PTMSAMPLE)
{
MODINSTRUMENT *pins = &Ins[ismp+1];
PTMSAMPLE *psmp = (PTMSAMPLE *)(lpStream+dwMemPos);
lstrcpyn(m_szNames[ismp+1], psmp->samplename, 28);
memcpy(pins->name, psmp->filename, 12);
pins->name[12] = 0;
pins->nGlobalVol = 64;
pins->nPan = 128;
pins->nVolume = psmp->volume << 2;
pins->nC4Speed = bswapLE16(psmp->nC4Spd) << 1;
pins->uFlags = 0;
if ((psmp->sampletype & 3) == 1)
{
UINT smpflg = RS_PCM8D;
pins->nLength = BS2WORD(psmp->length);
pins->nLoopStart = BS2WORD(psmp->loopbeg);
pins->nLoopEnd = BS2WORD(psmp->loopend);
DWORD samplepos = BS2WORD(psmp->fileofs);
if (psmp->sampletype & 4) pins->uFlags |= CHN_LOOP;
if (psmp->sampletype & 8) pins->uFlags |= CHN_PINGPONGLOOP;
if (psmp->sampletype & 16)
{
pins->uFlags |= CHN_16BIT;
pins->nLength >>= 1;
pins->nLoopStart >>= 1;
pins->nLoopEnd >>= 1;
smpflg = RS_PTM8DTO16;
}
if ((pins->nLength) && (samplepos) && (samplepos < dwMemLength))
{
ReadSample(pins, smpflg, (LPSTR)(lpStream+samplepos), dwMemLength-samplepos);
}
}
}
// Reading Patterns
for (UINT ipat=0; ipat<pfh.npatterns; ipat++)
{
dwMemPos = ((UINT)pfh.patseg[ipat]) << 4;
if ((!dwMemPos) || (dwMemPos >= dwMemLength)) continue;
PatternSize[ipat] = 64;
if ((Patterns[ipat] = AllocatePattern(64, m_nChannels)) == NULL) break;
//
MODCOMMAND *m = Patterns[ipat];
for (UINT row=0; ((row < 64) && (dwMemPos < dwMemLength)); )
{
UINT b = lpStream[dwMemPos++];
if (dwMemPos >= dwMemLength) break;
if (b)
{
UINT nChn = b & 0x1F;
if (b & 0x20)
{
if (dwMemPos + 2 > dwMemLength) break;
m[nChn].note = lpStream[dwMemPos++];
m[nChn].instr = lpStream[dwMemPos++];
}
if (b & 0x40)
{
if (dwMemPos + 2 > dwMemLength) break;
m[nChn].command = lpStream[dwMemPos++];
m[nChn].param = lpStream[dwMemPos++];
if ((m[nChn].command == 0x0E) && ((m[nChn].param & 0xF0) == 0x80))
{
m[nChn].command = CMD_S3MCMDEX;
} else
if (m[nChn].command < 0x10)
{
ConvertModCommand(&m[nChn]);
} else
{
switch(m[nChn].command)
{
case 16:
m[nChn].command = CMD_GLOBALVOLUME;
break;
case 17:
m[nChn].command = CMD_RETRIG;
break;
case 18:
m[nChn].command = CMD_FINEVIBRATO;
break;
default:
m[nChn].command = 0;
}
}
}
if (b & 0x80)
{
if (dwMemPos >= dwMemLength) break;
m[nChn].volcmd = VOLCMD_VOLUME;
m[nChn].vol = lpStream[dwMemPos++];
}
} else
{
row++;
m += m_nChannels;
}
}
}
return TRUE;
}

@ -0,0 +1,671 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
#include "stdafx.h"
#include "sndfile.h"
#include "tables.h"
#ifdef _MSC_VER
//#pragma warning(disable:4244)
#endif
//////////////////////////////////////////////////////
// ScreamTracker S3M file support
#pragma pack(1)
typedef struct tagS3MSAMPLESTRUCT
{
BYTE type;
CHAR dosname[12];
BYTE hmem;
WORD memseg;
DWORD length;
DWORD loopbegin;
DWORD loopend;
BYTE vol;
BYTE bReserved;
BYTE pack;
BYTE flags;
DWORD finetune;
DWORD dwReserved;
WORD intgp;
WORD int512;
DWORD lastused;
CHAR name[28];
CHAR scrs[4];
} S3MSAMPLESTRUCT;
typedef struct tagS3MFILEHEADER
{
CHAR name[28];
BYTE b1A;
BYTE type;
WORD reserved1;
WORD ordnum;
WORD insnum;
WORD patnum;
WORD flags;
WORD cwtv;
WORD version;
DWORD scrm; // "SCRM" = 0x4D524353
BYTE globalvol;
BYTE speed;
BYTE tempo;
BYTE mastervol;
BYTE ultraclicks;
BYTE panning_present;
BYTE reserved2[8];
WORD special;
BYTE channels[32];
} S3MFILEHEADER;
void CSoundFile::S3MConvert(MODCOMMAND *m, BOOL bIT) const
//--------------------------------------------------------
{
UINT command = m->command;
UINT param = m->param;
switch (command + 0x40)
{
case 'A': command = CMD_SPEED; break;
case 'B': command = CMD_POSITIONJUMP; break;
case 'C': command = CMD_PATTERNBREAK; if (!bIT) param = (param >> 4) * 10 + (param & 0x0F); break;
case 'D': command = CMD_VOLUMESLIDE; break;
case 'E': command = CMD_PORTAMENTODOWN; break;
case 'F': command = CMD_PORTAMENTOUP; break;
case 'G': command = CMD_TONEPORTAMENTO; break;
case 'H': command = CMD_VIBRATO; break;
case 'I': command = CMD_TREMOR; break;
case 'J': command = CMD_ARPEGGIO; break;
case 'K': command = CMD_VIBRATOVOL; break;
case 'L': command = CMD_TONEPORTAVOL; break;
case 'M': command = CMD_CHANNELVOLUME; break;
case 'N': command = CMD_CHANNELVOLSLIDE; break;
case 'O': command = CMD_OFFSET; break;
case 'P': command = CMD_PANNINGSLIDE; break;
case 'Q': command = CMD_RETRIG; break;
case 'R': command = CMD_TREMOLO; break;
case 'S': command = CMD_S3MCMDEX; break;
case 'T': command = CMD_TEMPO; break;
case 'U': command = CMD_FINEVIBRATO; break;
case 'V': command = CMD_GLOBALVOLUME; break;
case 'W': command = CMD_GLOBALVOLSLIDE; break;
case 'X': command = CMD_PANNING8; break;
case 'Y': command = CMD_PANBRELLO; break;
case 'Z': command = CMD_MIDI; break;
default: command = 0;
}
m->command = command;
m->param = param;
}
void CSoundFile::S3MSaveConvert(UINT *pcmd, UINT *pprm, BOOL bIT) const
//---------------------------------------------------------------------
{
UINT command = *pcmd;
UINT param = *pprm;
switch(command)
{
case CMD_SPEED: command = 'A'; break;
case CMD_POSITIONJUMP: command = 'B'; break;
case CMD_PATTERNBREAK: command = 'C'; if (!bIT) param = ((param / 10) << 4) + (param % 10); break;
case CMD_VOLUMESLIDE: command = 'D'; break;
case CMD_PORTAMENTODOWN: command = 'E'; if ((param >= 0xE0) && (m_nType & (MOD_TYPE_MOD|MOD_TYPE_XM))) param = 0xDF; break;
case CMD_PORTAMENTOUP: command = 'F'; if ((param >= 0xE0) && (m_nType & (MOD_TYPE_MOD|MOD_TYPE_XM))) param = 0xDF; break;
case CMD_TONEPORTAMENTO: command = 'G'; break;
case CMD_VIBRATO: command = 'H'; break;
case CMD_TREMOR: command = 'I'; break;
case CMD_ARPEGGIO: command = 'J'; break;
case CMD_VIBRATOVOL: command = 'K'; break;
case CMD_TONEPORTAVOL: command = 'L'; break;
case CMD_CHANNELVOLUME: command = 'M'; break;
case CMD_CHANNELVOLSLIDE: command = 'N'; break;
case CMD_OFFSET: command = 'O'; break;
case CMD_PANNINGSLIDE: command = 'P'; break;
case CMD_RETRIG: command = 'Q'; break;
case CMD_TREMOLO: command = 'R'; break;
case CMD_S3MCMDEX: command = 'S'; break;
case CMD_TEMPO: command = 'T'; break;
case CMD_FINEVIBRATO: command = 'U'; break;
case CMD_GLOBALVOLUME: command = 'V'; break;
case CMD_GLOBALVOLSLIDE: command = 'W'; break;
case CMD_PANNING8:
command = 'X';
if ((bIT) && (m_nType != MOD_TYPE_IT) && (m_nType != MOD_TYPE_XM))
{
if (param == 0xA4) { command = 'S'; param = 0x91; } else
if (param <= 0x80) { param <<= 1; if (param > 255) param = 255; } else
command = param = 0;
} else
if ((!bIT) && ((m_nType == MOD_TYPE_IT) || (m_nType == MOD_TYPE_XM)))
{
param >>= 1;
}
break;
case CMD_PANBRELLO: command = 'Y'; break;
case CMD_MIDI: command = 'Z'; break;
case CMD_XFINEPORTAUPDOWN:
if (param & 0x0F) switch(param & 0xF0)
{
case 0x10: command = 'F'; param = (param & 0x0F) | 0xE0; break;
case 0x20: command = 'E'; param = (param & 0x0F) | 0xE0; break;
case 0x90: command = 'S'; break;
default: command = param = 0;
} else command = param = 0;
break;
case CMD_MODCMDEX:
command = 'S';
switch(param & 0xF0)
{
case 0x00: command = param = 0; break;
case 0x10: command = 'F'; param |= 0xF0; break;
case 0x20: command = 'E'; param |= 0xF0; break;
case 0x30: param = (param & 0x0F) | 0x10; break;
case 0x40: param = (param & 0x0F) | 0x30; break;
case 0x50: param = (param & 0x0F) | 0x20; break;
case 0x60: param = (param & 0x0F) | 0xB0; break;
case 0x70: param = (param & 0x0F) | 0x40; break;
case 0x90: command = 'Q'; param &= 0x0F; break;
case 0xA0: if (param & 0x0F) { command = 'D'; param = (param << 4) | 0x0F; } else command=param=0; break;
case 0xB0: if (param & 0x0F) { command = 'D'; param |= 0xF0; } else command=param=0; break;
}
break;
default: command = param = 0;
}
command &= ~0x40;
*pcmd = command;
*pprm = param;
}
static DWORD boundInput(DWORD input, DWORD smin, DWORD smax)
{
if (input > smax) input = smax;
else if (input < smin) input = 0;
return(input);
}
BOOL CSoundFile::ReadS3M(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
UINT insnum,patnum,nins,npat;
DWORD insfile[MAX_SAMPLES];
WORD ptr[256];
DWORD dwMemPos;
BYTE insflags[MAX_SAMPLES], inspack[MAX_SAMPLES];
if ((!lpStream) || (dwMemLength <= sizeof(S3MFILEHEADER)+sizeof(S3MSAMPLESTRUCT)+64)) return FALSE;
S3MFILEHEADER psfh = *(S3MFILEHEADER *)lpStream;
psfh.reserved1 = bswapLE16(psfh.reserved1);
psfh.ordnum = bswapLE16(psfh.ordnum);
psfh.insnum = bswapLE16(psfh.insnum);
psfh.patnum = bswapLE16(psfh.patnum);
psfh.flags = bswapLE16(psfh.flags);
psfh.cwtv = bswapLE16(psfh.cwtv);
psfh.version = bswapLE16(psfh.version);
psfh.scrm = bswapLE32(psfh.scrm);
psfh.special = bswapLE16(psfh.special);
if (psfh.scrm != 0x4D524353) return FALSE;
dwMemPos = 0x60;
m_nType = MOD_TYPE_S3M;
memset(m_szNames,0,sizeof(m_szNames));
memcpy(m_szNames[0], psfh.name, 28);
// Speed
m_nDefaultSpeed = psfh.speed;
if (m_nDefaultSpeed < 1) m_nDefaultSpeed = 6;
if (m_nDefaultSpeed > 0x1F) m_nDefaultSpeed = 0x1F;
// Tempo
m_nDefaultTempo = psfh.tempo;
if (m_nDefaultTempo < 40) m_nDefaultTempo = 40;
if (m_nDefaultTempo > 240) m_nDefaultTempo = 240;
// Global Volume
m_nDefaultGlobalVolume = psfh.globalvol << 2;
if ((!m_nDefaultGlobalVolume) || (m_nDefaultGlobalVolume > 256)) m_nDefaultGlobalVolume = 256;
m_nSongPreAmp = psfh.mastervol & 0x7F;
// Channels
m_nChannels = 4;
for (UINT ich=0; ich<32; ich++)
{
ChnSettings[ich].nPan = 128;
ChnSettings[ich].nVolume = 64;
ChnSettings[ich].dwFlags = CHN_MUTE;
if (psfh.channels[ich] != 0xFF)
{
m_nChannels = ich+1;
UINT b = psfh.channels[ich] & 0x0F;
ChnSettings[ich].nPan = (b & 8) ? 0xC0 : 0x40;
ChnSettings[ich].dwFlags = 0;
}
}
if (m_nChannels < 4) m_nChannels = 4;
if ((psfh.cwtv < 0x1320) || (psfh.flags & 0x40)) m_dwSongFlags |= SONG_FASTVOLSLIDES;
// Reading pattern order
UINT iord = psfh.ordnum;
if (iord<1) iord = 1;
if (iord > MAX_ORDERS) iord = MAX_ORDERS;
if (iord)
{
memcpy(Order, lpStream+dwMemPos, iord);
dwMemPos += iord;
}
if ((iord & 1) && (lpStream[dwMemPos] == 0xFF)) dwMemPos++;
// Reading file pointers
insnum = nins = psfh.insnum;
if (insnum >= MAX_SAMPLES) insnum = MAX_SAMPLES-1;
m_nSamples = insnum;
patnum = npat = psfh.patnum;
if (patnum > MAX_PATTERNS) patnum = MAX_PATTERNS;
memset(ptr, 0, sizeof(ptr));
// Ignore file if it has a corrupted header.
if (nins+npat > 256) return FALSE;
if (nins+npat)
{
memcpy(ptr, lpStream+dwMemPos, 2*(nins+npat));
dwMemPos += 2*(nins+npat);
for (UINT j = 0; j < (nins+npat); ++j) {
ptr[j] = bswapLE16(ptr[j]);
}
if (psfh.panning_present == 252)
{
const BYTE *chnpan = lpStream+dwMemPos;
for (UINT i=0; i<32; i++) if (chnpan[i] & 0x20)
{
ChnSettings[i].nPan = ((chnpan[i] & 0x0F) << 4) + 8;
}
}
}
if (!m_nChannels) return TRUE;
// Reading instrument headers
memset(insfile, 0, sizeof(insfile));
for (UINT iSmp=1; iSmp<=insnum; iSmp++)
{
UINT nInd = ((DWORD)ptr[iSmp-1])*16;
if ((!nInd) || (nInd + 0x50 > dwMemLength)) {
// initialize basic variables.
insflags[iSmp-1] = 0;
inspack[iSmp-1] = 0;
continue;
}
S3MSAMPLESTRUCT pSmp;
memcpy(&pSmp, lpStream+nInd, 0x50);
memcpy(Ins[iSmp].name, &pSmp.dosname, 12);
insflags[iSmp-1] = pSmp.flags;
inspack[iSmp-1] = pSmp.pack;
memcpy(m_szNames[iSmp], pSmp.name, 28);
m_szNames[iSmp][28] = 0;
if ((pSmp.type==1) && (pSmp.scrs[2]=='R') && (pSmp.scrs[3]=='S'))
{
Ins[iSmp].nLength = boundInput(bswapLE32(pSmp.length), 4, MAX_SAMPLE_LENGTH);
Ins[iSmp].nLoopStart = boundInput(bswapLE32(pSmp.loopbegin), 4, Ins[iSmp].nLength - 1);
Ins[iSmp].nLoopEnd = boundInput(bswapLE32(pSmp.loopend), 4, Ins[iSmp].nLength);
Ins[iSmp].nVolume = boundInput(pSmp.vol, 0, 64) << 2;
Ins[iSmp].nGlobalVol = 64;
if (pSmp.flags&1) Ins[iSmp].uFlags |= CHN_LOOP;
UINT j = bswapLE32(pSmp.finetune);
if (!j) j = 8363;
if (j < 1024) j = 1024;
Ins[iSmp].nC4Speed = j;
insfile[iSmp] = (pSmp.hmem << 20) + (bswapLE16(pSmp.memseg) << 4);
// offset is invalid - ignore this sample.
if (insfile[iSmp] > dwMemLength) insfile[iSmp] = 0;
else if (insfile[iSmp]) {
// ignore duplicate samples.
for (int z=iSmp-1; z>=0; z--)
if (insfile[iSmp] == insfile[z])
insfile[iSmp] = 0;
}
if ((Ins[iSmp].nLoopStart >= Ins[iSmp].nLoopEnd) || (Ins[iSmp].nLoopEnd - Ins[iSmp].nLoopStart < 8))
Ins[iSmp].nLoopStart = Ins[iSmp].nLoopEnd = 0;
Ins[iSmp].nPan = 0x80;
}
}
// Reading patterns
for (UINT iPat=0; iPat<patnum; iPat++)
{
UINT nInd = ((DWORD)ptr[nins+iPat]) << 4;
if (nInd + 0x40 > dwMemLength) continue;
WORD len = bswapLE16(*((WORD *)(lpStream+nInd)));
nInd += 2;
PatternSize[iPat] = 64;
if ((!len) || (nInd + len > dwMemLength - 6)
|| ((Patterns[iPat] = AllocatePattern(64, m_nChannels)) == NULL)) continue;
LPBYTE src = (LPBYTE)(lpStream+nInd);
// Unpacking pattern
MODCOMMAND *p = Patterns[iPat];
UINT row = 0;
UINT j = 0;
while (j < len)
{
BYTE b = src[j++];
if (!b)
{
if (++row >= 64) break;
} else
{
UINT chn = b & 0x1F;
if (chn < m_nChannels)
{
MODCOMMAND *m = &p[row*m_nChannels+chn];
if (b & 0x20)
{
m->note = src[j++];
if (m->note < 0xF0) m->note = (m->note & 0x0F) + 12*(m->note >> 4) + 13;
else if (m->note == 0xFF) m->note = 0;
m->instr = src[j++];
}
if (b & 0x40)
{
UINT vol = src[j++];
if ((vol >= 128) && (vol <= 192))
{
vol -= 128;
m->volcmd = VOLCMD_PANNING;
} else
{
if (vol > 64) vol = 64;
m->volcmd = VOLCMD_VOLUME;
}
m->vol = vol;
}
if (b & 0x80)
{
m->command = src[j++];
m->param = src[j++];
if (m->command) S3MConvert(m, FALSE);
}
} else
{
if (b & 0x20) j += 2;
if (b & 0x40) j++;
if (b & 0x80) j += 2;
}
if (j >= len) break;
}
}
}
// Reading samples
for (UINT iRaw=1; iRaw<=insnum; iRaw++) if ((Ins[iRaw].nLength) && (insfile[iRaw]))
{
UINT flags = (psfh.version == 1) ? RS_PCM8S : RS_PCM8U;
if (insflags[iRaw-1] & 4) flags += 5;
if (insflags[iRaw-1] & 2) flags |= RSF_STEREO;
if (inspack[iRaw-1] == 4) flags = RS_ADPCM4;
dwMemPos = insfile[iRaw];
if (dwMemPos < dwMemLength)
dwMemPos += ReadSample(&Ins[iRaw], flags, (LPSTR)(lpStream + dwMemPos), dwMemLength - dwMemPos);
}
m_nMinPeriod = 64;
m_nMaxPeriod = 32767;
if (psfh.flags & 0x10) m_dwSongFlags |= SONG_AMIGALIMITS;
return TRUE;
}
#ifndef MODPLUG_NO_FILESAVE
#ifdef _MSC_VER
#pragma warning(disable:4100)
#endif
static BYTE S3MFiller[16] =
{
0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80,
0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80
};
BOOL CSoundFile::SaveS3M(LPCSTR lpszFileName, UINT nPacking)
//----------------------------------------------------------
{
FILE *f;
BYTE header[0x60];
UINT nbo,nbi,nbp,i;
WORD patptr[128];
WORD insptr[128];
BYTE buffer[5*1024];
S3MSAMPLESTRUCT insex[128];
if ((!m_nChannels) || (!lpszFileName)) return FALSE;
if ((f = fopen(lpszFileName, "wb")) == NULL) return FALSE;
// Writing S3M header
memset(header, 0, sizeof(header));
memset(insex, 0, sizeof(insex));
memcpy(header, m_szNames[0], 0x1C);
header[0x1B] = 0;
header[0x1C] = 0x1A;
header[0x1D] = 0x10;
nbo = (GetNumPatterns() + 15) & 0xF0;
if (!nbo) nbo = 16;
header[0x20] = nbo & 0xFF;
header[0x21] = nbo >> 8;
nbi = m_nInstruments;
if (!nbi) nbi = m_nSamples;
if (nbi > 99) nbi = 99;
header[0x22] = nbi & 0xFF;
header[0x23] = nbi >> 8;
nbp = 0;
for (i=0; Patterns[i]; i++) { nbp = i+1; if (nbp >= MAX_PATTERNS) break; }
for (i=0; i<MAX_ORDERS; i++) if ((Order[i] < MAX_PATTERNS) && (Order[i] >= nbp)) nbp = Order[i] + 1;
header[0x24] = nbp & 0xFF;
header[0x25] = nbp >> 8;
if (m_dwSongFlags & SONG_FASTVOLSLIDES) header[0x26] |= 0x40;
if ((m_nMaxPeriod < 20000) || (m_dwSongFlags & SONG_AMIGALIMITS)) header[0x26] |= 0x10;
header[0x28] = 0x20;
header[0x29] = 0x13;
header[0x2A] = 0x02; // Version = 1 => Signed samples
header[0x2B] = 0x00;
header[0x2C] = 'S';
header[0x2D] = 'C';
header[0x2E] = 'R';
header[0x2F] = 'M';
header[0x30] = m_nDefaultGlobalVolume >> 2;
header[0x31] = m_nDefaultSpeed;
header[0x32] = m_nDefaultTempo;
header[0x33] = ((m_nSongPreAmp < 0x20) ? 0x20 : m_nSongPreAmp) | 0x80; // Stereo
header[0x35] = 0xFC;
for (i=0; i<32; i++)
{
if (i < m_nChannels)
{
UINT tmp = (i & 0x0F) >> 1;
header[0x40+i] = (i & 0x10) | ((i & 1) ? 8+tmp : tmp);
} else header[0x40+i] = 0xFF;
}
fwrite(header, 0x60, 1, f);
fwrite(Order, nbo, 1, f);
memset(patptr, 0, sizeof(patptr));
memset(insptr, 0, sizeof(insptr));
UINT ofs0 = 0x60 + nbo;
UINT ofs1 = ((0x60 + nbo + nbi*2 + nbp*2 + 15) & 0xFFF0) + 0x20;
UINT ofs = ofs1;
for (i=0; i<nbi; i++) insptr[i] = (WORD)((ofs + i*0x50) / 16);
for (i=0; i<nbp; i++) patptr[i] = (WORD)((ofs + nbi*0x50) / 16);
fwrite(insptr, nbi, 2, f);
fwrite(patptr, nbp, 2, f);
if (header[0x35] == 0xFC)
{
BYTE chnpan[32];
for (i=0; i<32; i++)
{
chnpan[i] = 0x20 | (ChnSettings[i].nPan >> 4);
}
fwrite(chnpan, 0x20, 1, f);
}
if ((nbi*2+nbp*2) & 0x0F)
{
fwrite(S3MFiller, 0x10 - ((nbi*2+nbp*2) & 0x0F), 1, f);
}
ofs1 = ftell(f);
fwrite(insex, nbi, 0x50, f);
// Packing patterns
ofs += nbi*0x50;
for (i=0; i<nbp; i++)
{
WORD len = 64;
memset(buffer, 0, sizeof(buffer));
patptr[i] = ofs / 16;
if (Patterns[i])
{
len = 2;
MODCOMMAND *p = Patterns[i];
for (int row=0; row<64; row++) if (row < PatternSize[i])
{
for (UINT j=0; j<m_nChannels; j++)
{
UINT b = j;
MODCOMMAND *m = &p[row*m_nChannels+j];
UINT note = m->note;
UINT volcmd = m->volcmd;
UINT vol = m->vol;
UINT command = m->command;
UINT param = m->param;
if ((note) || (m->instr)) b |= 0x20;
if (!note) note = 0xFF; else
if (note >= 0xFE) note = 0xFE; else
if (note < 13) note = 0; else note -= 13;
if (note < 0xFE) note = (note % 12) + ((note / 12) << 4);
if (command == CMD_VOLUME)
{
command = 0;
if (param > 64) param = 64;
volcmd = VOLCMD_VOLUME;
vol = param;
}
if (volcmd == VOLCMD_VOLUME) b |= 0x40; else
if (volcmd == VOLCMD_PANNING) { vol |= 0x80; b |= 0x40; }
if (command)
{
S3MSaveConvert(&command, &param, FALSE);
if (command) b |= 0x80;
}
if (b & 0xE0)
{
buffer[len++] = b;
if (b & 0x20)
{
buffer[len++] = note;
buffer[len++] = m->instr;
}
if (b & 0x40)
{
buffer[len++] = vol;
}
if (b & 0x80)
{
buffer[len++] = command;
buffer[len++] = param;
}
if (len > sizeof(buffer) - 20) break;
}
}
buffer[len++] = 0;
if (len > sizeof(buffer) - 20) break;
}
}
buffer[0] = (len - 2) & 0xFF;
buffer[1] = (len - 2) >> 8;
len = (len+15) & (~0x0F);
fwrite(buffer, len, 1, f);
ofs += len;
}
// Writing samples
for (i=1; i<=nbi; i++)
{
MODINSTRUMENT *pins = &Ins[i];
if (m_nInstruments)
{
pins = Ins;
if (Headers[i])
{
for (UINT j=0; j<128; j++)
{
UINT n = Headers[i]->Keyboard[j];
if ((n) && (n < MAX_INSTRUMENTS))
{
pins = &Ins[n];
break;
}
}
}
}
memcpy(insex[i-1].dosname, pins->name, 12);
memcpy(insex[i-1].name, m_szNames[i], 28);
memcpy(insex[i-1].scrs, "SCRS", 4);
insex[i-1].hmem = (BYTE)((DWORD)ofs >> 20);
insex[i-1].memseg = (WORD)((DWORD)ofs >> 4);
if (pins->pSample)
{
insex[i-1].type = 1;
insex[i-1].length = pins->nLength;
insex[i-1].loopbegin = pins->nLoopStart;
insex[i-1].loopend = pins->nLoopEnd;
insex[i-1].vol = pins->nVolume / 4;
insex[i-1].flags = (pins->uFlags & CHN_LOOP) ? 1 : 0;
if (pins->nC4Speed)
insex[i-1].finetune = pins->nC4Speed;
else
insex[i-1].finetune = TransposeToFrequency(pins->RelativeTone, pins->nFineTune);
UINT flags = RS_PCM8U;
#ifndef NO_PACKING
if (nPacking)
{
if ((!(pins->uFlags & (CHN_16BIT|CHN_STEREO)))
&& (CanPackSample((char *)pins->pSample, pins->nLength, nPacking)))
{
insex[i-1].pack = 4;
flags = RS_ADPCM4;
}
} else
#endif // NO_PACKING
{
if (pins->uFlags & CHN_16BIT)
{
insex[i-1].flags |= 4;
flags = RS_PCM16U;
}
if (pins->uFlags & CHN_STEREO)
{
insex[i-1].flags |= 2;
flags = (pins->uFlags & CHN_16BIT) ? RS_STPCM16U : RS_STPCM8U;
}
}
DWORD len = WriteSample(f, pins, flags);
if (len & 0x0F)
{
fwrite(S3MFiller, 0x10 - (len & 0x0F), 1, f);
}
ofs += (len + 15) & (~0x0F);
} else
{
insex[i-1].length = 0;
}
}
// Updating parapointers
fseek(f, ofs0, SEEK_SET);
fwrite(insptr, nbi, 2, f);
fwrite(patptr, nbp, 2, f);
fseek(f, ofs1, SEEK_SET);
fwrite(insex, 0x50, nbi, f);
fclose(f);
return TRUE;
}
#ifdef _MSC_VER
#pragma warning(default:4100)
#endif
#endif // MODPLUG_NO_FILESAVE

@ -0,0 +1,186 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#pragma pack(1)
typedef struct tagSTMNOTE
{
BYTE note;
BYTE insvol;
BYTE volcmd;
BYTE cmdinf;
} STMNOTE;
// Raw STM sampleinfo struct:
typedef struct tagSTMSAMPLE
{
CHAR filename[14]; // Can't have long comments - just filename comments :)
WORD reserved; // ISA in memory when in ST 2
WORD length; // Sample length
WORD loopbeg; // Loop start point
WORD loopend; // Loop end point
BYTE volume; // Volume
BYTE reserved2; // More reserved crap
WORD c2spd; // Good old c2spd
BYTE reserved3[6]; // Yet more of PSi's reserved crap
} STMSAMPLE;
// Raw STM header struct:
typedef struct tagSTMHEADER
{
char songname[20]; // changed from CHAR
char trackername[8]; // !SCREAM! for ST 2.xx // changed from CHAR
CHAR unused; // 0x1A
CHAR filetype; // 1=song, 2=module (only 2 is supported, of course) :)
CHAR ver_major; // Like 2
CHAR ver_minor; // "ditto"
BYTE inittempo; // initspeed= stm inittempo>>4
BYTE numpat; // number of patterns
BYTE globalvol; // <- WoW! a RiGHT TRiANGLE =8*)
BYTE reserved[13]; // More of PSi's internal crap
STMSAMPLE sample[31]; // STM sample data
BYTE patorder[128]; // Docs say 64 - actually 128
} STMHEADER;
#pragma pack()
BOOL CSoundFile::ReadSTM(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
const STMHEADER *phdr = (STMHEADER *)lpStream;
DWORD dwMemPos = 0;
if ((!lpStream) || (dwMemLength < sizeof(STMHEADER))) return FALSE;
if ((phdr->filetype != 2) || (phdr->unused != 0x1A)
|| ((strncasecmp(phdr->trackername, "!SCREAM!", 8))
&& (strncasecmp(phdr->trackername, "BMOD2STM", 8)))) return FALSE;
memcpy(m_szNames[0], phdr->songname, 20);
// Read STM header
m_nType = MOD_TYPE_STM;
m_nSamples = 31;
m_nChannels = 4;
m_nInstruments = 0;
m_nMinPeriod = 64;
m_nMaxPeriod = 0x7FFF;
m_nDefaultSpeed = phdr->inittempo >> 4;
if (m_nDefaultSpeed < 1) m_nDefaultSpeed = 1;
m_nDefaultTempo = 125;
m_nDefaultGlobalVolume = phdr->globalvol << 2;
if (m_nDefaultGlobalVolume > 256) m_nDefaultGlobalVolume = 256;
memcpy(Order, phdr->patorder, 128);
// Setting up channels
for (UINT nSet=0; nSet<4; nSet++)
{
ChnSettings[nSet].dwFlags = 0;
ChnSettings[nSet].nVolume = 64;
ChnSettings[nSet].nPan = (nSet & 1) ? 0x40 : 0xC0;
}
// Reading samples
for (UINT nIns=0; nIns<31; nIns++)
{
MODINSTRUMENT *pIns = &Ins[nIns+1];
const STMSAMPLE *pStm = &phdr->sample[nIns]; // STM sample data
memcpy(pIns->name, pStm->filename, 13);
memcpy(m_szNames[nIns+1], pStm->filename, 12);
pIns->nC4Speed = bswapLE16(pStm->c2spd);
pIns->nGlobalVol = 64;
pIns->nVolume = pStm->volume << 2;
if (pIns->nVolume > 256) pIns->nVolume = 256;
pIns->nLength = bswapLE16(pStm->length);
if ((pIns->nLength < 4) || (!pIns->nVolume)) pIns->nLength = 0;
pIns->nLoopStart = bswapLE16(pStm->loopbeg);
pIns->nLoopEnd = bswapLE16(pStm->loopend);
if ((pIns->nLoopEnd > pIns->nLoopStart) && (pIns->nLoopEnd != 0xFFFF)) pIns->uFlags |= CHN_LOOP;
}
dwMemPos = sizeof(STMHEADER);
for (UINT nOrd=0; nOrd<MAX_ORDERS; nOrd++) if (Order[nOrd] >= 99) Order[nOrd] = 0xFF;
UINT nPatterns = phdr->numpat;
for (UINT nPat=0; nPat<nPatterns; nPat++)
{
if (dwMemPos + 64*4*4 > dwMemLength) return TRUE;
PatternSize[nPat] = 64;
if ((Patterns[nPat] = AllocatePattern(64, m_nChannels)) == NULL) return TRUE;
MODCOMMAND *m = Patterns[nPat];
const STMNOTE *p = (const STMNOTE *)(lpStream + dwMemPos);
for (UINT n=0; n<64*4; n++, p++, m++)
{
UINT note,ins,vol,cmd;
// extract the various information from the 4 bytes that
// make up a single note
note = p->note;
ins = p->insvol >> 3;
vol = (p->insvol & 0x07) + (p->volcmd >> 1);
cmd = p->volcmd & 0x0F;
if ((ins) && (ins < 32)) m->instr = ins;
// special values of [SBYTE0] are handled here ->
// we have no idea if these strange values will ever be encountered
// but it appears as though stms sound correct.
if ((note == 0xFE) || (note == 0xFC)) m->note = 0xFE; else
// if note < 251, then all three bytes are stored in the file
if (note < 0xFC) m->note = (note >> 4)*12 + (note&0xf) + 37;
if (vol <= 64) { m->volcmd = VOLCMD_VOLUME; m->vol = vol; }
m->param = p->cmdinf;
switch(cmd)
{
// Axx set speed to xx
case 1: m->command = CMD_SPEED; m->param >>= 4; break;
// Bxx position jump
case 2: m->command = CMD_POSITIONJUMP; break;
// Cxx patternbreak to row xx
case 3: m->command = CMD_PATTERNBREAK; m->param = (m->param & 0xF0) * 10 + (m->param & 0x0F); break;
// Dxy volumeslide
case 4: m->command = CMD_VOLUMESLIDE; break;
// Exy toneslide down
case 5: m->command = CMD_PORTAMENTODOWN; break;
// Fxy toneslide up
case 6: m->command = CMD_PORTAMENTOUP; break;
// Gxx Tone portamento,speed xx
case 7: m->command = CMD_TONEPORTAMENTO; break;
// Hxy vibrato
case 8: m->command = CMD_VIBRATO; break;
// Ixy tremor, ontime x, offtime y
case 9: m->command = CMD_TREMOR; break;
// Jxy arpeggio
case 10: m->command = CMD_ARPEGGIO; break;
// Kxy Dual command H00 & Dxy
case 11: m->command = CMD_VIBRATOVOL; break;
// Lxy Dual command G00 & Dxy
case 12: m->command = CMD_TONEPORTAVOL; break;
// Xxx amiga command 8xx
case 0x18: m->command = CMD_PANNING8; break;
default:
m->command = m->param = 0;
}
}
dwMemPos += 64*4*4;
}
// Reading Samples
for (UINT nSmp=1; nSmp<=31; nSmp++)
{
MODINSTRUMENT *pIns = &Ins[nSmp];
dwMemPos = (dwMemPos + 15) & (~15);
if (pIns->nLength)
{
UINT nPos = ((UINT)phdr->sample[nSmp-1].reserved) << 4;
if ((nPos >= sizeof(STMHEADER)) && (nPos+pIns->nLength <= dwMemLength)) dwMemPos = nPos;
if (dwMemPos < dwMemLength)
{
dwMemPos += ReadSample(pIns, RS_PCM8S, (LPSTR)(lpStream+dwMemPos),dwMemLength-dwMemPos);
}
}
}
return TRUE;
}

@ -0,0 +1,224 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
//#pragma warning(disable:4244)
#define ULT_16BIT 0x04
#define ULT_LOOP 0x08
#define ULT_BIDI 0x10
#pragma pack(1)
// Raw ULT header struct:
typedef struct tagULTHEADER
{
char id[15]; // changed from CHAR
char songtitle[32]; // changed from CHAR
BYTE reserved;
} ULTHEADER;
// Raw ULT sampleinfo struct:
typedef struct tagULTSAMPLE
{
CHAR samplename[32];
CHAR dosname[12];
LONG loopstart;
LONG loopend;
LONG sizestart;
LONG sizeend;
BYTE volume;
BYTE flags;
WORD finetune;
} ULTSAMPLE;
#pragma pack()
BOOL CSoundFile::ReadUlt(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
ULTHEADER *pmh = (ULTHEADER *)lpStream;
ULTSAMPLE *pus;
UINT nos, nop;
DWORD dwMemPos = 0;
// try to read module header
if ((!lpStream) || (dwMemLength < 0x100)) return FALSE;
if (strncmp(pmh->id,"MAS_UTrack_V00",14)) return FALSE;
// Warning! Not supported ULT format, trying anyway
// if ((pmh->id[14] < '1') || (pmh->id[14] > '4')) return FALSE;
m_nType = MOD_TYPE_ULT;
m_nDefaultSpeed = 6;
m_nDefaultTempo = 125;
memcpy(m_szNames[0], pmh->songtitle, 32);
m_szNames[0][31] = '\0';
// read songtext
dwMemPos = sizeof(ULTHEADER);
if ((pmh->reserved) && (dwMemPos + pmh->reserved * 32 < dwMemLength))
{
UINT len = pmh->reserved * 32;
m_lpszSongComments = new char[len + 1 + pmh->reserved];
if (m_lpszSongComments)
{
for (UINT l=0; l<pmh->reserved; l++)
{
memcpy(m_lpszSongComments+l*33, lpStream+dwMemPos+l*32, 32);
m_lpszSongComments[l*33+32] = 0x0D;
}
m_lpszSongComments[len] = 0;
}
dwMemPos += len;
}
if (dwMemPos >= dwMemLength) return TRUE;
nos = lpStream[dwMemPos++];
m_nSamples = nos;
if (m_nSamples >= MAX_SAMPLES) m_nSamples = MAX_SAMPLES-1;
UINT smpsize = 64;
if (pmh->id[14] >= '4') smpsize += 2;
if (dwMemPos + nos*smpsize + 256 + 2 > dwMemLength) return TRUE;
for (UINT ins=1; ins<=nos; ins++, dwMemPos+=smpsize) if (ins<=m_nSamples)
{
pus = (ULTSAMPLE *)(lpStream+dwMemPos);
MODINSTRUMENT *pins = &Ins[ins];
memcpy(m_szNames[ins], pus->samplename, 32);
m_szNames[ins][31] = '\0';
memcpy(pins->name, pus->dosname, 12);
pins->nLoopStart = pus->loopstart;
pins->nLoopEnd = pus->loopend;
pins->nLength = pus->sizeend - pus->sizestart;
pins->nVolume = pus->volume;
pins->nGlobalVol = 64;
pins->nC4Speed = 8363;
if (pmh->id[14] >= '4')
{
pins->nC4Speed = pus->finetune;
}
if (pus->flags & ULT_LOOP) pins->uFlags |= CHN_LOOP;
if (pus->flags & ULT_BIDI) pins->uFlags |= CHN_PINGPONGLOOP;
if (pus->flags & ULT_16BIT)
{
pins->uFlags |= CHN_16BIT;
pins->nLoopStart >>= 1;
pins->nLoopEnd >>= 1;
}
}
memcpy(Order, lpStream+dwMemPos, 256);
dwMemPos += 256;
m_nChannels = lpStream[dwMemPos] + 1;
nop = lpStream[dwMemPos+1] + 1;
dwMemPos += 2;
if (m_nChannels > 32) m_nChannels = 32;
// Default channel settings
for (UINT nSet=0; nSet<m_nChannels; nSet++)
{
ChnSettings[nSet].nVolume = 64;
ChnSettings[nSet].nPan = (nSet & 1) ? 0x40 : 0xC0;
}
// read pan position table for v1.5 and higher
if(pmh->id[14]>='3')
{
if (dwMemPos + m_nChannels > dwMemLength) return TRUE;
for(UINT t=0; t<m_nChannels; t++)
{
ChnSettings[t].nPan = (lpStream[dwMemPos++] << 4) + 8;
if (ChnSettings[t].nPan > 256) ChnSettings[t].nPan = 256;
}
}
// Allocating Patterns
for (UINT nAllocPat=0; nAllocPat<nop; nAllocPat++)
{
if (nAllocPat < MAX_PATTERNS)
{
PatternSize[nAllocPat] = 64;
Patterns[nAllocPat] = AllocatePattern(64, m_nChannels);
}
}
// Reading Patterns
for (UINT nChn=0; nChn<m_nChannels; nChn++)
{
for (UINT nPat=0; nPat<nop; nPat++)
{
MODCOMMAND *pat = NULL;
if (nPat < MAX_PATTERNS)
{
pat = Patterns[nPat];
if (pat) pat += nChn;
}
UINT row = 0;
while (row < 64)
{
if (dwMemPos + 6 > dwMemLength) return TRUE;
UINT rep = 1;
UINT note = lpStream[dwMemPos++];
if (note == 0xFC)
{
rep = lpStream[dwMemPos];
note = lpStream[dwMemPos+1];
dwMemPos += 2;
}
UINT instr = lpStream[dwMemPos++];
UINT eff = lpStream[dwMemPos++];
UINT dat1 = lpStream[dwMemPos++];
UINT dat2 = lpStream[dwMemPos++];
UINT cmd1 = eff & 0x0F;
UINT cmd2 = eff >> 4;
if (cmd1 == 0x0C) dat1 >>= 2; else
if (cmd1 == 0x0B) { cmd1 = dat1 = 0; }
if (cmd2 == 0x0C) dat2 >>= 2; else
if (cmd2 == 0x0B) { cmd2 = dat2 = 0; }
while ((rep != 0) && (row < 64))
{
if (pat)
{
pat->instr = instr;
if (note) pat->note = note + 36;
if (cmd1 | dat1)
{
if (cmd1 == 0x0C)
{
pat->volcmd = VOLCMD_VOLUME;
pat->vol = dat1;
} else
{
pat->command = cmd1;
pat->param = dat1;
ConvertModCommand(pat);
}
}
if (cmd2 == 0x0C)
{
pat->volcmd = VOLCMD_VOLUME;
pat->vol = dat2;
} else
if ((cmd2 | dat2) && (!pat->command))
{
pat->command = cmd2;
pat->param = dat2;
ConvertModCommand(pat);
}
pat += m_nChannels;
}
row++;
rep--;
}
}
}
}
// Reading Instruments
for (UINT smp=1; smp<=m_nSamples; smp++) if (Ins[smp].nLength)
{
if (dwMemPos >= dwMemLength) return TRUE;
UINT flags = (Ins[smp].uFlags & CHN_16BIT) ? RS_PCM16S : RS_PCM8S;
dwMemPos += ReadSample(&Ins[smp], flags, (LPSTR)(lpStream+dwMemPos), dwMemLength - dwMemPos);
}
return TRUE;
}

@ -0,0 +1,312 @@
/*
* This source code is public domain.
*
* Epic Games Unreal UMX container loading for libmodplug
* Written by O. Sezer <sezero@users.sourceforge.net>
* UPKG parsing partially based on Unreal Media Ripper (UMR) v0.3
* by Andy Ward <wardwh@swbell.net>, with additional updates
* by O. Sezer - see git repo at https://github.com/sezero/umr/
* Retrieves the offset, size and object type directly from umx.
*/
#include "stdafx.h"
#include "sndfile.h"
typedef LONG fci_t; /* FCompactIndex */
#define UPKG_HDR_TAG 0x9e2a83c1
struct _genhist { /* for upkg versions >= 68 */
LONG export_count;
LONG name_count;
};
struct upkg_hdr {
DWORD tag; /* UPKG_HDR_TAG */
LONG file_version;
DWORD pkg_flags;
LONG name_count; /* number of names in name table (>= 0) */
LONG name_offset; /* offset to name table (>= 0) */
LONG export_count; /* num. exports in export table (>= 0) */
LONG export_offset; /* offset to export table (>= 0) */
LONG import_count; /* num. imports in export table (>= 0) */
LONG import_offset; /* offset to import table (>= 0) */
/* number of GUIDs in heritage table (>= 1) and table's offset:
* only with versions < 68. */
LONG heritage_count;
LONG heritage_offset;
/* with versions >= 68: a GUID, a dword for generation count
* and export_count and name_count dwords for each generation: */
DWORD guid[4];
LONG generation_count;
#define UPKG_HDR_SIZE 64 /* 64 bytes up until here */
/*struct _genhist *gen;*/
};
#define UMUSIC_IT 0
#define UMUSIC_S3M 1
#define UMUSIC_XM 2
#define UMUSIC_MOD 3
#define UMUSIC_WAV 4
#define UMUSIC_MP2 5
static const char *mustype[] = {
"IT", "S3M", "XM", "MOD",
NULL
};
/* decode an FCompactIndex.
* original documentation by Tim Sweeney was at
* http://unreal.epicgames.com/Packages.htm
* also see Unreal Wiki:
* http://wiki.beyondunreal.com/Legacy:Package_File_Format/Data_Details
*/
static fci_t get_fci (const char *in, int *pos)
{
LONG a;
int size;
size = 1;
a = in[0] & 0x3f;
if (in[0] & 0x40) {
size++;
a |= (in[1] & 0x7f) << 6;
if (in[1] & 0x80) {
size++;
a |= (in[2] & 0x7f) << 13;
if (in[2] & 0x80) {
size++;
a |= (in[3] & 0x7f) << 20;
if (in[3] & 0x80) {
size++;
a |= (in[4] & 0x3f) << 27;
}
}
}
}
if (in[0] & 0x80)
a = -a;
*pos += size;
return a;
}
static int get_objtype (const BYTE *membase, LONG memlen,
LONG ofs, int type)
{
if (type == UMUSIC_IT) {
_retry:
if (memcmp(membase + ofs, "IMPM", 4) == 0)
return UMUSIC_IT;
return -1;
}
if (type == UMUSIC_XM) {
if (memcmp(membase + ofs, "Extended Module: ", 17) != 0)
return -1;
if (*(membase + ofs + 37) != 0x1a) return -1;
return UMUSIC_XM;
}
if (type == UMUSIC_S3M) {
if (memcmp(membase + ofs + 44, "SCRM", 4) == 0)
return UMUSIC_S3M;
/*return -1;*/
/* SpaceMarines.umx and Starseek.umx from Return to NaPali
* report as "s3m" whereas the actual music format is "it" */
goto _retry;
}
if (type == UMUSIC_MOD) {
membase += ofs + 1080;
if (memcmp(membase, "M.K.", 4) == 0 || memcmp(membase, "M!K!", 4) == 0)
return UMUSIC_MOD;
return -1;
}
return -1;
}
static int read_export (const BYTE *membase, LONG memlen,
const struct upkg_hdr *hdr,
LONG *ofs, LONG *objsize)
{
char buf[40];
int idx = 0, t;
memcpy(buf, membase + *ofs, 40);
if (hdr->file_version < 40) idx += 8; /* 00 00 00 00 00 00 00 00 */
if (hdr->file_version < 60) idx += 16; /* 81 00 00 00 00 00 FF FF FF FF FF FF FF FF 00 00 */
get_fci(&buf[idx], &idx); /* skip junk */
t = get_fci(&buf[idx], &idx); /* type_name */
if (hdr->file_version > 61) idx += 4; /* skip export size */
*objsize = get_fci(&buf[idx], &idx);
*ofs += idx; /* offset for real data */
return t; /* return type_name index */
}
static int read_typname(const BYTE *membase, LONG memlen,
const struct upkg_hdr *hdr,
int idx, char *out)
{
int i, s;
long l;
char buf[64];
if (idx >= hdr->name_count) return -1;
buf[63] = '\0';
for (i = 0, l = 0; i <= idx; i++) {
memcpy(buf, membase + hdr->name_offset + l, 63);
if (hdr->file_version >= 64) {
s = *(signed char *)buf; /* numchars *including* terminator */
if (s <= 0 || s > 64) return -1;
l += s + 5; /* 1 for buf[0], 4 for int32_t name_flags */
} else {
l += (long)strlen(buf);
l += 5; /* 1 for terminator, 4 for int32_t name_flags */
}
}
strcpy(out, (hdr->file_version >= 64)? &buf[1] : buf);
return 0;
}
static int probe_umx (const BYTE *membase, LONG memlen,
const struct upkg_hdr *hdr,
LONG *ofs, LONG *objsize)
{
int i, idx, t;
LONG s, pos;
char buf[64];
/* Find the offset and size of the first IT, S3M or XM
* by parsing the exports table. The umx files should
* have only one export. Kran32.umx from Unreal has two,
* but both pointing to the same music. */
s = memlen - hdr->export_offset;
if (s <= 0) return -1;
if (s > 64) s = 64;
memcpy(buf, membase + hdr->export_offset, s);
for (; s < 64; ++s) buf[s] = 0x0; /* really? */
idx = 0;
get_fci(&buf[idx], &idx); /* skip class_index */
get_fci(&buf[idx], &idx); /* skip super_index */
if (hdr->file_version >= 60) idx += 4; /* skip int32 package_index */
get_fci(&buf[idx], &idx); /* skip object_name */
idx += 4; /* skip int32 object_flags */
s = get_fci(&buf[idx], &idx); /* get serial_size */
if (s <= 0) return -1;
pos = get_fci(&buf[idx],&idx); /* get serial_offset */
if (pos < 0 || pos > memlen - 40) return -1;
if ((t = read_export(membase, memlen, hdr, &pos, &s)) < 0) return -1;
if (s <= 0 || s > memlen - pos) return -1;
if (read_typname(membase, memlen, hdr, t, buf) < 0) return -1;
for (i = 0; mustype[i] != NULL; i++) {
if (!strcasecmp(buf, mustype[i])) {
t = i;
break;
}
}
if (mustype[i] == NULL) return -1;
if ((t = get_objtype(membase, memlen, pos, t)) < 0) return -1;
*ofs = pos;
*objsize = s;
return t;
}
static int probe_header (void *header)
{
struct upkg_hdr *hdr;
unsigned char *p;
DWORD *swp;
int i;
/* byte swap the header - all members are 32 bit LE values */
p = (unsigned char *) header;
swp = (DWORD *) header;
for (i = 0; i < UPKG_HDR_SIZE/4; i++, p += 4) {
swp[i] = p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24);
}
hdr = (struct upkg_hdr *) header;
if (hdr->tag != UPKG_HDR_TAG) {
return -1;
}
if (hdr->name_count < 0 ||
hdr->name_offset < 0 ||
hdr->export_count < 0 ||
hdr->export_offset < 0 ||
hdr->import_count < 0 ||
hdr->import_offset < 0 ) {
return -1;
}
#if 0
return 0;
#else
switch (hdr->file_version) {
case 35: case 37: /* Unreal beta - */
case 40: case 41: /* 1998 */
case 61:/* Unreal */
case 62:/* Unreal Tournament */
case 63:/* Return to NaPali */
case 64:/* Unreal Tournament */
case 66:/* Unreal Tournament */
case 68:/* Unreal Tournament */
case 69:/* Tactical Ops */
case 83:/* Mobile Forces */
return 0;
}
return -1;/* Unknown upkg version for an UMX */
#endif
}
static int process_upkg (const BYTE *membase, LONG memlen,
LONG *ofs, LONG *objsize)
{
char header[UPKG_HDR_SIZE];
memcpy(header, membase, UPKG_HDR_SIZE);
if (probe_header(header) < 0)
return -1;
return probe_umx(membase, memlen, (struct upkg_hdr *)header, ofs, objsize);
}
BOOL CSoundFile::ReadUMX(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
int type;
LONG ofs = 0, size = 0;
if (!lpStream || dwMemLength < 0x800 || dwMemLength > 0x7fffffff)
return FALSE;
type = process_upkg(lpStream, (LONG)dwMemLength, &ofs, &size);
if (type < 0) return FALSE;
// Rip Mods from UMX
switch (type) {
case UMUSIC_IT: return ReadIT(lpStream + ofs, size);
case UMUSIC_S3M: return ReadS3M(lpStream + ofs, size);
case UMUSIC_XM: return ReadXM(lpStream + ofs, size);
case UMUSIC_MOD: return ReadMod(lpStream + ofs, size);
}
return FALSE;
}

@ -0,0 +1,220 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
#ifndef WAVE_FORMAT_EXTENSIBLE
#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
#endif
/////////////////////////////////////////////////////////////
// WAV file support
BOOL CSoundFile::ReadWav(const BYTE *lpStream, DWORD dwMemLength)
//---------------------------------------------------------------
{
DWORD dwMemPos = 0;
WAVEFILEHEADER *phdr = (WAVEFILEHEADER *)lpStream;
WAVEFORMATHEADER *pfmt = (WAVEFORMATHEADER *)(lpStream + sizeof(WAVEFILEHEADER));
if ((!lpStream) || (dwMemLength < (DWORD)sizeof(WAVEFILEHEADER))) return FALSE;
if ((phdr->id_RIFF != IFFID_RIFF) || (phdr->id_WAVE != IFFID_WAVE)
|| (pfmt->id_fmt != IFFID_fmt)) return FALSE;
dwMemPos = sizeof(WAVEFILEHEADER) + 8 + pfmt->hdrlen;
if ((dwMemPos + 8 >= dwMemLength)
|| ((pfmt->format != WAVE_FORMAT_PCM) && (pfmt->format != WAVE_FORMAT_EXTENSIBLE))
|| (pfmt->channels > 4)
|| (!pfmt->channels)
|| (!pfmt->freqHz)
|| (pfmt->bitspersample & 7)
|| (pfmt->bitspersample < 8)
|| (pfmt->bitspersample > 32)) return FALSE;
WAVEDATAHEADER *pdata;
for (;;)
{
pdata = (WAVEDATAHEADER *)(lpStream + dwMemPos);
if (pdata->id_data == IFFID_data) break;
dwMemPos += pdata->length + 8;
if (dwMemPos + 8 >= dwMemLength) return FALSE;
}
m_nType = MOD_TYPE_WAV;
m_nSamples = 0;
m_nInstruments = 0;
m_nChannels = 4;
m_nDefaultSpeed = 8;
m_nDefaultTempo = 125;
m_dwSongFlags |= SONG_LINEARSLIDES; // For no resampling
Order[0] = 0;
Order[1] = 0xFF;
PatternSize[0] = PatternSize[1] = 64;
if ((Patterns[0] = AllocatePattern(64, 4)) == NULL) return TRUE;
if ((Patterns[1] = AllocatePattern(64, 4)) == NULL) return TRUE;
UINT samplesize = (pfmt->channels * pfmt->bitspersample) >> 3;
UINT len = pdata->length, bytelen;
if (len > dwMemLength - 8 - dwMemPos) len = dwMemLength - dwMemPos - 8;
len /= samplesize;
bytelen = len;
if (pfmt->bitspersample >= 16) bytelen *= 2;
if (len > MAX_SAMPLE_LENGTH) len = MAX_SAMPLE_LENGTH;
if (!len) return TRUE;
// Setting up module length
DWORD dwTime = ((len * 50) / pfmt->freqHz) + 1;
DWORD framesperrow = (dwTime + 63) / 63;
if (framesperrow < 4) framesperrow = 4;
UINT norders = 1;
while (framesperrow >= 0x20)
{
Order[norders++] = 1;
Order[norders] = 0xFF;
framesperrow = (dwTime + (64 * norders - 1)) / (64 * norders);
if (norders >= MAX_ORDERS-1) break;
}
m_nDefaultSpeed = framesperrow;
for (UINT iChn=0; iChn<4; iChn++)
{
ChnSettings[iChn].nPan = (iChn & 1) ? 256 : 0;
ChnSettings[iChn].nVolume = 64;
ChnSettings[iChn].dwFlags = 0;
}
// Setting up speed command
MODCOMMAND *pcmd = Patterns[0];
pcmd[0].command = CMD_SPEED;
pcmd[0].param = (BYTE)m_nDefaultSpeed;
pcmd[0].note = 5*12+1;
pcmd[0].instr = 1;
pcmd[1].note = pcmd[0].note;
pcmd[1].instr = pcmd[0].instr;
m_nSamples = pfmt->channels;
// Support for Multichannel Wave
for (UINT nChn=0; nChn<m_nSamples; nChn++)
{
MODINSTRUMENT *pins = &Ins[nChn+1];
pcmd[nChn].note = pcmd[0].note;
pcmd[nChn].instr = (BYTE)(nChn+1);
pins->nLength = len;
pins->nC4Speed = pfmt->freqHz;
pins->nVolume = 256;
pins->nPan = 128;
pins->nGlobalVol = 64;
pins->uFlags = (WORD)((pfmt->bitspersample >= 16) ? CHN_16BIT : 0);
pins->uFlags |= CHN_PANNING;
if (m_nSamples > 1)
{
switch(nChn)
{
case 0: pins->nPan = 0; break;
case 1: pins->nPan = 256; break;
case 2: pins->nPan = (WORD)((m_nSamples == 3) ? 128 : 64); pcmd[nChn].command = CMD_S3MCMDEX; pcmd[nChn].param = 0x91; break;
case 3: pins->nPan = 192; pcmd[nChn].command = CMD_S3MCMDEX; pcmd[nChn].param = 0x91; break;
default: pins->nPan = 128; break;
}
}
if ((pins->pSample = AllocateSample(bytelen+8)) == NULL) return TRUE;
if (pfmt->bitspersample >= 16)
{
int slsize = pfmt->bitspersample >> 3;
signed short *p = (signed short *)pins->pSample;
signed char *psrc = (signed char *)(lpStream+dwMemPos+8+nChn*slsize+slsize-2);
for (UINT i=0; i<len; i++)
{
p[i] = *((signed short *)psrc);
psrc += samplesize;
}
p[len+1] = p[len] = p[len-1];
} else
{
signed char *p = (signed char *)pins->pSample;
signed char *psrc = (signed char *)(lpStream+dwMemPos+8+nChn);
for (UINT i=0; i<len; i++)
{
p[i] = (signed char)((*psrc) + 0x80);
psrc += samplesize;
}
p[len+1] = p[len] = p[len-1];
}
}
return TRUE;
}
////////////////////////////////////////////////////////////////////////
// IMA ADPCM Support
#pragma pack(1)
typedef struct IMAADPCMBLOCK
{
WORD sample;
BYTE index;
BYTE Reserved;
} DVI_ADPCMBLOCKHEADER;
#pragma pack()
static const int gIMAUnpackTable[90] =
{
7, 8, 9, 10, 11, 12, 13, 14,
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767, 0
};
BOOL IMAADPCMUnpack16(signed short *pdest, UINT nLen, LPBYTE psrc, DWORD dwBytes, UINT pkBlkAlign)
//------------------------------------------------------------------------------------------------
{
static const int gIMAIndexTab[8] = { -1, -1, -1, -1, 2, 4, 6, 8 };
UINT nPos;
int value;
if ((nLen < 4) || (!pdest) || (!psrc)
|| (pkBlkAlign < 5) || (pkBlkAlign > dwBytes)) return FALSE;
nPos = 0;
while ((nPos < nLen) && (dwBytes > 4))
{
int nIndex;
value = *((short int *)psrc);
nIndex = psrc[2];
psrc += 4;
dwBytes -= 4;
pdest[nPos++] = (short int)value;
for (UINT i=0; ((i<(pkBlkAlign-4)*2) && (nPos < nLen) && (dwBytes)); i++)
{
BYTE delta;
if (i & 1)
{
delta = (BYTE)(((*(psrc++)) >> 4) & 0x0F);
dwBytes--;
} else
{
delta = (BYTE)((*psrc) & 0x0F);
}
int v = gIMAUnpackTable[nIndex] >> 3;
if (delta & 1) v += gIMAUnpackTable[nIndex] >> 2;
if (delta & 2) v += gIMAUnpackTable[nIndex] >> 1;
if (delta & 4) v += gIMAUnpackTable[nIndex];
if (delta & 8) value -= v; else value += v;
nIndex += gIMAIndexTab[delta & 7];
if (nIndex < 0) nIndex = 0; else
if (nIndex > 88) nIndex = 88;
if (value > 32767) value = 32767; else
if (value < -32768) value = -32768;
pdest[nPos++] = (short int)value;
}
}
return TRUE;
}

@ -0,0 +1,893 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
#include "stdafx.h"
#include "sndfile.h"
////////////////////////////////////////////////////////
// FastTracker II XM file support
#ifdef MSC_VER
#pragma warning(disable:4244)
#endif
#pragma pack(1)
typedef struct tagXMFILEHEADER
{
DWORD size;
WORD norder;
WORD restartpos;
WORD channels;
WORD patterns;
WORD instruments;
WORD flags;
WORD speed;
WORD tempo;
BYTE order[256];
} XMFILEHEADER;
typedef struct tagXMINSTRUMENTHEADER
{
DWORD size;
CHAR name[22];
BYTE type;
BYTE samples;
BYTE samplesh;
} XMINSTRUMENTHEADER;
typedef struct tagXMSAMPLEHEADER
{
DWORD shsize;
BYTE snum[96];
WORD venv[24];
WORD penv[24];
BYTE vnum, pnum;
BYTE vsustain, vloops, vloope, psustain, ploops, ploope;
BYTE vtype, ptype;
BYTE vibtype, vibsweep, vibdepth, vibrate;
WORD volfade;
WORD res;
BYTE reserved1[20];
} XMSAMPLEHEADER;
typedef struct tagXMSAMPLESTRUCT
{
DWORD samplen;
DWORD loopstart;
DWORD looplen;
BYTE vol;
signed char finetune;
BYTE type;
BYTE pan;
signed char relnote;
BYTE res;
char name[22];
} XMSAMPLESTRUCT;
#pragma pack()
BOOL CSoundFile::ReadXM(const BYTE *lpStream, DWORD dwMemLength)
//--------------------------------------------------------------
{
XMSAMPLEHEADER xmsh;
XMSAMPLESTRUCT xmss;
DWORD dwMemPos, dwHdrSize;
WORD norders=0, restartpos=0, channels=0, patterns=0, instruments=0;
WORD xmflags=0, deftempo=125, defspeed=6;
BOOL InstUsed[256];
BYTE channels_used[MAX_CHANNELS];
BYTE pattern_map[256];
BOOL samples_used[MAX_SAMPLES];
UINT unused_samples;
tagXMFILEHEADER xmhead;
m_nChannels = 0;
if ((!lpStream) || (dwMemLength < 0x200)) return FALSE;
if (strncmp((LPCSTR)lpStream, "Extended Module:", 16)) return FALSE;
memcpy(m_szNames[0], lpStream+17, 20);
xmhead = *(tagXMFILEHEADER *)(lpStream+60);
dwHdrSize = bswapLE32(xmhead.size);
norders = bswapLE16(xmhead.norder);
if ((!norders) || (norders > MAX_ORDERS)) return FALSE;
restartpos = bswapLE16(xmhead.restartpos);
channels = bswapLE16(xmhead.channels);
if ((!channels) || (channels > 64)) return FALSE;
m_nType = MOD_TYPE_XM;
m_nMinPeriod = 27;
m_nMaxPeriod = 54784;
m_nChannels = channels;
if (restartpos < norders) m_nRestartPos = restartpos;
patterns = bswapLE16(xmhead.patterns);
if (patterns > 256) patterns = 256;
instruments = bswapLE16(xmhead.instruments);
if (instruments >= MAX_INSTRUMENTS) instruments = MAX_INSTRUMENTS-1;
m_nInstruments = instruments;
m_nSamples = 0;
xmflags = bswapLE16(xmhead.flags);
if (xmflags & 1) m_dwSongFlags |= SONG_LINEARSLIDES;
if (xmflags & 0x1000) m_dwSongFlags |= SONG_EXFILTERRANGE;
defspeed = bswapLE16(xmhead.speed);
deftempo = bswapLE16(xmhead.tempo);
if ((deftempo >= 32) && (deftempo < 256)) m_nDefaultTempo = deftempo;
if ((defspeed > 0) && (defspeed < 40)) m_nDefaultSpeed = defspeed;
memcpy(Order, lpStream+80, norders);
memset(InstUsed, 0, sizeof(InstUsed));
if (patterns > MAX_PATTERNS)
{
UINT i, j;
for (i=0; i<norders; i++)
{
if (Order[i] < patterns) InstUsed[Order[i]] = TRUE;
}
j = 0;
for (i=0; i<256; i++)
{
if (InstUsed[i]) pattern_map[i] = j++;
}
for (i=0; i<256; i++)
{
if (!InstUsed[i])
{
pattern_map[i] = (j < MAX_PATTERNS) ? j : 0xFE;
j++;
}
}
for (i=0; i<norders; i++)
{
Order[i] = pattern_map[Order[i]];
}
} else
{
for (UINT i=0; i<256; i++) pattern_map[i] = i;
}
memset(InstUsed, 0, sizeof(InstUsed));
dwMemPos = dwHdrSize + 60;
if (dwMemPos + 8 >= dwMemLength) return TRUE;
// Reading patterns
memset(channels_used, 0, sizeof(channels_used));
for (UINT ipat=0; ipat<patterns; ipat++)
{
UINT ipatmap = pattern_map[ipat];
DWORD dwSize = 0;
WORD rows=64, packsize=0;
dwSize = bswapLE32(*((DWORD *)(lpStream+dwMemPos)));
while ((dwMemPos + dwSize >= dwMemLength) || (dwSize & 0xFFFFFF00))
{
if (dwMemPos + 4 >= dwMemLength) break;
dwMemPos++;
dwSize = bswapLE32(*((DWORD *)(lpStream+dwMemPos)));
}
rows = bswapLE16(*((WORD *)(lpStream+dwMemPos+5)));
if ((!rows) || (rows > 256)) rows = 64;
packsize = bswapLE16(*((WORD *)(lpStream+dwMemPos+7)));
if (dwMemPos + dwSize + 4 > dwMemLength) return TRUE;
dwMemPos += dwSize;
if (dwMemPos + packsize + 4 > dwMemLength) return TRUE;
MODCOMMAND *p;
if (ipatmap < MAX_PATTERNS)
{
PatternSize[ipatmap] = rows;
if ((Patterns[ipatmap] = AllocatePattern(rows, m_nChannels)) == NULL) return TRUE;
if (!packsize) continue;
p = Patterns[ipatmap];
} else p = NULL;
const BYTE *src = lpStream+dwMemPos;
UINT j=0;
for (UINT row=0; row<rows; row++)
{
for (UINT chn=0; chn<m_nChannels; chn++)
{
if ((p) && (j < packsize))
{
BYTE b = src[j++];
UINT vol = 0;
if (b & 0x80)
{
if (b & 1) p->note = src[j++];
if (b & 2) p->instr = src[j++];
if (b & 4) vol = src[j++];
if (b & 8) p->command = src[j++];
if (b & 16) p->param = src[j++];
} else
{
p->note = b;
p->instr = src[j++];
vol = src[j++];
p->command = src[j++];
p->param = src[j++];
}
if (p->note == 97) p->note = 0xFF; else
if ((p->note) && (p->note < 97)) p->note += 12;
if (p->note) channels_used[chn] = 1;
if (p->command | p->param) ConvertModCommand(p);
if (p->instr == 0xff) p->instr = 0;
if (p->instr) InstUsed[p->instr] = TRUE;
if ((vol >= 0x10) && (vol <= 0x50))
{
p->volcmd = VOLCMD_VOLUME;
p->vol = vol - 0x10;
} else
if (vol >= 0x60)
{
UINT v = vol & 0xF0;
vol &= 0x0F;
p->vol = vol;
switch(v)
{
// 60-6F: Volume Slide Down
case 0x60: p->volcmd = VOLCMD_VOLSLIDEDOWN; break;
// 70-7F: Volume Slide Up:
case 0x70: p->volcmd = VOLCMD_VOLSLIDEUP; break;
// 80-8F: Fine Volume Slide Down
case 0x80: p->volcmd = VOLCMD_FINEVOLDOWN; break;
// 90-9F: Fine Volume Slide Up
case 0x90: p->volcmd = VOLCMD_FINEVOLUP; break;
// A0-AF: Set Vibrato Speed
case 0xA0: p->volcmd = VOLCMD_VIBRATOSPEED; break;
// B0-BF: Vibrato
case 0xB0: p->volcmd = VOLCMD_VIBRATO; break;
// C0-CF: Set Panning
case 0xC0: p->volcmd = VOLCMD_PANNING; p->vol = (vol << 2) + 2; break;
// D0-DF: Panning Slide Left
case 0xD0: p->volcmd = VOLCMD_PANSLIDELEFT; break;
// E0-EF: Panning Slide Right
case 0xE0: p->volcmd = VOLCMD_PANSLIDERIGHT; break;
// F0-FF: Tone Portamento
case 0xF0: p->volcmd = VOLCMD_TONEPORTAMENTO; break;
}
}
p++;
} else
if (j < packsize)
{
BYTE b = src[j++];
if (b & 0x80)
{
if (b & 1) j++;
if (b & 2) j++;
if (b & 4) j++;
if (b & 8) j++;
if (b & 16) j++;
} else j += 4;
} else break;
}
}
dwMemPos += packsize;
}
// Wrong offset check
while (dwMemPos + 4 < dwMemLength)
{
DWORD d = bswapLE32(*((DWORD *)(lpStream+dwMemPos)));
if (d < 0x300) break;
dwMemPos++;
}
memset(samples_used, 0, sizeof(samples_used));
unused_samples = 0;
// Reading instruments
for (UINT iIns=1; iIns<=instruments; iIns++)
{
XMINSTRUMENTHEADER *pih;
BYTE flags[32];
DWORD samplesize[32];
UINT samplemap[32];
WORD nsamples;
if (dwMemPos + sizeof(XMINSTRUMENTHEADER) >= dwMemLength) return TRUE;
pih = (XMINSTRUMENTHEADER *)(lpStream+dwMemPos);
if (dwMemPos + bswapLE32(pih->size) > dwMemLength) return TRUE;
if ((Headers[iIns] = new INSTRUMENTHEADER) == NULL) continue;
memset(Headers[iIns], 0, sizeof(INSTRUMENTHEADER));
memcpy(Headers[iIns]->name, pih->name, 22);
if ((nsamples = pih->samples) > 0)
{
if (dwMemPos + sizeof(XMSAMPLEHEADER) > dwMemLength) return TRUE;
memcpy(&xmsh, lpStream+dwMemPos+sizeof(XMINSTRUMENTHEADER), sizeof(XMSAMPLEHEADER));
xmsh.shsize = bswapLE32(xmsh.shsize);
for (int i = 0; i < 24; ++i) {
xmsh.venv[i] = bswapLE16(xmsh.venv[i]);
xmsh.penv[i] = bswapLE16(xmsh.penv[i]);
}
xmsh.volfade = bswapLE16(xmsh.volfade);
xmsh.res = bswapLE16(xmsh.res);
dwMemPos += bswapLE32(pih->size);
} else
{
if (bswapLE32(pih->size)) dwMemPos += bswapLE32(pih->size);
else dwMemPos += sizeof(XMINSTRUMENTHEADER);
continue;
}
memset(samplemap, 0, sizeof(samplemap));
if (nsamples > 32) return TRUE;
UINT newsamples = m_nSamples;
for (UINT nmap=0; nmap<nsamples; nmap++)
{
UINT n = m_nSamples+nmap+1;
if (n >= MAX_SAMPLES)
{
n = m_nSamples;
while (n > 0)
{
if (!Ins[n].pSample)
{
for (UINT xmapchk=0; xmapchk < nmap; xmapchk++)
{
if (samplemap[xmapchk] == n) goto alreadymapped;
}
for (UINT clrs=1; clrs<iIns; clrs++) if (Headers[clrs])
{
INSTRUMENTHEADER *pks = Headers[clrs];
for (UINT ks=0; ks<128; ks++)
{
if (pks->Keyboard[ks] == n) pks->Keyboard[ks] = 0;
}
}
break;
}
alreadymapped:
n--;
}
#ifndef MODPLUG_FASTSOUNDLIB
// Damn! more than 200 samples: look for duplicates
if (!n)
{
if (!unused_samples)
{
unused_samples = DetectUnusedSamples(samples_used);
if (!unused_samples) unused_samples = 0xFFFF;
}
if ((unused_samples) && (unused_samples != 0xFFFF))
{
for (UINT iext=m_nSamples; iext>=1; iext--) if (!samples_used[iext])
{
unused_samples--;
samples_used[iext] = TRUE;
DestroySample(iext);
n = iext;
for (UINT mapchk=0; mapchk<nmap; mapchk++)
{
if (samplemap[mapchk] == n) samplemap[mapchk] = 0;
}
for (UINT clrs=1; clrs<iIns; clrs++) if (Headers[clrs])
{
INSTRUMENTHEADER *pks = Headers[clrs];
for (UINT ks=0; ks<128; ks++)
{
if (pks->Keyboard[ks] == n) pks->Keyboard[ks] = 0;
}
}
memset(&Ins[n], 0, sizeof(Ins[0]));
break;
}
}
}
#endif // MODPLUG_FASTSOUNDLIB
}
if (newsamples < n) newsamples = n;
samplemap[nmap] = n;
}
m_nSamples = newsamples;
// Reading Volume Envelope
INSTRUMENTHEADER *penv = Headers[iIns];
penv->nMidiProgram = pih->type;
penv->nFadeOut = xmsh.volfade;
penv->nPan = 128;
penv->nPPC = 5*12;
if (xmsh.vtype & 1) penv->dwFlags |= ENV_VOLUME;
if (xmsh.vtype & 2) penv->dwFlags |= ENV_VOLSUSTAIN;
if (xmsh.vtype & 4) penv->dwFlags |= ENV_VOLLOOP;
if (xmsh.ptype & 1) penv->dwFlags |= ENV_PANNING;
if (xmsh.ptype & 2) penv->dwFlags |= ENV_PANSUSTAIN;
if (xmsh.ptype & 4) penv->dwFlags |= ENV_PANLOOP;
if (xmsh.vnum > 12) xmsh.vnum = 12;
if (xmsh.pnum > 12) xmsh.pnum = 12;
penv->nVolEnv = xmsh.vnum;
if (!xmsh.vnum) penv->dwFlags &= ~ENV_VOLUME;
if (!xmsh.pnum) penv->dwFlags &= ~ENV_PANNING;
penv->nPanEnv = xmsh.pnum;
penv->nVolSustainBegin = penv->nVolSustainEnd = xmsh.vsustain;
if (xmsh.vsustain >= 12) penv->dwFlags &= ~ENV_VOLSUSTAIN;
penv->nVolLoopStart = xmsh.vloops;
penv->nVolLoopEnd = xmsh.vloope;
if (penv->nVolLoopEnd >= 12) penv->nVolLoopEnd = 0;
if (penv->nVolLoopStart >= penv->nVolLoopEnd) penv->dwFlags &= ~ENV_VOLLOOP;
penv->nPanSustainBegin = penv->nPanSustainEnd = xmsh.psustain;
if (xmsh.psustain >= 12) penv->dwFlags &= ~ENV_PANSUSTAIN;
penv->nPanLoopStart = xmsh.ploops;
penv->nPanLoopEnd = xmsh.ploope;
if (penv->nPanLoopEnd >= 12) penv->nPanLoopEnd = 0;
if (penv->nPanLoopStart >= penv->nPanLoopEnd) penv->dwFlags &= ~ENV_PANLOOP;
penv->nGlobalVol = 64;
for (UINT ienv=0; ienv<12; ienv++)
{
penv->VolPoints[ienv] = (WORD)xmsh.venv[ienv*2];
penv->VolEnv[ienv] = (BYTE)xmsh.venv[ienv*2+1];
penv->PanPoints[ienv] = (WORD)xmsh.penv[ienv*2];
penv->PanEnv[ienv] = (BYTE)xmsh.penv[ienv*2+1];
if (ienv)
{
if (penv->VolPoints[ienv] < penv->VolPoints[ienv-1])
{
penv->VolPoints[ienv] &= 0xFF;
penv->VolPoints[ienv] += penv->VolPoints[ienv-1] & 0xFF00;
if (penv->VolPoints[ienv] < penv->VolPoints[ienv-1]) penv->VolPoints[ienv] += 0x100;
}
if (penv->PanPoints[ienv] < penv->PanPoints[ienv-1])
{
penv->PanPoints[ienv] &= 0xFF;
penv->PanPoints[ienv] += penv->PanPoints[ienv-1] & 0xFF00;
if (penv->PanPoints[ienv] < penv->PanPoints[ienv-1]) penv->PanPoints[ienv] += 0x100;
}
}
}
for (UINT j=0; j<96; j++)
{
penv->NoteMap[j+12] = j+1+12;
if (xmsh.snum[j] < nsamples)
penv->Keyboard[j+12] = samplemap[xmsh.snum[j]];
}
// Reading samples
for (UINT ins=0; ins<nsamples; ins++)
{
if ((dwMemPos + sizeof(xmss) > dwMemLength)
|| (dwMemPos + xmsh.shsize > dwMemLength)) return TRUE;
memcpy(&xmss, lpStream+dwMemPos, sizeof(xmss));
xmss.samplen = bswapLE32(xmss.samplen);
xmss.loopstart = bswapLE32(xmss.loopstart);
xmss.looplen = bswapLE32(xmss.looplen);
dwMemPos += xmsh.shsize;
flags[ins] = (xmss.type & 0x10) ? RS_PCM16D : RS_PCM8D;
if (xmss.type & 0x20) flags[ins] = (xmss.type & 0x10) ? RS_STPCM16D : RS_STPCM8D;
samplesize[ins] = xmss.samplen;
if (!samplemap[ins]) continue;
if (xmss.type & 0x10)
{
xmss.looplen >>= 1;
xmss.loopstart >>= 1;
xmss.samplen >>= 1;
}
if (xmss.type & 0x20)
{
xmss.looplen >>= 1;
xmss.loopstart >>= 1;
xmss.samplen >>= 1;
}
if (xmss.samplen > MAX_SAMPLE_LENGTH) xmss.samplen = MAX_SAMPLE_LENGTH;
if (xmss.loopstart >= xmss.samplen) xmss.type &= ~3;
xmss.looplen += xmss.loopstart;
if (xmss.looplen > xmss.samplen) xmss.looplen = xmss.samplen;
if (!xmss.looplen) xmss.type &= ~3;
UINT imapsmp = samplemap[ins];
memcpy(m_szNames[imapsmp], xmss.name, 22);
m_szNames[imapsmp][22] = 0;
MODINSTRUMENT *pins = &Ins[imapsmp];
pins->nLength = (xmss.samplen > MAX_SAMPLE_LENGTH) ? MAX_SAMPLE_LENGTH : xmss.samplen;
pins->nLoopStart = xmss.loopstart;
pins->nLoopEnd = xmss.looplen;
if (pins->nLoopEnd > pins->nLength) pins->nLoopEnd = pins->nLength;
if (pins->nLoopStart >= pins->nLoopEnd)
{
pins->nLoopStart = pins->nLoopEnd = 0;
}
if (xmss.type & 3) pins->uFlags |= CHN_LOOP;
if (xmss.type & 2) pins->uFlags |= CHN_PINGPONGLOOP;
pins->nVolume = xmss.vol << 2;
if (pins->nVolume > 256) pins->nVolume = 256;
pins->nGlobalVol = 64;
if ((xmss.res == 0xAD) && (!(xmss.type & 0x30)))
{
flags[ins] = RS_ADPCM4;
samplesize[ins] = (samplesize[ins]+1)/2 + 16;
}
pins->nFineTune = xmss.finetune;
pins->RelativeTone = (int)xmss.relnote;
pins->nPan = xmss.pan;
pins->uFlags |= CHN_PANNING;
pins->nVibType = xmsh.vibtype;
pins->nVibSweep = xmsh.vibsweep;
pins->nVibDepth = xmsh.vibdepth;
pins->nVibRate = xmsh.vibrate;
memcpy(pins->name, xmss.name, 22);
pins->name[21] = 0;
}
#if 0
if ((xmsh.reserved2 > nsamples) && (xmsh.reserved2 <= 16))
{
dwMemPos += (((UINT)xmsh.reserved2) - nsamples) * xmsh.shsize;
}
#endif
for (UINT ismpd=0; ismpd<nsamples; ismpd++)
{
if ((samplemap[ismpd]) && (samplesize[ismpd]) && (dwMemPos < dwMemLength))
{
ReadSample(&Ins[samplemap[ismpd]], flags[ismpd], (LPSTR)(lpStream + dwMemPos), dwMemLength - dwMemPos);
}
dwMemPos += samplesize[ismpd];
if (dwMemPos >= dwMemLength) break;
}
}
// Read song comments: "TEXT"
if ((dwMemPos + 8 < dwMemLength) && (bswapLE32(*((DWORD *)(lpStream+dwMemPos))) == 0x74786574))
{
UINT len = *((DWORD *)(lpStream+dwMemPos+4));
dwMemPos += 8;
if ((dwMemPos + len <= dwMemLength) && (len < 16384))
{
m_lpszSongComments = new char[len+1];
if (m_lpszSongComments)
{
memcpy(m_lpszSongComments, lpStream+dwMemPos, len);
m_lpszSongComments[len] = 0;
}
dwMemPos += len;
}
}
// Read midi config: "MIDI"
if ((dwMemPos + 8 < dwMemLength) && (bswapLE32(*((DWORD *)(lpStream+dwMemPos))) == 0x4944494D))
{
UINT len = *((DWORD *)(lpStream+dwMemPos+4));
dwMemPos += 8;
if (len == sizeof(MODMIDICFG))
{
memcpy(&m_MidiCfg, lpStream+dwMemPos, len);
m_dwSongFlags |= SONG_EMBEDMIDICFG;
}
}
// Read pattern names: "PNAM"
if ((dwMemPos + 8 < dwMemLength) && (bswapLE32(*((DWORD *)(lpStream+dwMemPos))) == 0x4d414e50))
{
UINT len = *((DWORD *)(lpStream+dwMemPos+4));
dwMemPos += 8;
if ((dwMemPos + len <= dwMemLength) && (len <= MAX_PATTERNS*MAX_PATTERNNAME) && (len >= MAX_PATTERNNAME))
{
m_lpszPatternNames = new char[len];
if (m_lpszPatternNames)
{
m_nPatternNames = len / MAX_PATTERNNAME;
memcpy(m_lpszPatternNames, lpStream+dwMemPos, len);
}
dwMemPos += len;
}
}
// Read channel names: "CNAM"
if ((dwMemPos + 8 < dwMemLength) && (bswapLE32(*((DWORD *)(lpStream+dwMemPos))) == 0x4d414e43))
{
UINT len = *((DWORD *)(lpStream+dwMemPos+4));
dwMemPos += 8;
if ((dwMemPos + len <= dwMemLength) && (len <= MAX_BASECHANNELS*MAX_CHANNELNAME))
{
UINT n = len / MAX_CHANNELNAME;
for (UINT i=0; i<n; i++)
{
memcpy(ChnSettings[i].szName, (lpStream+dwMemPos+i*MAX_CHANNELNAME), MAX_CHANNELNAME);
ChnSettings[i].szName[MAX_CHANNELNAME-1] = 0;
}
dwMemPos += len;
}
}
// Read mix plugins information
if (dwMemPos + 8 < dwMemLength)
{
dwMemPos += LoadMixPlugins(lpStream+dwMemPos, dwMemLength-dwMemPos);
}
return TRUE;
}
#ifndef MODPLUG_NO_FILESAVE
BOOL CSoundFile::SaveXM(LPCSTR lpszFileName, UINT nPacking)
//---------------------------------------------------------
{
BYTE s[64*64*5];
XMFILEHEADER header;
XMINSTRUMENTHEADER xmih;
XMSAMPLEHEADER xmsh;
XMSAMPLESTRUCT xmss;
BYTE smptable[32];
BYTE xmph[9];
FILE *f;
int i;
if ((!m_nChannels) || (!lpszFileName)) return FALSE;
if ((f = fopen(lpszFileName, "wb")) == NULL) return FALSE;
fwrite("Extended Module: ", 17, 1, f);
fwrite(m_szNames[0], 20, 1, f);
s[0] = 0x1A;
lstrcpy((LPSTR)&s[1], (nPacking) ? "MOD Plugin packed " : "FastTracker v2.00 ");
s[21] = 0x04;
s[22] = 0x01;
fwrite(s, 23, 1, f);
// Writing song header
memset(&header, 0, sizeof(header));
header.size = sizeof(XMFILEHEADER);
header.norder = 0;
header.restartpos = m_nRestartPos;
header.channels = m_nChannels;
header.patterns = 0;
for (i=0; i<MAX_ORDERS; i++)
{
if (Order[i] == 0xFF) break;
header.norder++;
if ((Order[i] >= header.patterns) && (Order[i] < MAX_PATTERNS)) header.patterns = Order[i]+1;
}
header.instruments = m_nInstruments;
if (!header.instruments) header.instruments = m_nSamples;
header.flags = (m_dwSongFlags & SONG_LINEARSLIDES) ? 0x01 : 0x00;
if (m_dwSongFlags & SONG_EXFILTERRANGE) header.flags |= 0x1000;
header.tempo = m_nDefaultTempo;
header.speed = m_nDefaultSpeed;
memcpy(header.order, Order, header.norder);
fwrite(&header, 1, sizeof(header), f);
// Writing patterns
for (i=0; i<header.patterns; i++) if (Patterns[i])
{
MODCOMMAND *p = Patterns[i];
UINT len = 0;
memset(&xmph, 0, sizeof(xmph));
xmph[0] = 9;
xmph[5] = (BYTE)(PatternSize[i] & 0xFF);
xmph[6] = (BYTE)(PatternSize[i] >> 8);
for (UINT j=m_nChannels*PatternSize[i]; j; j--,p++)
{
UINT note = p->note;
UINT param = ModSaveCommand(p, TRUE);
UINT command = param >> 8;
param &= 0xFF;
if (note >= 0xFE) note = 97; else
if ((note <= 12) || (note > 96+12)) note = 0; else
note -= 12;
UINT vol = 0;
if (p->volcmd)
{
UINT volcmd = p->volcmd;
switch(volcmd)
{
case VOLCMD_VOLUME: vol = 0x10 + p->vol; break;
case VOLCMD_VOLSLIDEDOWN: vol = 0x60 + (p->vol & 0x0F); break;
case VOLCMD_VOLSLIDEUP: vol = 0x70 + (p->vol & 0x0F); break;
case VOLCMD_FINEVOLDOWN: vol = 0x80 + (p->vol & 0x0F); break;
case VOLCMD_FINEVOLUP: vol = 0x90 + (p->vol & 0x0F); break;
case VOLCMD_VIBRATOSPEED: vol = 0xA0 + (p->vol & 0x0F); break;
case VOLCMD_VIBRATO: vol = 0xB0 + (p->vol & 0x0F); break;
case VOLCMD_PANNING: vol = 0xC0 + (p->vol >> 2); if (vol > 0xCF) vol = 0xCF; break;
case VOLCMD_PANSLIDELEFT: vol = 0xD0 + (p->vol & 0x0F); break;
case VOLCMD_PANSLIDERIGHT: vol = 0xE0 + (p->vol & 0x0F); break;
case VOLCMD_TONEPORTAMENTO: vol = 0xF0 + (p->vol & 0x0F); break;
}
}
if ((note) && (p->instr) && (vol > 0x0F) && (command) && (param))
{
s[len++] = note;
s[len++] = p->instr;
s[len++] = vol;
s[len++] = command;
s[len++] = param;
} else
{
BYTE b = 0x80;
if (note) b |= 0x01;
if (p->instr) b |= 0x02;
if (vol >= 0x10) b |= 0x04;
if (command) b |= 0x08;
if (param) b |= 0x10;
s[len++] = b;
if (b & 1) s[len++] = note;
if (b & 2) s[len++] = p->instr;
if (b & 4) s[len++] = vol;
if (b & 8) s[len++] = command;
if (b & 16) s[len++] = param;
}
if (len > sizeof(s) - 5) break;
}
xmph[7] = (BYTE)(len & 0xFF);
xmph[8] = (BYTE)(len >> 8);
fwrite(xmph, 1, 9, f);
fwrite(s, 1, len, f);
} else
{
memset(&xmph, 0, sizeof(xmph));
xmph[0] = 9;
xmph[5] = (BYTE)(PatternSize[i] & 0xFF);
xmph[6] = (BYTE)(PatternSize[i] >> 8);
fwrite(xmph, 1, 9, f);
}
// Writing instruments
for (i=1; i<=header.instruments; i++)
{
MODINSTRUMENT *pins;
BYTE flags[32];
memset(&xmih, 0, sizeof(xmih));
memset(&xmsh, 0, sizeof(xmsh));
xmih.size = sizeof(xmih) + sizeof(xmsh);
memcpy(xmih.name, m_szNames[i], 22);
xmih.type = 0;
xmih.samples = 0;
if (m_nInstruments)
{
INSTRUMENTHEADER *penv = Headers[i];
if (penv)
{
memcpy(xmih.name, penv->name, 22);
xmih.type = penv->nMidiProgram;
xmsh.volfade = penv->nFadeOut;
xmsh.vnum = (BYTE)penv->nVolEnv;
xmsh.pnum = (BYTE)penv->nPanEnv;
if (xmsh.vnum > 12) xmsh.vnum = 12;
if (xmsh.pnum > 12) xmsh.pnum = 12;
for (UINT ienv=0; ienv<12; ienv++)
{
xmsh.venv[ienv*2] = penv->VolPoints[ienv];
xmsh.venv[ienv*2+1] = penv->VolEnv[ienv];
xmsh.penv[ienv*2] = penv->PanPoints[ienv];
xmsh.penv[ienv*2+1] = penv->PanEnv[ienv];
}
if (penv->dwFlags & ENV_VOLUME) xmsh.vtype |= 1;
if (penv->dwFlags & ENV_VOLSUSTAIN) xmsh.vtype |= 2;
if (penv->dwFlags & ENV_VOLLOOP) xmsh.vtype |= 4;
if (penv->dwFlags & ENV_PANNING) xmsh.ptype |= 1;
if (penv->dwFlags & ENV_PANSUSTAIN) xmsh.ptype |= 2;
if (penv->dwFlags & ENV_PANLOOP) xmsh.ptype |= 4;
xmsh.vsustain = (BYTE)penv->nVolSustainBegin;
xmsh.vloops = (BYTE)penv->nVolLoopStart;
xmsh.vloope = (BYTE)penv->nVolLoopEnd;
xmsh.psustain = (BYTE)penv->nPanSustainBegin;
xmsh.ploops = (BYTE)penv->nPanLoopStart;
xmsh.ploope = (BYTE)penv->nPanLoopEnd;
for (UINT j=0; j<96; j++) if (penv->Keyboard[j+12])
{
UINT k;
for (k=0; k<xmih.samples; k++) if (smptable[k] == penv->Keyboard[j+12]) break;
if (k == xmih.samples)
{
smptable[xmih.samples++] = penv->Keyboard[j+12];
}
if (xmih.samples >= 32) break;
xmsh.snum[j] = k;
}
// xmsh.reserved2 = xmih.samples;
}
} else
{
xmih.samples = 1;
// xmsh.reserved2 = 1;
smptable[0] = i;
}
xmsh.shsize = (xmih.samples) ? 40 : 0;
fwrite(&xmih, 1, sizeof(xmih), f);
if (smptable[0])
{
MODINSTRUMENT *pvib = &Ins[smptable[0]];
xmsh.vibtype = pvib->nVibType;
xmsh.vibsweep = pvib->nVibSweep;
xmsh.vibdepth = pvib->nVibDepth;
xmsh.vibrate = pvib->nVibRate;
}
fwrite(&xmsh, 1, xmih.size - sizeof(xmih), f);
if (!xmih.samples) continue;
for (UINT ins=0; ins<xmih.samples; ins++)
{
memset(&xmss, 0, sizeof(xmss));
if (smptable[ins]) memcpy(xmss.name, m_szNames[smptable[ins]], 22);
pins = &Ins[smptable[ins]];
xmss.samplen = pins->nLength;
xmss.loopstart = pins->nLoopStart;
xmss.looplen = pins->nLoopEnd - pins->nLoopStart;
xmss.vol = pins->nVolume / 4;
xmss.finetune = (char)pins->nFineTune;
xmss.type = 0;
if (pins->uFlags & CHN_LOOP) xmss.type = (pins->uFlags & CHN_PINGPONGLOOP) ? 2 : 1;
flags[ins] = RS_PCM8D;
#ifndef NO_PACKING
if (nPacking)
{
if ((!(pins->uFlags & (CHN_16BIT|CHN_STEREO)))
&& (CanPackSample((char *)pins->pSample, pins->nLength, nPacking)))
{
flags[ins] = RS_ADPCM4;
xmss.res = 0xAD;
}
} else
#endif
{
if (pins->uFlags & CHN_16BIT)
{
flags[ins] = RS_PCM16D;
xmss.type |= 0x10;
xmss.looplen *= 2;
xmss.loopstart *= 2;
xmss.samplen *= 2;
}
if (pins->uFlags & CHN_STEREO)
{
flags[ins] = (pins->uFlags & CHN_16BIT) ? RS_STPCM16D : RS_STPCM8D;
xmss.type |= 0x20;
xmss.looplen *= 2;
xmss.loopstart *= 2;
xmss.samplen *= 2;
}
}
xmss.pan = 255;
if (pins->nPan < 256) xmss.pan = (BYTE)pins->nPan;
xmss.relnote = (signed char)pins->RelativeTone;
fwrite(&xmss, 1, xmsh.shsize, f);
}
for (UINT ismpd=0; ismpd<xmih.samples; ismpd++)
{
pins = &Ins[smptable[ismpd]];
if (pins->pSample)
{
#ifndef NO_PACKING
if ((flags[ismpd] == RS_ADPCM4) && (xmih.samples>1)) CanPackSample((char *)pins->pSample, pins->nLength, nPacking);
#endif // NO_PACKING
WriteSample(f, pins, flags[ismpd]);
}
}
}
// Writing song comments
if ((m_lpszSongComments) && (m_lpszSongComments[0]))
{
DWORD d = 0x74786574;
fwrite(&d, 1, 4, f);
d = strlen(m_lpszSongComments);
fwrite(&d, 1, 4, f);
fwrite(m_lpszSongComments, 1, d, f);
}
// Writing midi cfg
if (m_dwSongFlags & SONG_EMBEDMIDICFG)
{
DWORD d = 0x4944494D;
fwrite(&d, 1, 4, f);
d = sizeof(MODMIDICFG);
fwrite(&d, 1, 4, f);
fwrite(&m_MidiCfg, 1, sizeof(MODMIDICFG), f);
}
// Writing Pattern Names
if ((m_nPatternNames) && (m_lpszPatternNames))
{
DWORD dwLen = m_nPatternNames * MAX_PATTERNNAME;
while ((dwLen >= MAX_PATTERNNAME) && (!m_lpszPatternNames[dwLen-MAX_PATTERNNAME])) dwLen -= MAX_PATTERNNAME;
if (dwLen >= MAX_PATTERNNAME)
{
DWORD d = 0x4d414e50;
fwrite(&d, 1, 4, f);
fwrite(&dwLen, 1, 4, f);
fwrite(m_lpszPatternNames, 1, dwLen, f);
}
}
// Writing Channel Names
{
UINT nChnNames = 0;
for (UINT inam=0; inam<m_nChannels; inam++)
{
if (ChnSettings[inam].szName[0]) nChnNames = inam+1;
}
// Do it!
if (nChnNames)
{
DWORD dwLen = nChnNames * MAX_CHANNELNAME;
DWORD d = 0x4d414e43;
fwrite(&d, 1, 4, f);
fwrite(&dwLen, 1, 4, f);
for (UINT inam=0; inam<nChnNames; inam++)
{
fwrite(ChnSettings[inam].szName, 1, MAX_CHANNELNAME, f);
}
}
}
// Save mix plugins information
SaveMixPlugins(f);
fclose(f);
return TRUE;
}
#endif // MODPLUG_NO_FILESAVE

@ -0,0 +1,471 @@
/*
* This source code is public domain.
*
* Handles unpacking of Powerpack PP20
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
BOOL PP20_Unpack(LPCBYTE *ppMemFile, LPDWORD pdwMemLength);
#pragma pack(1)
typedef struct MMCMPFILEHEADER
{
char id[8]; // "ziRCONia"
WORD hdrsize;
} MMCMPFILEHEADER, *LPMMCMPFILEHEADER;
typedef struct MMCMPHEADER
{
WORD version;
WORD nblocks;
DWORD filesize;
DWORD blktable;
BYTE glb_comp;
BYTE fmt_comp;
} MMCMPHEADER, *LPMMCMPHEADER;
typedef struct MMCMPBLOCK
{
DWORD unpk_size;
DWORD pk_size;
DWORD xor_chk;
WORD sub_blk;
WORD flags;
WORD tt_entries;
USHORT num_bits;
} MMCMPBLOCK, *LPMMCMPBLOCK;
typedef struct MMCMPSUBBLOCK
{
DWORD unpk_pos;
DWORD unpk_size;
} MMCMPSUBBLOCK, *LPMMCMPSUBBLOCK;
#pragma pack()
// make sure of structure sizes
typedef int chk_MMCMPFILEHEADER[(sizeof(struct MMCMPFILEHEADER) == 10) * 2 - 1];
typedef int chk_MMCMPHEADER[(sizeof(struct MMCMPHEADER) == 14) * 2 - 1];
typedef int chk_MMCMPBLOCK[(sizeof(struct MMCMPBLOCK) == 20) * 2 - 1];
typedef int chk_MMCMPSUBBLOCK[(sizeof(struct MMCMPSUBBLOCK) == 8) * 2 - 1];
#define MMCMP_COMP 0x0001
#define MMCMP_DELTA 0x0002
#define MMCMP_16BIT 0x0004
#define MMCMP_STEREO 0x0100
#define MMCMP_ABS16 0x0200
#define MMCMP_ENDIAN 0x0400
typedef struct MMCMPBITBUFFER
{
UINT bitcount;
DWORD bitbuffer;
LPCBYTE pSrc;
LPCBYTE pEnd;
DWORD GetBits(UINT nBits);
} MMCMPBITBUFFER;
DWORD MMCMPBITBUFFER::GetBits(UINT nBits)
//---------------------------------------
{
DWORD d;
if (!nBits) return 0;
while (bitcount < 24)
{
bitbuffer |= ((pSrc < pEnd) ? *pSrc++ : 0) << bitcount;
bitcount += 8;
}
d = bitbuffer & ((1 << nBits) - 1);
bitbuffer >>= nBits;
bitcount -= nBits;
return d;
}
//#define MMCMP_LOG
#ifdef MMCMP_LOG
extern void Log(LPCSTR s, ...);
#endif
static const DWORD MMCMP8BitCommands[8] =
{
0x01, 0x03, 0x07, 0x0F, 0x1E, 0x3C, 0x78, 0xF8
};
static const UINT MMCMP8BitFetch[8] =
{
3, 3, 3, 3, 2, 1, 0, 0
};
static const DWORD MMCMP16BitCommands[16] =
{
0x01, 0x03, 0x07, 0x0F, 0x1E, 0x3C, 0x78, 0xF0,
0x1F0, 0x3F0, 0x7F0, 0xFF0, 0x1FF0, 0x3FF0, 0x7FF0, 0xFFF0
};
static const UINT MMCMP16BitFetch[16] =
{
4, 4, 4, 4, 3, 2, 1, 0,
0, 0, 0, 0, 0, 0, 0, 0
};
static void swap_mfh(LPMMCMPFILEHEADER fh)
{
fh->hdrsize = bswapLE16(fh->hdrsize);
}
static void swap_mmh(LPMMCMPHEADER mh)
{
mh->version = bswapLE16(mh->version);
mh->nblocks = bswapLE16(mh->nblocks);
mh->filesize = bswapLE32(mh->filesize);
mh->blktable = bswapLE32(mh->blktable);
}
static void swap_block (LPMMCMPBLOCK blk)
{
blk->unpk_size = bswapLE32(blk->unpk_size);
blk->pk_size = bswapLE32(blk->pk_size);
blk->xor_chk = bswapLE32(blk->xor_chk);
blk->sub_blk = bswapLE16(blk->sub_blk);
blk->flags = bswapLE16(blk->flags);
blk->tt_entries = bswapLE16(blk->tt_entries);
blk->num_bits = bswapLE16(blk->num_bits);
}
static void swap_subblock (LPMMCMPSUBBLOCK sblk)
{
sblk->unpk_pos = bswapLE32(sblk->unpk_pos);
sblk->unpk_size = bswapLE32(sblk->unpk_size);
}
BOOL MMCMP_Unpack(LPCBYTE *ppMemFile, LPDWORD pdwMemLength)
//---------------------------------------------------------
{
DWORD dwMemLength;
LPCBYTE lpMemFile;
LPBYTE pBuffer;
LPMMCMPFILEHEADER pmfh;
LPMMCMPHEADER pmmh;
const DWORD *pblk_table;
DWORD dwFileSize;
BYTE tmp0[32], tmp1[32];
if (PP20_Unpack(ppMemFile, pdwMemLength))
{
return TRUE;
}
dwMemLength = *pdwMemLength;
lpMemFile = *ppMemFile;
if ((dwMemLength < 256) || (!lpMemFile)) return FALSE;
memcpy(tmp0, lpMemFile, 24);
pmfh = (LPMMCMPFILEHEADER)(tmp0);
pmmh = (LPMMCMPHEADER)(tmp0+10);
swap_mfh(pmfh);
swap_mmh(pmmh);
if ((memcmp(pmfh->id,"ziRCONia",8) != 0) || (pmfh->hdrsize < 14)
|| (!pmmh->nblocks) || (pmmh->filesize < 16) || (pmmh->filesize > 0x8000000)
|| (pmmh->blktable >= dwMemLength) || (pmmh->blktable + 4*pmmh->nblocks > dwMemLength)) return FALSE;
dwFileSize = pmmh->filesize;
if ((pBuffer = (LPBYTE)GlobalAllocPtr(GHND, (dwFileSize + 31) & ~15)) == NULL) return FALSE;
pblk_table = (const DWORD *)(lpMemFile+pmmh->blktable);
for (UINT nBlock=0; nBlock<pmmh->nblocks; nBlock++)
{
DWORD dwMemPos = bswapLE32(pblk_table[nBlock]);
DWORD dwSubPos;
LPMMCMPBLOCK pblk;
LPMMCMPSUBBLOCK psubblk;
if (dwMemPos + 20 >= dwMemLength) break;
memcpy(tmp1,lpMemFile+dwMemPos,28);
pblk = (LPMMCMPBLOCK)(tmp1);
psubblk = (LPMMCMPSUBBLOCK)(tmp1+20);
swap_block(pblk);
swap_subblock(psubblk);
if (dwMemPos + 20 + pblk->sub_blk*8 >= dwMemLength) break;
dwSubPos = dwMemPos + 20;
dwMemPos += 20 + pblk->sub_blk*8;
#ifdef MMCMP_LOG
Log("block %d: flags=%04X sub_blocks=%d", nBlock, (UINT)pblk->flags, (UINT)pblk->sub_blk);
Log(" pksize=%d unpksize=%d", pblk->pk_size, pblk->unpk_size);
Log(" tt_entries=%d num_bits=%d\n", pblk->tt_entries, pblk->num_bits);
#endif
// Data is not packed
if (!(pblk->flags & MMCMP_COMP))
{
for (UINT i=0; i<pblk->sub_blk; i++)
{
if ((psubblk->unpk_pos >= dwFileSize) ||
(psubblk->unpk_size >= dwFileSize) ||
(psubblk->unpk_size > dwFileSize - psubblk->unpk_pos)) break;
#ifdef MMCMP_LOG
Log(" Unpacked sub-block %d: offset %d, size=%d\n", i, psubblk->unpk_pos, psubblk->unpk_size);
#endif
memcpy(pBuffer+psubblk->unpk_pos, lpMemFile+dwMemPos, psubblk->unpk_size);
dwMemPos += psubblk->unpk_size;
memcpy(tmp1+20,lpMemFile+dwSubPos+i*8,8);
swap_subblock(psubblk);
}
} else
// Data is 16-bit packed
if (pblk->flags & MMCMP_16BIT && pblk->num_bits < 16)
{
MMCMPBITBUFFER bb;
LPWORD pDest = (LPWORD)(pBuffer + psubblk->unpk_pos);
DWORD dwSize = psubblk->unpk_size >> 1;
DWORD dwPos = 0;
UINT numbits = pblk->num_bits;
UINT subblk = 0, oldval = 0;
#ifdef MMCMP_LOG
Log(" 16-bit block: pos=%d size=%d ", psubblk->unpk_pos, psubblk->unpk_size);
if (pblk->flags & MMCMP_DELTA) Log("DELTA ");
if (pblk->flags & MMCMP_ABS16) Log("ABS16 ");
Log("\n");
#endif
bb.bitcount = 0;
bb.bitbuffer = 0;
bb.pSrc = lpMemFile+dwMemPos+pblk->tt_entries;
bb.pEnd = lpMemFile+dwMemPos+pblk->pk_size;
while (subblk < pblk->sub_blk)
{
UINT newval = 0x10000;
DWORD d = bb.GetBits(numbits+1);
if (d >= MMCMP16BitCommands[numbits])
{
UINT nFetch = MMCMP16BitFetch[numbits];
UINT newbits = bb.GetBits(nFetch) + ((d - MMCMP16BitCommands[numbits]) << nFetch);
if (newbits != numbits)
{
numbits = newbits & 0x0F;
} else
{
if ((d = bb.GetBits(4)) == 0x0F)
{
if (bb.GetBits(1)) break;
newval = 0xFFFF;
} else
{
newval = 0xFFF0 + d;
}
}
} else
{
newval = d;
}
if (newval < 0x10000)
{
newval = (newval & 1) ? (UINT)(-(LONG)((newval+1) >> 1)) : (UINT)(newval >> 1);
if (pblk->flags & MMCMP_DELTA)
{
newval += oldval;
oldval = newval;
} else
if (!(pblk->flags & MMCMP_ABS16))
{
newval ^= 0x8000;
}
pDest[dwPos++] = (WORD)newval;
}
if (dwPos >= dwSize)
{
subblk++;
memcpy(tmp1+20,lpMemFile+dwSubPos+subblk*8,8);
swap_subblock(psubblk);
dwPos = 0;
dwSize = psubblk->unpk_size >> 1;
pDest = (LPWORD)(pBuffer + psubblk->unpk_pos);
}
}
} else if (pblk->num_bits < 8)
// Data is 8-bit packed
{
MMCMPBITBUFFER bb;
LPBYTE pDest = pBuffer + psubblk->unpk_pos;
DWORD dwSize = psubblk->unpk_size;
DWORD dwPos = 0;
UINT numbits = pblk->num_bits;
UINT subblk = 0, oldval = 0;
LPCBYTE ptable = lpMemFile+dwMemPos;
bb.bitcount = 0;
bb.bitbuffer = 0;
bb.pSrc = lpMemFile+dwMemPos+pblk->tt_entries;
bb.pEnd = lpMemFile+dwMemPos+pblk->pk_size;
while (subblk < pblk->sub_blk)
{
UINT newval = 0x100;
DWORD d = bb.GetBits(numbits+1);
if (d >= MMCMP8BitCommands[numbits])
{
UINT nFetch = MMCMP8BitFetch[numbits];
UINT newbits = bb.GetBits(nFetch) + ((d - MMCMP8BitCommands[numbits]) << nFetch);
if (newbits != numbits)
{
numbits = newbits & 0x07;
} else
{
if ((d = bb.GetBits(3)) == 7)
{
if (bb.GetBits(1)) break;
newval = 0xFF;
} else
{
newval = 0xF8 + d;
}
}
} else
{
newval = d;
}
if (newval < 0x100)
{
int n = ptable[newval];
if (pblk->flags & MMCMP_DELTA)
{
n += oldval;
oldval = n;
}
pDest[dwPos++] = (BYTE)n;
}
if (dwPos >= dwSize)
{
subblk++;
memcpy(tmp1+20,lpMemFile+dwSubPos+subblk*8,8);
swap_subblock(psubblk);
dwPos = 0;
dwSize = psubblk->unpk_size;
pDest = pBuffer + psubblk->unpk_pos;
}
}
} else
{
return FALSE;
}
}
*ppMemFile = pBuffer;
*pdwMemLength = dwFileSize;
return TRUE;
}
//////////////////////////////////////////////////////////////////////////////
//
// PowerPack PP20 Unpacker
//
typedef struct _PPBITBUFFER
{
UINT bitcount;
ULONG bitbuffer;
LPCBYTE pStart;
LPCBYTE pSrc;
ULONG GetBits(UINT n);
} PPBITBUFFER;
ULONG PPBITBUFFER::GetBits(UINT n)
{
ULONG result = 0;
for (UINT i=0; i<n; i++)
{
if (!bitcount)
{
bitcount = 8;
if (pSrc != pStart) pSrc--;
bitbuffer = *pSrc;
}
result = (result<<1) | (bitbuffer&1);
bitbuffer >>= 1;
bitcount--;
}
return result;
}
static VOID PP20_DoUnpack(const BYTE *pSrc, UINT nSrcLen, BYTE *pDst, UINT nDstLen)
{
PPBITBUFFER BitBuffer;
ULONG nBytesLeft;
BitBuffer.pStart = pSrc;
BitBuffer.pSrc = pSrc + nSrcLen - 4;
BitBuffer.bitbuffer = 0;
BitBuffer.bitcount = 0;
BitBuffer.GetBits(pSrc[nSrcLen-1]);
nBytesLeft = nDstLen;
while (nBytesLeft > 0)
{
if (!BitBuffer.GetBits(1))
{
UINT n = 1;
while (n < nBytesLeft)
{
UINT code = BitBuffer.GetBits(2);
n += code;
if (code != 3) break;
}
for (UINT i=0; i<n; i++)
{
pDst[--nBytesLeft] = (BYTE)BitBuffer.GetBits(8);
}
if (!nBytesLeft) break;
}
{
UINT n = BitBuffer.GetBits(2)+1;
UINT nbits = pSrc[n-1];
UINT nofs;
if (n==4)
{
nofs = BitBuffer.GetBits( (BitBuffer.GetBits(1)) ? nbits : 7 );
while (n < nBytesLeft)
{
UINT code = BitBuffer.GetBits(3);
n += code;
if (code != 7) break;
}
} else
{
nofs = BitBuffer.GetBits(nbits);
}
for (UINT i=0; i<=n; i++)
{
pDst[nBytesLeft-1] = (nBytesLeft+nofs < nDstLen) ? pDst[nBytesLeft+nofs] : 0;
if (!--nBytesLeft) break;
}
}
}
}
BOOL PP20_Unpack(LPCBYTE *ppMemFile, LPDWORD pdwMemLength)
{
DWORD dwMemLength = *pdwMemLength;
LPCBYTE lpMemFile = *ppMemFile;
DWORD dwDstLen;
LPBYTE pBuffer;
if ((!lpMemFile) || (dwMemLength < 256) || (memcmp(lpMemFile,"PP20",4) != 0)) return FALSE;
dwDstLen = (lpMemFile[dwMemLength-4]<<16) | (lpMemFile[dwMemLength-3]<<8) | (lpMemFile[dwMemLength-2]);
//Log("PP20 detected: Packed length=%d, Unpacked length=%d\n", dwMemLength, dwDstLen);
if ((dwDstLen < 512) || (dwDstLen > 0x400000) || (dwDstLen > 16*dwMemLength)) return FALSE;
if ((pBuffer = (LPBYTE)GlobalAllocPtr(GHND, (dwDstLen + 31) & ~15)) == NULL) return FALSE;
PP20_DoUnpack(lpMemFile+4, dwMemLength-4, pBuffer, dwDstLen);
*ppMemFile = pBuffer;
*pdwMemLength = dwDstLen;
return TRUE;
}

@ -0,0 +1,277 @@
/*
* This source code is public domain.
*
* Authors: Kenton Varda <temporal@gauge3d.org> (C interface wrapper)
*/
#include "stdafx.h"
#include "modplug.h"
#include "sndfile.h"
struct _ModPlugFile
{
CSoundFile mSoundFile;
};
namespace ModPlug
{
ModPlug_Settings gSettings =
{
MODPLUG_ENABLE_OVERSAMPLING | MODPLUG_ENABLE_NOISE_REDUCTION,
2, // mChannels
16, // mBits
44100, // mFrequency
MODPLUG_RESAMPLE_LINEAR, //mResamplingMode
128, // mStereoSeparation
32, // mMaxMixChannels
0,
0,
0,
0,
0,
0,
0
};
int gSampleSize;
void UpdateSettings(bool updateBasicConfig)
{
if(gSettings.mFlags & MODPLUG_ENABLE_REVERB)
{
CSoundFile::SetReverbParameters(gSettings.mReverbDepth,
gSettings.mReverbDelay);
}
if(gSettings.mFlags & MODPLUG_ENABLE_MEGABASS)
{
CSoundFile::SetXBassParameters(gSettings.mBassAmount,
gSettings.mBassRange);
}
else // modplug seems to ignore the SetWaveConfigEx() setting for bass boost
CSoundFile::SetXBassParameters(0, 0);
if(gSettings.mFlags & MODPLUG_ENABLE_SURROUND)
{
CSoundFile::SetSurroundParameters(gSettings.mSurroundDepth,
gSettings.mSurroundDelay);
}
if(updateBasicConfig)
{
CSoundFile::SetWaveConfig(gSettings.mFrequency,
gSettings.mBits,
gSettings.mChannels);
CSoundFile::SetMixConfig(gSettings.mStereoSeparation,
gSettings.mMaxMixChannels);
gSampleSize = gSettings.mBits / 8 * gSettings.mChannels;
}
CSoundFile::SetWaveConfigEx(gSettings.mFlags & MODPLUG_ENABLE_SURROUND,
!(gSettings.mFlags & MODPLUG_ENABLE_OVERSAMPLING),
gSettings.mFlags & MODPLUG_ENABLE_REVERB,
true,
gSettings.mFlags & MODPLUG_ENABLE_MEGABASS,
gSettings.mFlags & MODPLUG_ENABLE_NOISE_REDUCTION,
false);
CSoundFile::SetResamplingMode(gSettings.mResamplingMode);
}
}
ModPlugFile* ModPlug_Load(const void* data, int size)
{
ModPlugFile* result = new ModPlugFile;
ModPlug::UpdateSettings(true);
if(result->mSoundFile.Create((const BYTE*)data, size))
{
result->mSoundFile.SetRepeatCount(ModPlug::gSettings.mLoopCount);
return result;
}
else
{
delete result;
return NULL;
}
}
void ModPlug_Unload(ModPlugFile* file)
{
file->mSoundFile.Destroy();
delete file;
}
int ModPlug_Read(ModPlugFile* file, void* buffer, int size)
{
return file->mSoundFile.Read(buffer, size) * ModPlug::gSampleSize;
}
const char* ModPlug_GetName(ModPlugFile* file)
{
return file->mSoundFile.GetTitle();
}
int ModPlug_GetLength(ModPlugFile* file)
{
return file->mSoundFile.GetSongTime() * 1000;
}
void ModPlug_InitMixerCallback(ModPlugFile* file,ModPlugMixerProc proc)
{
file->mSoundFile.gpSndMixHook = (LPSNDMIXHOOKPROC)proc ;
return;
}
void ModPlug_UnloadMixerCallback(ModPlugFile* file)
{
file->mSoundFile.gpSndMixHook = NULL;
return ;
}
unsigned int ModPlug_GetMasterVolume(ModPlugFile* file)
{
return (unsigned int)file->mSoundFile.m_nMasterVolume;
}
void ModPlug_SetMasterVolume(ModPlugFile* file,unsigned int cvol)
{
(void)file->mSoundFile.SetMasterVolume( (UINT)cvol,
FALSE );
return ;
}
int ModPlug_GetCurrentSpeed(ModPlugFile* file)
{
return file->mSoundFile.m_nMusicSpeed;
}
int ModPlug_GetCurrentTempo(ModPlugFile* file)
{
return file->mSoundFile.m_nMusicTempo;
}
int ModPlug_GetCurrentOrder(ModPlugFile* file)
{
return file->mSoundFile.GetCurrentOrder();
}
int ModPlug_GetCurrentPattern(ModPlugFile* file)
{
return file->mSoundFile.GetCurrentPattern();
}
int ModPlug_GetCurrentRow(ModPlugFile* file)
{
return file->mSoundFile.m_nRow;
}
int ModPlug_GetPlayingChannels(ModPlugFile* file)
{
return ( file->mSoundFile.m_nMixChannels < file->mSoundFile.m_nMaxMixChannels ? file->mSoundFile.m_nMixChannels : file->mSoundFile.m_nMaxMixChannels );
}
void ModPlug_SeekOrder(ModPlugFile* file,int order)
{
file->mSoundFile.SetCurrentOrder(order);
}
int ModPlug_GetModuleType(ModPlugFile* file)
{
return file->mSoundFile.m_nType;
}
char* ModPlug_GetMessage(ModPlugFile* file)
{
return file->mSoundFile.m_lpszSongComments;
}
#ifndef MODPLUG_NO_FILESAVE
char ModPlug_ExportS3M(ModPlugFile* file,const char* filepath)
{
return (char)file->mSoundFile.SaveS3M(filepath,0);
}
char ModPlug_ExportXM(ModPlugFile* file,const char* filepath)
{
return (char)file->mSoundFile.SaveXM(filepath,0);
}
char ModPlug_ExportMOD(ModPlugFile* file,const char* filepath)
{
return (char)file->mSoundFile.SaveMod(filepath,0);
}
char ModPlug_ExportIT(ModPlugFile* file,const char* filepath)
{
return (char)file->mSoundFile.SaveIT(filepath,0);
}
#endif // MODPLUG_NO_FILESAVE
unsigned int ModPlug_NumInstruments(ModPlugFile* file)
{
return file->mSoundFile.m_nInstruments;
}
unsigned int ModPlug_NumSamples(ModPlugFile* file)
{
return file->mSoundFile.m_nSamples;
}
unsigned int ModPlug_NumPatterns(ModPlugFile* file)
{
return file->mSoundFile.GetNumPatterns();
}
unsigned int ModPlug_NumChannels(ModPlugFile* file)
{
return file->mSoundFile.GetNumChannels();
}
unsigned int ModPlug_SampleName(ModPlugFile* file,unsigned int qual,char* buff)
{
return file->mSoundFile.GetSampleName(qual,buff);
}
unsigned int ModPlug_InstrumentName(ModPlugFile* file,unsigned int qual,char* buff)
{
return file->mSoundFile.GetInstrumentName(qual,buff);
}
ModPlugNote* ModPlug_GetPattern(ModPlugFile* file,int pattern,unsigned int* numrows) {
if ( pattern<MAX_PATTERNS && pattern >= 0) {
if (file->mSoundFile.Patterns[pattern]) {
if (numrows) *numrows=(unsigned int)file->mSoundFile.PatternSize[pattern];
return (ModPlugNote*)file->mSoundFile.Patterns[pattern];
}
}
return NULL;
}
void ModPlug_Seek(ModPlugFile* file, int millisecond)
{
int maxpos;
int maxtime = file->mSoundFile.GetSongTime() * 1000;
float postime;
if(millisecond > maxtime)
millisecond = maxtime;
maxpos = file->mSoundFile.GetMaxPosition();
postime = 0.0f;
if (maxtime != 0)
postime = (float)maxpos / (float)maxtime;
file->mSoundFile.SetCurrentPos((int)(millisecond * postime));
}
void ModPlug_GetSettings(ModPlug_Settings* settings)
{
memcpy(settings, &ModPlug::gSettings, sizeof(ModPlug_Settings));
}
void ModPlug_SetSettings(const ModPlug_Settings* settings)
{
memcpy(&ModPlug::gSettings, settings, sizeof(ModPlug_Settings));
ModPlug::UpdateSettings(false); // do not update basic config.
}

@ -0,0 +1,185 @@
/*
* This source code is public domain.
*
* Authors: Kenton Varda <temporal@gauge3d.org> (C interface wrapper)
*/
#ifndef MODPLUG_H__INCLUDED
#define MODPLUG_H__INCLUDED
#ifdef __cplusplus
extern "C" {
#endif
#if defined(_WIN32) || defined(__CYGWIN__)
# if defined(MODPLUG_BUILD) && defined(DLL_EXPORT) /* building libmodplug as a dll for windows */
# define MODPLUG_EXPORT __declspec(dllexport)
# elif defined(MODPLUG_BUILD) || defined(MODPLUG_STATIC) /* building or using static libmodplug for windows */
# define MODPLUG_EXPORT
# else
# define MODPLUG_EXPORT __declspec(dllimport) /* using libmodplug dll for windows */
# endif
#elif defined(MODPLUG_BUILD) && defined(SYM_VISIBILITY)
# define MODPLUG_EXPORT __attribute__((visibility("default")))
#else
#define MODPLUG_EXPORT
#endif
struct _ModPlugFile;
typedef struct _ModPlugFile ModPlugFile;
struct _ModPlugNote {
unsigned char Note;
unsigned char Instrument;
unsigned char VolumeEffect;
unsigned char Effect;
unsigned char Volume;
unsigned char Parameter;
};
typedef struct _ModPlugNote ModPlugNote;
typedef void (*ModPlugMixerProc)(int*, unsigned long, unsigned long);
/* Load a mod file. [data] should point to a block of memory containing the complete
* file, and [size] should be the size of that block.
* Return the loaded mod file on success, or NULL on failure. */
MODPLUG_EXPORT ModPlugFile* ModPlug_Load(const void* data, int size);
/* Unload a mod file. */
MODPLUG_EXPORT void ModPlug_Unload(ModPlugFile* file);
/* Read sample data into the buffer. Returns the number of bytes read. If the end
* of the mod has been reached, zero is returned. */
MODPLUG_EXPORT int ModPlug_Read(ModPlugFile* file, void* buffer, int size);
/* Get the name of the mod. The returned buffer is stored within the ModPlugFile
* structure and will remain valid until you unload the file. */
MODPLUG_EXPORT const char* ModPlug_GetName(ModPlugFile* file);
/* Get the length of the mod, in milliseconds. Note that this result is not always
* accurate, especially in the case of mods with loops. */
MODPLUG_EXPORT int ModPlug_GetLength(ModPlugFile* file);
/* Seek to a particular position in the song. Note that seeking and MODs don't mix very
* well. Some mods will be missing instruments for a short time after a seek, as ModPlug
* does not scan the sequence backwards to find out which instruments were supposed to be
* playing at that time. (Doing so would be difficult and not very reliable.) Also,
* note that seeking is not very exact in some mods -- especially those for which
* ModPlug_GetLength() does not report the full length. */
MODPLUG_EXPORT void ModPlug_Seek(ModPlugFile* file, int millisecond);
enum _ModPlug_Flags
{
MODPLUG_ENABLE_OVERSAMPLING = 1 << 0, /* Enable oversampling (*highly* recommended) */
MODPLUG_ENABLE_NOISE_REDUCTION = 1 << 1, /* Enable noise reduction */
MODPLUG_ENABLE_REVERB = 1 << 2, /* Enable reverb */
MODPLUG_ENABLE_MEGABASS = 1 << 3, /* Enable megabass */
MODPLUG_ENABLE_SURROUND = 1 << 4 /* Enable surround sound. */
};
enum _ModPlug_ResamplingMode
{
MODPLUG_RESAMPLE_NEAREST = 0, /* No interpolation (very fast, extremely bad sound quality) */
MODPLUG_RESAMPLE_LINEAR = 1, /* Linear interpolation (fast, good quality) */
MODPLUG_RESAMPLE_SPLINE = 2, /* Cubic spline interpolation (high quality) */
MODPLUG_RESAMPLE_FIR = 3 /* 8-tap fir filter (extremely high quality) */
};
typedef struct _ModPlug_Settings
{
int mFlags; /* One or more of the MODPLUG_ENABLE_* flags above, bitwise-OR'ed */
/* Note that ModPlug always decodes sound at 44100kHz, 32 bit, stereo and then
* down-mixes to the settings you choose. */
int mChannels; /* Number of channels - 1 for mono or 2 for stereo */
int mBits; /* Bits per sample - 8, 16, or 32 */
int mFrequency; /* Sampling rate - 11025, 22050, or 44100 */
int mResamplingMode; /* One of MODPLUG_RESAMPLE_*, above */
int mStereoSeparation; /* Stereo separation, 1 - 256 */
int mMaxMixChannels; /* Maximum number of mixing channels (polyphony), 32 - 256 */
int mReverbDepth; /* Reverb level 0(quiet)-100(loud) */
int mReverbDelay; /* Reverb delay in ms, usually 40-200ms */
int mBassAmount; /* XBass level 0(quiet)-100(loud) */
int mBassRange; /* XBass cutoff in Hz 10-100 */
int mSurroundDepth; /* Surround level 0(quiet)-100(heavy) */
int mSurroundDelay; /* Surround delay in ms, usually 5-40ms */
int mLoopCount; /* Number of times to loop. Zero prevents looping.
* -1 loops forever. */
} ModPlug_Settings;
/* Get and set the mod decoder settings. All options, except for channels, bits-per-sample,
* sampling rate, and loop count, will take effect immediately. Those options which don't
* take effect immediately will take effect the next time you load a mod. */
MODPLUG_EXPORT void ModPlug_GetSettings(ModPlug_Settings* settings);
MODPLUG_EXPORT void ModPlug_SetSettings(const ModPlug_Settings* settings);
/* New ModPlug API Functions */
/* NOTE: Master Volume (1-512) */
MODPLUG_EXPORT unsigned int ModPlug_GetMasterVolume(ModPlugFile* file) ;
MODPLUG_EXPORT void ModPlug_SetMasterVolume(ModPlugFile* file,unsigned int cvol) ;
MODPLUG_EXPORT int ModPlug_GetCurrentSpeed(ModPlugFile* file);
MODPLUG_EXPORT int ModPlug_GetCurrentTempo(ModPlugFile* file);
MODPLUG_EXPORT int ModPlug_GetCurrentOrder(ModPlugFile* file);
MODPLUG_EXPORT int ModPlug_GetCurrentPattern(ModPlugFile* file);
MODPLUG_EXPORT int ModPlug_GetCurrentRow(ModPlugFile* file);
MODPLUG_EXPORT int ModPlug_GetPlayingChannels(ModPlugFile* file);
MODPLUG_EXPORT void ModPlug_SeekOrder(ModPlugFile* file,int order);
MODPLUG_EXPORT int ModPlug_GetModuleType(ModPlugFile* file);
MODPLUG_EXPORT char* ModPlug_GetMessage(ModPlugFile* file);
#define MODPLUG_NO_FILESAVE /* experimental yet. must match stdafx.h. */
#ifndef MODPLUG_NO_FILESAVE
/*
* EXPERIMENTAL Export Functions
*/
/*Export to a Scream Tracker 3 S3M module. EXPERIMENTAL (only works on Little-Endian platforms)*/
MODPLUG_EXPORT char ModPlug_ExportS3M(ModPlugFile* file, const char* filepath);
/*Export to a Extended Module (XM). EXPERIMENTAL (only works on Little-Endian platforms)*/
MODPLUG_EXPORT char ModPlug_ExportXM(ModPlugFile* file, const char* filepath);
/*Export to a Amiga MOD file. EXPERIMENTAL.*/
MODPLUG_EXPORT char ModPlug_ExportMOD(ModPlugFile* file, const char* filepath);
/*Export to a Impulse Tracker IT file. Should work OK in Little-Endian & Big-Endian platforms :-) */
MODPLUG_EXPORT char ModPlug_ExportIT(ModPlugFile* file, const char* filepath);
#endif /* MODPLUG_NO_FILESAVE */
MODPLUG_EXPORT unsigned int ModPlug_NumInstruments(ModPlugFile* file);
MODPLUG_EXPORT unsigned int ModPlug_NumSamples(ModPlugFile* file);
MODPLUG_EXPORT unsigned int ModPlug_NumPatterns(ModPlugFile* file);
MODPLUG_EXPORT unsigned int ModPlug_NumChannels(ModPlugFile* file);
MODPLUG_EXPORT unsigned int ModPlug_SampleName(ModPlugFile* file, unsigned int qual, char* buff);
MODPLUG_EXPORT unsigned int ModPlug_InstrumentName(ModPlugFile* file, unsigned int qual, char* buff);
/*
* Retrieve pattern note-data
*/
MODPLUG_EXPORT ModPlugNote* ModPlug_GetPattern(ModPlugFile* file, int pattern, unsigned int* numrows);
/*
* =================
* Mixer callback
* =================
*
* Use this callback if you want to 'modify' the mixed data of LibModPlug.
*
* void proc(int* buffer,unsigned long channels,unsigned long nsamples) ;
*
* 'buffer': A buffer of mixed samples
* 'channels': N. of channels in the buffer
* 'nsamples': N. of samples in the buffeer (without taking care of n.channels)
*
* (Samples are signed 32-bit integers)
*/
MODPLUG_EXPORT void ModPlug_InitMixerCallback(ModPlugFile* file,ModPlugMixerProc proc) ;
MODPLUG_EXPORT void ModPlug_UnloadMixerCallback(ModPlugFile* file) ;
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif

@ -0,0 +1,485 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
#ifdef MODPLUG_FASTSOUNDLIB
#define MODPLUG_NO_REVERB
#endif
// Delayed Surround Filters
#ifndef MODPLUG_FASTSOUNDLIB
#define nDolbyHiFltAttn 6
#define nDolbyHiFltMask 3
#define DOLBYATTNROUNDUP 31
#else
#define nDolbyHiFltAttn 3
#define nDolbyHiFltMask 3
#define DOLBYATTNROUNDUP 3
#endif
// Bass Expansion
#define XBASS_DELAY 14 // 2.5 ms
// Buffer Sizes
#define XBASSBUFFERSIZE 64 // 2 ms at 50KHz
#define FILTERBUFFERSIZE 64 // 1.25 ms
#define SURROUNDBUFFERSIZE ((MAX_SAMPLE_RATE * 50) / 1000)
#define REVERBBUFFERSIZE ((MAX_SAMPLE_RATE * 200) / 1000)
#define REVERBBUFFERSIZE2 ((REVERBBUFFERSIZE*13) / 17)
#define REVERBBUFFERSIZE3 ((REVERBBUFFERSIZE*7) / 13)
#define REVERBBUFFERSIZE4 ((REVERBBUFFERSIZE*7) / 19)
// DSP Effects: PUBLIC members
UINT CSoundFile::m_nXBassDepth = 6;
UINT CSoundFile::m_nXBassRange = XBASS_DELAY;
UINT CSoundFile::m_nReverbDepth = 1;
UINT CSoundFile::m_nReverbDelay = 100;
UINT CSoundFile::m_nProLogicDepth = 12;
UINT CSoundFile::m_nProLogicDelay = 20;
////////////////////////////////////////////////////////////////////
// DSP Effects internal state
// Bass Expansion: low-pass filter
static LONG nXBassSum = 0;
static LONG nXBassBufferPos = 0;
static LONG nXBassDlyPos = 0;
static LONG nXBassMask = 0;
// Noise Reduction: simple low-pass filter
static LONG nLeftNR = 0;
static LONG nRightNR = 0;
// Surround Encoding: 1 delay line + low-pass filter + high-pass filter
static LONG nSurroundSize = 0;
static LONG nSurroundPos = 0;
static LONG nDolbyDepth = 0;
static LONG nDolbyLoDlyPos = 0;
static LONG nDolbyLoFltPos = 0;
static LONG nDolbyLoFltSum = 0;
static LONG nDolbyHiFltPos = 0;
static LONG nDolbyHiFltSum = 0;
// Reverb: 4 delay lines + high-pass filter + low-pass filter
#ifndef MODPLUG_NO_REVERB
static LONG nReverbSize = 0;
static LONG nReverbBufferPos = 0;
static LONG nReverbSize2 = 0;
static LONG nReverbBufferPos2 = 0;
static LONG nReverbSize3 = 0;
static LONG nReverbBufferPos3 = 0;
static LONG nReverbSize4 = 0;
static LONG nReverbBufferPos4 = 0;
static LONG nReverbLoFltSum = 0;
static LONG nReverbLoFltPos = 0;
static LONG nReverbLoDlyPos = 0;
static LONG nFilterAttn = 0;
static LONG gRvbLowPass[8];
static LONG gRvbLPPos = 0;
static LONG gRvbLPSum = 0;
static LONG ReverbLoFilterBuffer[XBASSBUFFERSIZE];
static LONG ReverbLoFilterDelay[XBASSBUFFERSIZE];
static LONG ReverbBuffer[REVERBBUFFERSIZE];
static LONG ReverbBuffer2[REVERBBUFFERSIZE2];
static LONG ReverbBuffer3[REVERBBUFFERSIZE3];
static LONG ReverbBuffer4[REVERBBUFFERSIZE4];
#endif
static LONG XBassBuffer[XBASSBUFFERSIZE];
static LONG XBassDelay[XBASSBUFFERSIZE];
static LONG DolbyLoFilterBuffer[XBASSBUFFERSIZE];
static LONG DolbyLoFilterDelay[XBASSBUFFERSIZE];
static LONG DolbyHiFilterBuffer[FILTERBUFFERSIZE];
static LONG SurroundBuffer[SURROUNDBUFFERSIZE];
// Access the main temporary mix buffer directly: avoids an extra pointer
extern int MixSoundBuffer[MIXBUFFERSIZE*4];
//cextern int MixReverbBuffer[MIXBUFFERSIZE*2];
extern int MixReverbBuffer[MIXBUFFERSIZE*2];
static UINT GetMaskFromSize(UINT len)
//-----------------------------------
{
UINT n = 2;
while (n <= len) n <<= 1;
return ((n >> 1) - 1);
}
void CSoundFile::InitializeDSP(BOOL bReset)
//-----------------------------------------
{
if (!m_nReverbDelay) m_nReverbDelay = 100;
if (!m_nXBassRange) m_nXBassRange = XBASS_DELAY;
if (!m_nProLogicDelay) m_nProLogicDelay = 20;
if (m_nXBassDepth > 8) m_nXBassDepth = 8;
if (m_nXBassDepth < 2) m_nXBassDepth = 2;
if (bReset)
{
// Noise Reduction
nLeftNR = nRightNR = 0;
}
// Pro-Logic Surround
nSurroundPos = nSurroundSize = 0;
nDolbyLoFltPos = nDolbyLoFltSum = nDolbyLoDlyPos = 0;
nDolbyHiFltPos = nDolbyHiFltSum = 0;
if (gdwSoundSetup & SNDMIX_SURROUND)
{
memset(DolbyLoFilterBuffer, 0, sizeof(DolbyLoFilterBuffer));
memset(DolbyHiFilterBuffer, 0, sizeof(DolbyHiFilterBuffer));
memset(DolbyLoFilterDelay, 0, sizeof(DolbyLoFilterDelay));
memset(SurroundBuffer, 0, sizeof(SurroundBuffer));
nSurroundSize = (gdwMixingFreq * m_nProLogicDelay) / 1000;
if (nSurroundSize > SURROUNDBUFFERSIZE) nSurroundSize = SURROUNDBUFFERSIZE;
if (m_nProLogicDepth < 8) nDolbyDepth = (32 >> m_nProLogicDepth) + 32;
else nDolbyDepth = (m_nProLogicDepth < 16) ? (8 + (m_nProLogicDepth - 8) * 7) : 64;
nDolbyDepth >>= 2;
}
// Reverb Setup
#ifndef MODPLUG_NO_REVERB
if (gdwSoundSetup & SNDMIX_REVERB)
{
UINT nrs = (gdwMixingFreq * m_nReverbDelay) / 1000;
UINT nfa = m_nReverbDepth+1;
if (nrs > REVERBBUFFERSIZE) nrs = REVERBBUFFERSIZE;
if ((bReset) || (nrs != (UINT)nReverbSize) || (nfa != (UINT)nFilterAttn))
{
nFilterAttn = nfa;
nReverbSize = nrs;
nReverbBufferPos = nReverbBufferPos2 = nReverbBufferPos3 = nReverbBufferPos4 = 0;
nReverbLoFltSum = nReverbLoFltPos = nReverbLoDlyPos = 0;
gRvbLPSum = gRvbLPPos = 0;
nReverbSize2 = (nReverbSize * 13) / 17;
if (nReverbSize2 > REVERBBUFFERSIZE2) nReverbSize2 = REVERBBUFFERSIZE2;
nReverbSize3 = (nReverbSize * 7) / 13;
if (nReverbSize3 > REVERBBUFFERSIZE3) nReverbSize3 = REVERBBUFFERSIZE3;
nReverbSize4 = (nReverbSize * 7) / 19;
if (nReverbSize4 > REVERBBUFFERSIZE4) nReverbSize4 = REVERBBUFFERSIZE4;
memset(ReverbLoFilterBuffer, 0, sizeof(ReverbLoFilterBuffer));
memset(ReverbLoFilterDelay, 0, sizeof(ReverbLoFilterDelay));
memset(ReverbBuffer, 0, sizeof(ReverbBuffer));
memset(ReverbBuffer2, 0, sizeof(ReverbBuffer2));
memset(ReverbBuffer3, 0, sizeof(ReverbBuffer3));
memset(ReverbBuffer4, 0, sizeof(ReverbBuffer4));
memset(gRvbLowPass, 0, sizeof(gRvbLowPass));
}
} else nReverbSize = 0;
#endif
BOOL bResetBass = FALSE;
// Bass Expansion Reset
if (gdwSoundSetup & SNDMIX_MEGABASS)
{
UINT nXBassSamples = (gdwMixingFreq * m_nXBassRange) / 10000;
if (nXBassSamples > XBASSBUFFERSIZE) nXBassSamples = XBASSBUFFERSIZE;
UINT mask = GetMaskFromSize(nXBassSamples);
if ((bReset) || (mask != (UINT)nXBassMask))
{
nXBassMask = mask;
bResetBass = TRUE;
}
} else
{
nXBassMask = 0;
bResetBass = TRUE;
}
if (bResetBass)
{
nXBassSum = nXBassBufferPos = nXBassDlyPos = 0;
memset(XBassBuffer, 0, sizeof(XBassBuffer));
memset(XBassDelay, 0, sizeof(XBassDelay));
}
}
void CSoundFile::ProcessStereoDSP(int count)
//------------------------------------------
{
#ifndef MODPLUG_NO_REVERB
// Reverb
if (gdwSoundSetup & SNDMIX_REVERB)
{
int *pr = MixSoundBuffer, *pin = MixReverbBuffer, rvbcount = count;
do
{
int echo = ReverbBuffer[nReverbBufferPos] + ReverbBuffer2[nReverbBufferPos2]
+ ReverbBuffer3[nReverbBufferPos3] + ReverbBuffer4[nReverbBufferPos4]; // echo = reverb signal
// Delay line and remove Low Frequencies // v = original signal
int echodly = ReverbLoFilterDelay[nReverbLoDlyPos]; // echodly = delayed signal
ReverbLoFilterDelay[nReverbLoDlyPos] = echo >> 1;
nReverbLoDlyPos++;
nReverbLoDlyPos &= 0x1F;
int n = nReverbLoFltPos;
nReverbLoFltSum -= ReverbLoFilterBuffer[n];
int tmp = echo / 128;
ReverbLoFilterBuffer[n] = tmp;
nReverbLoFltSum += tmp;
echodly -= nReverbLoFltSum;
nReverbLoFltPos = (n + 1) & 0x3F;
// Reverb
int v = (pin[0]+pin[1]) >> nFilterAttn;
pr[0] += pin[0] + echodly;
pr[1] += pin[1] + echodly;
v += echodly >> 2;
ReverbBuffer3[nReverbBufferPos3] = v;
ReverbBuffer4[nReverbBufferPos4] = v;
v += echodly >> 4;
v >>= 1;
gRvbLPSum -= gRvbLowPass[gRvbLPPos];
gRvbLPSum += v;
gRvbLowPass[gRvbLPPos] = v;
gRvbLPPos++;
gRvbLPPos &= 7;
int vlp = gRvbLPSum >> 2;
ReverbBuffer[nReverbBufferPos] = vlp;
ReverbBuffer2[nReverbBufferPos2] = vlp;
if (++nReverbBufferPos >= nReverbSize) nReverbBufferPos = 0;
if (++nReverbBufferPos2 >= nReverbSize2) nReverbBufferPos2 = 0;
if (++nReverbBufferPos3 >= nReverbSize3) nReverbBufferPos3 = 0;
if (++nReverbBufferPos4 >= nReverbSize4) nReverbBufferPos4 = 0;
pr += 2;
pin += 2;
} while (--rvbcount);
}
#endif
// Dolby Pro-Logic Surround
if (gdwSoundSetup & SNDMIX_SURROUND)
{
int *pr = MixSoundBuffer, n = nDolbyLoFltPos;
for (int r=count; r; r--)
{
int v = (pr[0]+pr[1]+DOLBYATTNROUNDUP) >> (nDolbyHiFltAttn+1);
#ifndef MODPLUG_FASTSOUNDLIB
v *= (int)nDolbyDepth;
#endif
// Low-Pass Filter
nDolbyHiFltSum -= DolbyHiFilterBuffer[nDolbyHiFltPos];
DolbyHiFilterBuffer[nDolbyHiFltPos] = v;
nDolbyHiFltSum += v;
v = nDolbyHiFltSum;
nDolbyHiFltPos++;
nDolbyHiFltPos &= nDolbyHiFltMask;
// Surround
int secho = SurroundBuffer[nSurroundPos];
SurroundBuffer[nSurroundPos] = v;
// Delay line and remove low frequencies
v = DolbyLoFilterDelay[nDolbyLoDlyPos]; // v = delayed signal
DolbyLoFilterDelay[nDolbyLoDlyPos] = secho; // secho = signal
nDolbyLoDlyPos++;
nDolbyLoDlyPos &= 0x1F;
nDolbyLoFltSum -= DolbyLoFilterBuffer[n];
int tmp = secho / 64;
DolbyLoFilterBuffer[n] = tmp;
nDolbyLoFltSum += tmp;
v -= nDolbyLoFltSum;
n++;
n &= 0x3F;
// Add echo
pr[0] += v;
pr[1] -= v;
if (++nSurroundPos >= nSurroundSize) nSurroundPos = 0;
pr += 2;
}
nDolbyLoFltPos = n;
}
// Bass Expansion
if (gdwSoundSetup & SNDMIX_MEGABASS)
{
int *px = MixSoundBuffer;
int xba = m_nXBassDepth+1, xbamask = (1 << xba) - 1;
int n = nXBassBufferPos;
for (int x=count; x; x--)
{
nXBassSum -= XBassBuffer[n];
int tmp0 = px[0] + px[1];
int tmp = (tmp0 + ((tmp0 >> 31) & xbamask)) >> xba;
XBassBuffer[n] = tmp;
nXBassSum += tmp;
int v = XBassDelay[nXBassDlyPos];
XBassDelay[nXBassDlyPos] = px[0];
px[0] = v + nXBassSum;
v = XBassDelay[nXBassDlyPos+1];
XBassDelay[nXBassDlyPos+1] = px[1];
px[1] = v + nXBassSum;
nXBassDlyPos = (nXBassDlyPos + 2) & nXBassMask;
px += 2;
n++;
n &= nXBassMask;
}
nXBassBufferPos = n;
}
// Noise Reduction
if (gdwSoundSetup & SNDMIX_NOISEREDUCTION)
{
int n1 = nLeftNR, n2 = nRightNR;
int *pnr = MixSoundBuffer;
for (int nr=count; nr; nr--)
{
int vnr = pnr[0] >> 1;
pnr[0] = vnr + n1;
n1 = vnr;
vnr = pnr[1] >> 1;
pnr[1] = vnr + n2;
n2 = vnr;
pnr += 2;
}
nLeftNR = n1;
nRightNR = n2;
}
}
void CSoundFile::ProcessMonoDSP(int count)
//----------------------------------------
{
#ifndef MODPLUG_NO_REVERB
// Reverb
if (gdwSoundSetup & SNDMIX_REVERB)
{
int *pr = MixSoundBuffer, rvbcount = count, *pin = MixReverbBuffer;
do
{
int echo = ReverbBuffer[nReverbBufferPos] + ReverbBuffer2[nReverbBufferPos2]
+ ReverbBuffer3[nReverbBufferPos3] + ReverbBuffer4[nReverbBufferPos4]; // echo = reverb signal
// Delay line and remove Low Frequencies // v = original signal
int echodly = ReverbLoFilterDelay[nReverbLoDlyPos]; // echodly = delayed signal
ReverbLoFilterDelay[nReverbLoDlyPos] = echo >> 1;
nReverbLoDlyPos++;
nReverbLoDlyPos &= 0x1F;
int n = nReverbLoFltPos;
nReverbLoFltSum -= ReverbLoFilterBuffer[n];
int tmp = echo / 128;
ReverbLoFilterBuffer[n] = tmp;
nReverbLoFltSum += tmp;
echodly -= nReverbLoFltSum;
nReverbLoFltPos = (n + 1) & 0x3F;
// Reverb
int v = pin[0] >> (nFilterAttn-1);
*pr++ += pin[0] + echodly;
pin++;
v += echodly >> 2;
ReverbBuffer3[nReverbBufferPos3] = v;
ReverbBuffer4[nReverbBufferPos4] = v;
v += echodly >> 4;
v >>= 1;
gRvbLPSum -= gRvbLowPass[gRvbLPPos];
gRvbLPSum += v;
gRvbLowPass[gRvbLPPos] = v;
gRvbLPPos++;
gRvbLPPos &= 7;
int vlp = gRvbLPSum >> 2;
ReverbBuffer[nReverbBufferPos] = vlp;
ReverbBuffer2[nReverbBufferPos2] = vlp;
if (++nReverbBufferPos >= nReverbSize) nReverbBufferPos = 0;
if (++nReverbBufferPos2 >= nReverbSize2) nReverbBufferPos2 = 0;
if (++nReverbBufferPos3 >= nReverbSize3) nReverbBufferPos3 = 0;
if (++nReverbBufferPos4 >= nReverbSize4) nReverbBufferPos4 = 0;
} while (--rvbcount);
}
#endif
// Bass Expansion
if (gdwSoundSetup & SNDMIX_MEGABASS)
{
int *px = MixSoundBuffer;
int xba = m_nXBassDepth, xbamask = (1 << xba)-1;
int n = nXBassBufferPos;
for (int x=count; x; x--)
{
nXBassSum -= XBassBuffer[n];
int tmp0 = *px;
int tmp = (tmp0 + ((tmp0 >> 31) & xbamask)) >> xba;
XBassBuffer[n] = tmp;
nXBassSum += tmp;
int v = XBassDelay[nXBassDlyPos];
XBassDelay[nXBassDlyPos] = *px;
*px++ = v + nXBassSum;
nXBassDlyPos = (nXBassDlyPos + 2) & nXBassMask;
n++;
n &= nXBassMask;
}
nXBassBufferPos = n;
}
// Noise Reduction
if (gdwSoundSetup & SNDMIX_NOISEREDUCTION)
{
int n = nLeftNR;
int *pnr = MixSoundBuffer;
for (int nr=count; nr; pnr++, nr--)
{
int vnr = *pnr >> 1;
*pnr = vnr + n;
n = vnr;
}
nLeftNR = n;
}
}
/////////////////////////////////////////////////////////////////
// Clean DSP Effects interface
// [Reverb level 0(quiet)-100(loud)], [delay in ms, usually 40-200ms]
BOOL CSoundFile::SetReverbParameters(UINT nDepth, UINT nDelay)
//------------------------------------------------------------
{
if (nDepth > 100) nDepth = 100;
UINT gain = nDepth / 20;
if (gain > 4) gain = 4;
m_nReverbDepth = 4 - gain;
if (nDelay < 40) nDelay = 40;
if (nDelay > 250) nDelay = 250;
m_nReverbDelay = nDelay;
return TRUE;
}
// [XBass level 0(quiet)-100(loud)], [cutoff in Hz 20-100]
BOOL CSoundFile::SetXBassParameters(UINT nDepth, UINT nRange)
//-----------------------------------------------------------
{
if (nDepth > 100) nDepth = 100;
UINT gain = nDepth / 20;
if (gain > 4) gain = 4;
m_nXBassDepth = 8 - gain; // filter attenuation 1/256 .. 1/16
UINT range = nRange / 5;
if (range > 5) range -= 5; else range = 0;
if (nRange > 16) nRange = 16;
m_nXBassRange = 21 - range; // filter average on 0.5-1.6ms
return TRUE;
}
// [Surround level 0(quiet)-100(heavy)] [delay in ms, usually 5-50ms]
BOOL CSoundFile::SetSurroundParameters(UINT nDepth, UINT nDelay)
//--------------------------------------------------------------
{
UINT gain = (nDepth * 16) / 100;
if (gain > 16) gain = 16;
if (gain < 1) gain = 1;
m_nProLogicDepth = gain;
if (nDelay < 4) nDelay = 4;
if (nDelay > 50) nDelay = 50;
m_nProLogicDelay = nDelay;
return TRUE;
}
BOOL CSoundFile::SetWaveConfigEx(BOOL bSurround,BOOL bNoOverSampling,BOOL bReverb,BOOL hqido,BOOL bMegaBass,BOOL bNR,BOOL bEQ)
//----------------------------------------------------------------------------------------------------------------------------
{
DWORD d = gdwSoundSetup & ~(SNDMIX_SURROUND | SNDMIX_NORESAMPLING | SNDMIX_REVERB | SNDMIX_HQRESAMPLER | SNDMIX_MEGABASS | SNDMIX_NOISEREDUCTION | SNDMIX_EQ);
if (bSurround) d |= SNDMIX_SURROUND;
if (bNoOverSampling) d |= SNDMIX_NORESAMPLING;
if (bReverb) d |= SNDMIX_REVERB;
if (hqido) d |= SNDMIX_HQRESAMPLER;
if (bMegaBass) d |= SNDMIX_MEGABASS;
if (bNR) d |= SNDMIX_NOISEREDUCTION;
if (bEQ) d |= SNDMIX_EQ;
gdwSoundSetup = d;
InitPlayer(FALSE);
return TRUE;
}

@ -0,0 +1,101 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#include "stdafx.h"
#include "sndfile.h"
// AWE32: cutoff = reg[0-255] * 31.25 + 100 -> [100Hz-8060Hz]
// EMU10K1 docs: cutoff = reg[0-127]*62+100
#define FILTER_PRECISION 8192
#ifndef NO_FILTER
#ifdef MSC_VER
#define _ASM_MATH
#endif
#ifdef _ASM_MATH
// pow(a,b) returns a^^b -> 2^^(b.log2(a))
static float pow(float a, float b)
{
long tmpint;
float result;
_asm {
fld b // Load b
fld a // Load a
fyl2x // ST(0) = b.log2(a)
fist tmpint // Store integer exponent
fisub tmpint // ST(0) = -1 <= (b*log2(a)) <= 1
f2xm1 // ST(0) = 2^(x)-1
fild tmpint // load integer exponent
fld1 // Load 1
fscale // ST(0) = 2^ST(1)
fstp ST(1) // Remove the integer from the stack
fmul ST(1), ST(0) // multiply with fractional part
faddp ST(1), ST(0) // add integer_part
fstp result // Store the result
}
return result;
}
#else
#include <math.h>
#endif // _ASM_MATH
DWORD CSoundFile::CutOffToFrequency(UINT nCutOff, int flt_modifier) const
//-----------------------------------------------------------------------
{
float Fc;
if (m_dwSongFlags & SONG_EXFILTERRANGE)
Fc = 110.0f * pow(2.0f, 0.25f + ((float)(nCutOff*(flt_modifier+256)))/(21.0f*512.0f));
else
Fc = 110.0f * pow(2.0f, 0.25f + ((float)(nCutOff*(flt_modifier+256)))/(24.0f*512.0f));
LONG freq = (LONG)Fc;
if (freq < 120) return 120;
if (freq > 10000) return 10000;
if (freq*2 > (LONG)gdwMixingFreq) freq = gdwMixingFreq>>1;
return (DWORD)freq;
}
// Simple 2-poles resonant filter
void CSoundFile::SetupChannelFilter(MODCHANNEL *pChn, BOOL bReset, int flt_modifier) const
//----------------------------------------------------------------------------------------
{
float fc = (float)CutOffToFrequency(pChn->nCutOff, flt_modifier);
float fs = (float)gdwMixingFreq;
float fg, fb0, fb1;
fc *= (float)(2.0*3.14159265358/fs);
float dmpfac = pow(10.0f, -((24.0f / 128.0f)*(float)pChn->nResonance) / 20.0f);
float d = (1.0f-2.0f*dmpfac)* fc;
if (d>2.0) d = 2.0;
d = (2.0f*dmpfac - d)/fc;
float e = pow(1.0f/fc,2.0f);
fg=1/(1+d+e);
fb0=(d+e+e)/(1+d+e);
fb1=-e/(1+d+e);
pChn->nFilter_A0 = (int)(fg * FILTER_PRECISION);
pChn->nFilter_B0 = (int)(fb0 * FILTER_PRECISION);
pChn->nFilter_B1 = (int)(fb1 * FILTER_PRECISION);
if (bReset)
{
pChn->nFilter_Y1 = pChn->nFilter_Y2 = 0;
pChn->nFilter_Y3 = pChn->nFilter_Y4 = 0;
}
pChn->dwFlags |= CHN_FILTER;
}
#endif // NO_FILTER

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

@ -0,0 +1,141 @@
/*
* This source code is public domain.
*
* Authors: Rani Assaf <rani@magic.metawire.com>,
* Olivier Lapicque <olivierl@jps.net>,
* Adam Goode <adam@evdebs.org> (endian and char fixes for PPC)
*/
#ifndef _STDAFX_H_
#define _STDAFX_H_
/* Autoconf detection of stdint/inttypes */
#if defined(HAVE_CONFIG_H) && !defined(CONFIG_H_INCLUDED)
# include "config.h"
# define CONFIG_H_INCLUDED 1
#endif
#ifdef HAVE_INTTYPES_H
# include <inttypes.h>
#endif
#ifdef HAVE_STDINT_H
# include <stdint.h>
#endif
/* disable AGC and FILESAVE for all targets for uniformity. */
#define NO_AGC
#define MODPLUG_NO_FILESAVE
#ifdef _WIN32
#ifdef MSC_VER
#pragma warning (disable:4201)
#pragma warning (disable:4514)
#endif
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <windowsx.h>
#include <mmsystem.h>
#include <stdio.h>
#include <malloc.h>
#include <stdint.h>
#define srandom(_seed) srand(_seed)
#define random() rand()
#define sleep(_ms) Sleep(_ms)
inline void ProcessPlugins(int n) {}
#undef strcasecmp
#undef strncasecmp
#define strcasecmp(a,b) _stricmp(a,b)
#define strncasecmp(a,b,c) _strnicmp(a,b,c)
#define HAVE_SINF 1
#ifndef isblank
#define isblank(c) ((c) == ' ' || (c) == '\t')
#endif
#else
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif
typedef int8_t CHAR;
typedef uint8_t UCHAR;
typedef uint8_t* PUCHAR;
typedef uint16_t USHORT;
typedef uint32_t ULONG;
typedef uint32_t UINT;
typedef uint32_t DWORD;
typedef int32_t LONG;
typedef int64_t LONGLONG;
typedef int32_t* LPLONG;
typedef uint32_t* LPDWORD;
typedef uint16_t WORD;
typedef uint8_t BYTE;
typedef uint8_t* LPBYTE;
typedef bool BOOL;
typedef char* LPSTR;
typedef void* LPVOID;
typedef uint16_t* LPWORD;
typedef const char* LPCSTR;
typedef void* PVOID;
typedef void VOID;
inline LONG MulDiv (long a, long b, long c)
{
/*if (!c) return 0;*/
return ((uint64_t) a * (uint64_t) b ) / c;
}
#define LPCTSTR LPCSTR
#define lstrcpyn strncpy
#define lstrcpy strcpy
#define lstrcmp strcmp
#define wsprintf sprintf
#define WAVE_FORMAT_PCM 1
#define GHND 0
#define GlobalFreePtr(p) free((void *)(p))
inline int8_t * GlobalAllocPtr(unsigned int, size_t size)
{
int8_t * p = (int8_t *) malloc(size);
if (p != NULL) memset(p, 0, size);
return p;
}
inline void ProcessPlugins(int n) {}
#ifndef FALSE
#define FALSE false
#endif
#ifndef TRUE
#define TRUE true
#endif
#endif /* _WIN32 */
#if defined(_WIN32) || defined(__CYGWIN__)
# if defined(MODPLUG_BUILD) && defined(DLL_EXPORT) /* building libmodplug as a dll for windows */
# define MODPLUG_EXPORT __declspec(dllexport)
# elif defined(MODPLUG_BUILD) || defined(MODPLUG_STATIC) /* building or using static libmodplug for windows */
# define MODPLUG_EXPORT
# else
# define MODPLUG_EXPORT __declspec(dllimport) /* using libmodplug dll for windows */
# endif
#elif defined(MODPLUG_BUILD) && defined(SYM_VISIBILITY)
# define MODPLUG_EXPORT __attribute__((visibility("default")))
#else
#define MODPLUG_EXPORT
#endif
#endif

@ -0,0 +1,377 @@
/*
* This source code is public domain.
*
* Authors: Olivier Lapicque <olivierl@jps.net>
*/
#pragma once
#include "stdafx.h"
#include "sndfile.h"
#ifndef MODPLUG_FASTSOUNDLIB
//#pragma data_seg(".tables")
#endif
static const BYTE ImpulseTrackerPortaVolCmd[16] =
{
0x00, 0x01, 0x04, 0x08, 0x10, 0x20, 0x40, 0x60,
0x80, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF
};
// Period table for Protracker octaves 0-5:
static const WORD ProTrackerPeriodTable[6*12] =
{
1712,1616,1524,1440,1356,1280,1208,1140,1076,1016,960,907,
856,808,762,720,678,640,604,570,538,508,480,453,
428,404,381,360,339,320,302,285,269,254,240,226,
214,202,190,180,170,160,151,143,135,127,120,113,
107,101,95,90,85,80,75,71,67,63,60,56,
53,50,47,45,42,40,37,35,33,31,30,28
};
static const WORD ProTrackerTunedPeriods[16*12] =
{
1712,1616,1524,1440,1356,1280,1208,1140,1076,1016,960,907,
1700,1604,1514,1430,1348,1274,1202,1134,1070,1010,954,900,
1688,1592,1504,1418,1340,1264,1194,1126,1064,1004,948,894,
1676,1582,1492,1408,1330,1256,1184,1118,1056,996,940,888,
1664,1570,1482,1398,1320,1246,1176,1110,1048,990,934,882,
1652,1558,1472,1388,1310,1238,1168,1102,1040,982,926,874,
1640,1548,1460,1378,1302,1228,1160,1094,1032,974,920,868,
1628,1536,1450,1368,1292,1220,1150,1086,1026,968,914,862,
1814,1712,1616,1524,1440,1356,1280,1208,1140,1076,1016,960,
1800,1700,1604,1514,1430,1350,1272,1202,1134,1070,1010,954,
1788,1688,1592,1504,1418,1340,1264,1194,1126,1064,1004,948,
1774,1676,1582,1492,1408,1330,1256,1184,1118,1056,996,940,
1762,1664,1570,1482,1398,1320,1246,1176,1110,1048,988,934,
1750,1652,1558,1472,1388,1310,1238,1168,1102,1040,982,926,
1736,1640,1548,1460,1378,1302,1228,1160,1094,1032,974,920,
1724,1628,1536,1450,1368,1292,1220,1150,1086,1026,968,914
};
// S3M C-4 periods
static const WORD FreqS3MTable[16] =
{
1712,1616,1524,1440,1356,1280,
1208,1140,1076,1016,960,907,
0,0,0,0
};
// S3M FineTune frequencies
static const WORD S3MFineTuneTable[16] =
{
7895,7941,7985,8046,8107,8169,8232,8280,
8363,8413,8463,8529,8581,8651,8723,8757, // 8363*2^((i-8)/(12*8))
};
// Sinus table
static const int16_t ModSinusTable[64] =
{
0,12,25,37,49,60,71,81,90,98,106,112,117,122,125,126,
127,126,125,122,117,112,106,98,90,81,71,60,49,37,25,12,
0,-12,-25,-37,-49,-60,-71,-81,-90,-98,-106,-112,-117,-122,-125,-126,
-127,-126,-125,-122,-117,-112,-106,-98,-90,-81,-71,-60,-49,-37,-25,-12
};
// Triangle wave table (ramp down)
static const int16_t ModRampDownTable[64] =
{
0,-4,-8,-12,-16,-20,-24,-28,-32,-36,-40,-44,-48,-52,-56,-60,
-64,-68,-72,-76,-80,-84,-88,-92,-96,-100,-104,-108,-112,-116,-120,-124,
127,123,119,115,111,107,103,99,95,91,87,83,79,75,71,67,
63,59,55,51,47,43,39,35,31,27,23,19,15,11,7,3
};
// Square wave table
static const int16_t ModSquareTable[64] =
{
127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,
127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,
-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,
-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127,-127
};
// Random wave table
static const int16_t ModRandomTable[64] =
{
98,-127,-43,88,102,41,-65,-94,125,20,-71,-86,-70,-32,-16,-96,
17,72,107,-5,116,-69,-62,-40,10,-61,65,109,-18,-38,-13,-76,
-23,88,21,-94,8,106,21,-112,6,109,20,-88,-30,9,-127,118,
42,-34,89,-4,-51,-72,21,-29,112,123,84,-101,-92,98,-54,-95
};
// volume fade tables for Retrig Note:
static const int8_t retrigTable1[16] =
{ 0, 0, 0, 0, 0, 0, 10, 8, 0, 0, 0, 0, 0, 0, 24, 32 };
static const int8_t retrigTable2[16] =
{ 0, -1, -2, -4, -8, -16, 0, 0, 0, 1, 2, 4, 8, 16, 0, 0 };
static const WORD XMPeriodTable[104] =
{
907,900,894,887,881,875,868,862,856,850,844,838,832,826,820,814,
808,802,796,791,785,779,774,768,762,757,752,746,741,736,730,725,
720,715,709,704,699,694,689,684,678,675,670,665,660,655,651,646,
640,636,632,628,623,619,614,610,604,601,597,592,588,584,580,575,
570,567,563,559,555,551,547,543,538,535,532,528,524,520,516,513,
508,505,502,498,494,491,487,484,480,477,474,470,467,463,460,457,
453,450,447,443,440,437,434,431
};
static const uint32_t XMLinearTable[768] =
{
535232,534749,534266,533784,533303,532822,532341,531861,
531381,530902,530423,529944,529466,528988,528511,528034,
527558,527082,526607,526131,525657,525183,524709,524236,
523763,523290,522818,522346,521875,521404,520934,520464,
519994,519525,519057,518588,518121,517653,517186,516720,
516253,515788,515322,514858,514393,513929,513465,513002,
512539,512077,511615,511154,510692,510232,509771,509312,
508852,508393,507934,507476,507018,506561,506104,505647,
505191,504735,504280,503825,503371,502917,502463,502010,
501557,501104,500652,500201,499749,499298,498848,498398,
497948,497499,497050,496602,496154,495706,495259,494812,
494366,493920,493474,493029,492585,492140,491696,491253,
490809,490367,489924,489482,489041,488600,488159,487718,
487278,486839,486400,485961,485522,485084,484647,484210,
483773,483336,482900,482465,482029,481595,481160,480726,
480292,479859,479426,478994,478562,478130,477699,477268,
476837,476407,475977,475548,475119,474690,474262,473834,
473407,472979,472553,472126,471701,471275,470850,470425,
470001,469577,469153,468730,468307,467884,467462,467041,
466619,466198,465778,465358,464938,464518,464099,463681,
463262,462844,462427,462010,461593,461177,460760,460345,
459930,459515,459100,458686,458272,457859,457446,457033,
456621,456209,455797,455386,454975,454565,454155,453745,
453336,452927,452518,452110,451702,451294,450887,450481,
450074,449668,449262,448857,448452,448048,447644,447240,
446836,446433,446030,445628,445226,444824,444423,444022,
443622,443221,442821,442422,442023,441624,441226,440828,
440430,440033,439636,439239,438843,438447,438051,437656,
437261,436867,436473,436079,435686,435293,434900,434508,
434116,433724,433333,432942,432551,432161,431771,431382,
430992,430604,430215,429827,429439,429052,428665,428278,
427892,427506,427120,426735,426350,425965,425581,425197,
424813,424430,424047,423665,423283,422901,422519,422138,
421757,421377,420997,420617,420237,419858,419479,419101,
418723,418345,417968,417591,417214,416838,416462,416086,
415711,415336,414961,414586,414212,413839,413465,413092,
412720,412347,411975,411604,411232,410862,410491,410121,
409751,409381,409012,408643,408274,407906,407538,407170,
406803,406436,406069,405703,405337,404971,404606,404241,
403876,403512,403148,402784,402421,402058,401695,401333,
400970,400609,400247,399886,399525,399165,398805,398445,
398086,397727,397368,397009,396651,396293,395936,395579,
395222,394865,394509,394153,393798,393442,393087,392733,
392378,392024,391671,391317,390964,390612,390259,389907,
389556,389204,388853,388502,388152,387802,387452,387102,
386753,386404,386056,385707,385359,385012,384664,384317,
383971,383624,383278,382932,382587,382242,381897,381552,
381208,380864,380521,380177,379834,379492,379149,378807,
378466,378124,377783,377442,377102,376762,376422,376082,
375743,375404,375065,374727,374389,374051,373714,373377,
373040,372703,372367,372031,371695,371360,371025,370690,
370356,370022,369688,369355,369021,368688,368356,368023,
367691,367360,367028,366697,366366,366036,365706,365376,
365046,364717,364388,364059,363731,363403,363075,362747,
362420,362093,361766,361440,361114,360788,360463,360137,
359813,359488,359164,358840,358516,358193,357869,357547,
357224,356902,356580,356258,355937,355616,355295,354974,
354654,354334,354014,353695,353376,353057,352739,352420,
352103,351785,351468,351150,350834,350517,350201,349885,
349569,349254,348939,348624,348310,347995,347682,347368,
347055,346741,346429,346116,345804,345492,345180,344869,
344558,344247,343936,343626,343316,343006,342697,342388,
342079,341770,341462,341154,340846,340539,340231,339924,
339618,339311,339005,338700,338394,338089,337784,337479,
337175,336870,336566,336263,335959,335656,335354,335051,
334749,334447,334145,333844,333542,333242,332941,332641,
332341,332041,331741,331442,331143,330844,330546,330247,
329950,329652,329355,329057,328761,328464,328168,327872,
327576,327280,326985,326690,326395,326101,325807,325513,
325219,324926,324633,324340,324047,323755,323463,323171,
322879,322588,322297,322006,321716,321426,321136,320846,
320557,320267,319978,319690,319401,319113,318825,318538,
318250,317963,317676,317390,317103,316817,316532,316246,
315961,315676,315391,315106,314822,314538,314254,313971,
313688,313405,313122,312839,312557,312275,311994,311712,
311431,311150,310869,310589,310309,310029,309749,309470,
309190,308911,308633,308354,308076,307798,307521,307243,
306966,306689,306412,306136,305860,305584,305308,305033,
304758,304483,304208,303934,303659,303385,303112,302838,
302565,302292,302019,301747,301475,301203,300931,300660,
300388,300117,299847,299576,299306,299036,298766,298497,
298227,297958,297689,297421,297153,296884,296617,296349,
296082,295815,295548,295281,295015,294749,294483,294217,
293952,293686,293421,293157,292892,292628,292364,292100,
291837,291574,291311,291048,290785,290523,290261,289999,
289737,289476,289215,288954,288693,288433,288173,287913,
287653,287393,287134,286875,286616,286358,286099,285841,
285583,285326,285068,284811,284554,284298,284041,283785,
283529,283273,283017,282762,282507,282252,281998,281743,
281489,281235,280981,280728,280475,280222,279969,279716,
279464,279212,278960,278708,278457,278206,277955,277704,
277453,277203,276953,276703,276453,276204,275955,275706,
275457,275209,274960,274712,274465,274217,273970,273722,
273476,273229,272982,272736,272490,272244,271999,271753,
271508,271263,271018,270774,270530,270286,270042,269798,
269555,269312,269069,268826,268583,268341,268099,267857
};
static const int8_t ft2VibratoTable[256] =
{
0,-2,-3,-5,-6,-8,-9,-11,-12,-14,-16,-17,-19,-20,-22,-23,
-24,-26,-27,-29,-30,-32,-33,-34,-36,-37,-38,-39,-41,-42,
-43,-44,-45,-46,-47,-48,-49,-50,-51,-52,-53,-54,-55,-56,
-56,-57,-58,-59,-59,-60,-60,-61,-61,-62,-62,-62,-63,-63,
-63,-64,-64,-64,-64,-64,-64,-64,-64,-64,-64,-64,-63,-63,
-63,-62,-62,-62,-61,-61,-60,-60,-59,-59,-58,-57,-56,-56,
-55,-54,-53,-52,-51,-50,-49,-48,-47,-46,-45,-44,-43,-42,
-41,-39,-38,-37,-36,-34,-33,-32,-30,-29,-27,-26,-24,-23,
-22,-20,-19,-17,-16,-14,-12,-11,-9,-8,-6,-5,-3,-2,0,
2,3,5,6,8,9,11,12,14,16,17,19,20,22,23,24,26,27,29,30,
32,33,34,36,37,38,39,41,42,43,44,45,46,47,48,49,50,51,
52,53,54,55,56,56,57,58,59,59,60,60,61,61,62,62,62,63,
63,63,64,64,64,64,64,64,64,64,64,64,64,63,63,63,62,62,
62,61,61,60,60,59,59,58,57,56,56,55,54,53,52,51,50,49,
48,47,46,45,44,43,42,41,39,38,37,36,34,33,32,30,29,27,
26,24,23,22,20,19,17,16,14,12,11,9,8,6,5,3,2
};
static const DWORD FineLinearSlideUpTable[16] =
{
65536, 65595, 65654, 65714, 65773, 65832, 65892, 65951,
66011, 66071, 66130, 66190, 66250, 66309, 66369, 66429
};
static const DWORD FineLinearSlideDownTable[16] =
{
65535, 65477, 65418, 65359, 65300, 65241, 65182, 65123,
65065, 65006, 64947, 64888, 64830, 64772, 64713, 64645
};
static const DWORD LinearSlideUpTable[256] =
{
65536, 65773, 66010, 66249, 66489, 66729, 66971, 67213,
67456, 67700, 67945, 68190, 68437, 68685, 68933, 69182,
69432, 69684, 69936, 70189, 70442, 70697, 70953, 71209,
71467, 71725, 71985, 72245, 72507, 72769, 73032, 73296,
73561, 73827, 74094, 74362, 74631, 74901, 75172, 75444,
75717, 75991, 76265, 76541, 76818, 77096, 77375, 77655,
77935, 78217, 78500, 78784, 79069, 79355, 79642, 79930,
80219, 80509, 80800, 81093, 81386, 81680, 81976, 82272,
82570, 82868, 83168, 83469, 83771, 84074, 84378, 84683,
84989, 85297, 85605, 85915, 86225, 86537, 86850, 87164,
87480, 87796, 88113, 88432, 88752, 89073, 89395, 89718,
90043, 90369, 90695, 91023, 91353, 91683, 92015, 92347,
92681, 93017, 93353, 93691, 94029, 94370, 94711, 95053,
95397, 95742, 96088, 96436, 96785, 97135, 97486, 97839,
98193, 98548, 98904, 99262, 99621, 99981, 100343, 100706,
101070, 101435, 101802, 102170, 102540, 102911, 103283, 103657,
104031, 104408, 104785, 105164, 105545, 105926, 106309, 106694,
107080, 107467, 107856, 108246, 108637, 109030, 109425, 109820,
110217, 110616, 111016, 111418, 111821, 112225, 112631, 113038,
113447, 113857, 114269, 114682, 115097, 115514, 115931, 116351,
116771, 117194, 117618, 118043, 118470, 118898, 119328, 119760,
120193, 120628, 121064, 121502, 121941, 122382, 122825, 123269,
123715, 124162, 124611, 125062, 125514, 125968, 126424, 126881,
127340, 127801, 128263, 128727, 129192, 129660, 130129, 130599,
131072, 131546, 132021, 132499, 132978, 133459, 133942, 134426,
134912, 135400, 135890, 136381, 136875, 137370, 137866, 138365,
138865, 139368, 139872, 140378, 140885, 141395, 141906, 142419,
142935, 143451, 143970, 144491, 145014, 145538, 146064, 146593,
147123, 147655, 148189, 148725, 149263, 149803, 150344, 150888,
151434, 151982, 152531, 153083, 153637, 154192, 154750, 155310,
155871, 156435, 157001, 157569, 158138, 158710, 159284, 159860,
160439, 161019, 161601, 162186, 162772, 163361, 163952, 164545,
};
static const DWORD LinearSlideDownTable[256] =
{
65536, 65299, 65064, 64830, 64596, 64363, 64131, 63900,
63670, 63440, 63212, 62984, 62757, 62531, 62305, 62081,
61857, 61634, 61412, 61191, 60970, 60751, 60532, 60314,
60096, 59880, 59664, 59449, 59235, 59021, 58809, 58597,
58385, 58175, 57965, 57757, 57548, 57341, 57134, 56928,
56723, 56519, 56315, 56112, 55910, 55709, 55508, 55308,
55108, 54910, 54712, 54515, 54318, 54123, 53928, 53733,
53540, 53347, 53154, 52963, 52772, 52582, 52392, 52204,
52015, 51828, 51641, 51455, 51270, 51085, 50901, 50717,
50535, 50353, 50171, 49990, 49810, 49631, 49452, 49274,
49096, 48919, 48743, 48567, 48392, 48218, 48044, 47871,
47698, 47526, 47355, 47185, 47014, 46845, 46676, 46508,
46340, 46173, 46007, 45841, 45676, 45511, 45347, 45184,
45021, 44859, 44697, 44536, 44376, 44216, 44056, 43898,
43740, 43582, 43425, 43268, 43112, 42957, 42802, 42648,
42494, 42341, 42189, 42037, 41885, 41734, 41584, 41434,
41285, 41136, 40988, 40840, 40693, 40546, 40400, 40254,
40109, 39965, 39821, 39677, 39534, 39392, 39250, 39108,
38967, 38827, 38687, 38548, 38409, 38270, 38132, 37995,
37858, 37722, 37586, 37450, 37315, 37181, 37047, 36913,
36780, 36648, 36516, 36384, 36253, 36122, 35992, 35862,
35733, 35604, 35476, 35348, 35221, 35094, 34968, 34842,
34716, 34591, 34466, 34342, 34218, 34095, 33972, 33850,
33728, 33606, 33485, 33364, 33244, 33124, 33005, 32886,
32768, 32649, 32532, 32415, 32298, 32181, 32065, 31950,
31835, 31720, 31606, 31492, 31378, 31265, 31152, 31040,
30928, 30817, 30706, 30595, 30485, 30375, 30266, 30157,
30048, 29940, 29832, 29724, 29617, 29510, 29404, 29298,
29192, 29087, 28982, 28878, 28774, 28670, 28567, 28464,
28361, 28259, 28157, 28056, 27955, 27854, 27754, 27654,
27554, 27455, 27356, 27257, 27159, 27061, 26964, 26866,
26770, 26673, 26577, 26481, 26386, 26291, 26196, 26102,
};
static const int SpectrumSinusTable[256*2] =
{
0, 1, 1, 2, 3, 3, 4, 5, 6, 7, 7, 8, 9, 10, 10, 11,
12, 13, 14, 14, 15, 16, 17, 17, 18, 19, 20, 20, 21, 22, 22, 23,
24, 25, 25, 26, 27, 28, 28, 29, 30, 30, 31, 32, 32, 33, 34, 34,
35, 36, 36, 37, 38, 38, 39, 39, 40, 41, 41, 42, 42, 43, 44, 44,
45, 45, 46, 46, 47, 47, 48, 48, 49, 49, 50, 50, 51, 51, 52, 52,
53, 53, 53, 54, 54, 55, 55, 55, 56, 56, 57, 57, 57, 58, 58, 58,
59, 59, 59, 59, 60, 60, 60, 60, 61, 61, 61, 61, 61, 62, 62, 62,
62, 62, 62, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63,
63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 63, 62, 62,
62, 62, 62, 62, 61, 61, 61, 61, 61, 60, 60, 60, 60, 59, 59, 59,
59, 58, 58, 58, 57, 57, 57, 56, 56, 55, 55, 55, 54, 54, 53, 53,
53, 52, 52, 51, 51, 50, 50, 49, 49, 48, 48, 47, 47, 46, 46, 45,
45, 44, 44, 43, 42, 42, 41, 41, 40, 39, 39, 38, 38, 37, 36, 36,
35, 34, 34, 33, 32, 32, 31, 30, 30, 29, 28, 28, 27, 26, 25, 25,
24, 23, 22, 22, 21, 20, 20, 19, 18, 17, 17, 16, 15, 14, 14, 13,
12, 11, 10, 10, 9, 8, 7, 7, 6, 5, 4, 3, 3, 2, 1, 0,
0, -1, -1, -2, -3, -3, -4, -5, -6, -7, -7, -8, -9, -10, -10, -11,
-12, -13, -14, -14, -15, -16, -17, -17, -18, -19, -20, -20, -21, -22, -22, -23,
-24, -25, -25, -26, -27, -28, -28, -29, -30, -30, -31, -32, -32, -33, -34, -34,
-35, -36, -36, -37, -38, -38, -39, -39, -40, -41, -41, -42, -42, -43, -44, -44,
-45, -45, -46, -46, -47, -47, -48, -48, -49, -49, -50, -50, -51, -51, -52, -52,
-53, -53, -53, -54, -54, -55, -55, -55, -56, -56, -57, -57, -57, -58, -58, -58,
-59, -59, -59, -59, -60, -60, -60, -60, -61, -61, -61, -61, -61, -62, -62, -62,
-62, -62, -62, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63,
-63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -63, -62, -62,
-62, -62, -62, -62, -61, -61, -61, -61, -61, -60, -60, -60, -60, -59, -59, -59,
-59, -58, -58, -58, -57, -57, -57, -56, -56, -55, -55, -55, -54, -54, -53, -53,
-53, -52, -52, -51, -51, -50, -50, -49, -49, -48, -48, -47, -47, -46, -46, -45,
-45, -44, -44, -43, -42, -42, -41, -41, -40, -39, -39, -38, -38, -37, -36, -36,
-35, -34, -34, -33, -32, -32, -31, -30, -30, -29, -28, -28, -27, -26, -25, -25,
-24, -23, -22, -22, -21, -20, -20, -19, -18, -17, -17, -16, -15, -14, -14, -13,
-12, -11, -10, -10, -9, -8, -7, -7, -6, -5, -4, -3, -3, -2, -1, 0,
};

@ -1,117 +0,0 @@
CC0 1.0 Universal
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File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

@ -96,6 +96,12 @@ struct OpusCustomDecoder {
/* opus_val16 backgroundLogE[], Size = 2*mode->nbEBands */
};
int celt_decoder_get_size(int channels)
{
const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
return opus_custom_decoder_get_size(mode, channels);
}
OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int channels)
{
int size = sizeof(struct CELTDecoder)
@ -105,12 +111,6 @@ OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int
return size;
}
int celt_decoder_get_size(int channels)
{
const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
return opus_custom_decoder_get_size(mode, channels);
}
#ifdef CUSTOM_MODES
CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int *error)
{
@ -128,6 +128,18 @@ CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int
}
#endif /* CUSTOM_MODES */
int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels)
{
int ret;
ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels);
if (ret != OPUS_OK)
return ret;
st->downsample = resampling_factor(sampling_rate);
if (st->downsample==0)
return OPUS_BAD_ARG;
else
return OPUS_OK;
}
OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMode *mode, int channels)
{
@ -156,20 +168,6 @@ OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMod
return OPUS_OK;
}
int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels)
{
int ret;
ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels);
if (ret != OPUS_OK)
return ret;
st->downsample = resampling_factor(sampling_rate);
if (st->downsample==0)
return OPUS_BAD_ARG;
else
return OPUS_OK;
}
#ifdef CUSTOM_MODES
void opus_custom_decoder_destroy(CELTDecoder *st)
{

@ -126,6 +126,12 @@ struct OpusCustomEncoder {
/* opus_val16 oldLogE2[], Size = channels*mode->nbEBands */
};
int celt_encoder_get_size(int channels)
{
CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
return opus_custom_encoder_get_size(mode, channels);
}
OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_get_size(const CELTMode *mode, int channels)
{
int size = sizeof(struct CELTEncoder)
@ -137,12 +143,6 @@ OPUS_CUSTOM_NOSTATIC int opus_custom_encoder_get_size(const CELTMode *mode, int
return size;
}
int celt_encoder_get_size(int channels)
{
CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);
return opus_custom_encoder_get_size(mode, channels);
}
#ifdef CUSTOM_MODES
CELTEncoder *opus_custom_encoder_create(const CELTMode *mode, int channels, int *error)
{

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