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libnyquist
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merge with modulesio/master

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pull/35/head
dimitri 8 years ago
parent af0720ba97 b6ad5b49c0
commit 3857ab49e9
147 changed files with 25175 additions and 9763 deletions
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6
examples/src/Main.cpp
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@ -70,11 +70,13 @@ int main(int argc, const char **argv) try
//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"
// 1 + 2 channel wavepack
// 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");
@ -110,7 +112,7 @@ int main(int argc, const char **argv) try
// Resample
std::vector<float> outputBuffer;
outputBuffer.reserve(fileData->samples.size());
linear_resample(44100.0 / 48000.0, fileData->samples, outputBuffer, 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 << "Output Samples: " << outputBuffer.size() << std::endl;

2
include/libnyquist/AudioDecoder.h
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@ -68,7 +68,9 @@ class NyquistIO
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);

51
include/libnyquist/Common.h
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@ -201,6 +201,57 @@ inline std::array<uint8_t, 3> Unpack(uint32_t a)
return output;
}
//////////////////////////
// Resampling Utilities //
//////////////////////////
// This is a naieve implementation of a resampling filter where a lerp is used as a bad low-pass.
// It very far from the ideal case and should be used with caution (or not at all) on signals that matter.
// It is included here to upsample 44.1k to 48k for the purposes of microphone input => Opus, where the the
// nominal frequencies of speech are particularly far from Nyquist.
inline void linear_resample(const double rate, const std::vector<float> & input, std::vector<float> & output, const uint32_t samplesToProcess)
{
double virtualReadIndex = 0;
double a, b, i, sample;
uint32_t n = samplesToProcess - 1;
while (n--)
{
uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
i = virtualReadIndex - readIndex;
a = input[readIndex + 0];
b = input[readIndex + 1];
sample = (1.0 - i) * a + i * b; // linear interpolate
output.push_back(static_cast<float>(sample));
virtualReadIndex += rate;
}
}
inline double sample_hermite_4p_3o(double x, double * y)
{
static double c0, c1, c2, c3;
c0 = y[1];
c1 = (1.0 / 2.0)*(y[2] - y[0]);
c2 = (y[0] - (5.0 / 2.0)*y[1]) + (2.0*y[2] - (1.0 / 2.0)*y[3]);
c3 = (1.0 / 2.0)*(y[3] - y[0]) + (3.0 / 2.0)*(y[1] - y[2]);
return ((c3*x + c2)*x + c1)*x + c0;
}
inline void hermite_resample(const double rate, const std::vector<float> & input, std::vector<float> & output, const uint32_t samplesToProcess)
{
double virtualReadIndex = 1;
double i, sample;
uint32_t n = samplesToProcess - 1;
while (n--)
{
uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
i = virtualReadIndex - readIndex;
double samps[4] = { input[readIndex - 1], input[readIndex], input[readIndex + 1], input[readIndex + 2] };
sample = sample_hermite_4p_3o(i, samps); // cubic hermite interpolate over 4 samples
output.push_back(static_cast<float>(sample));
virtualReadIndex += rate;
}
}
//////////////////////////
// Conversion Utilities //
//////////////////////////

2
include/libnyquist/IMA4Util.h
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@ -100,7 +100,7 @@ namespace nqr
const uint8_t * data = state.inBuffer;
// Loop over the interleaved channels
for (int32_t ch = 0; ch < num_channels; ch++)
for (uint32_t ch = 0; ch < num_channels; ch++)
{
const int byteOffset = ch * 4;

2
include/libnyquist/ModplugDecoder.h
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@ -42,4 +42,4 @@ struct ModplugDecoder : public nqr::BaseDecoder
} // end namespace nqr
#endif
#endif

47
include/libnyquist/WavEncoder.h
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@ -32,53 +32,6 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
namespace nqr
{
// This is a naieve implementation of a resampling filter where a lerp is used as a bad low-pass.
// It very far from the ideal case and should be used with caution (or not at all) on signals that matter.
// It is included here to upsample 44.1k to 48k for the purposes of microphone input => Opus, where the the
// nominal frequencies of speech are particularly far from Nyquist.
static inline void linear_resample(const double rate, const std::vector<float> & input, std::vector<float> & output, size_t samplesToProcess)
{
double virtualReadIndex = 0;
double a, b, i, sample;
uint32_t n = samplesToProcess - 1;
while (n--)
{
uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
i = virtualReadIndex - readIndex;
a = input[readIndex + 0];
b = input[readIndex + 1];
sample = (1.0 - i) * a + i * b; // linear interpolate
output.push_back(sample);
virtualReadIndex += rate;
}
}
static inline double sample_hermite_4p_3o(double x, double * y)
{
static double c0, c1, c2, c3;
c0 = y[1];
c1 = (1.0/2.0)*(y[2]-y[0]);
c2 = (y[0] - (5.0/2.0)*y[1]) + (2.0*y[2] - (1.0/2.0)*y[3]);
c3 = (1.0/2.0)*(y[3]-y[0]) + (3.0/2.0)*(y[1]-y[2]);
return ((c3*x+c2)*x+c1)*x+c0;
}
static inline void hermite_resample(const double rate, const std::vector<float> & input, std::vector<float> & output, size_t samplesToProcess)
{
double virtualReadIndex = 1;
double i, sample;
uint32_t n = samplesToProcess - 1;
while (n--)
{
uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
i = virtualReadIndex - readIndex;
double samps[4] = {input[readIndex - 1], input[readIndex], input[readIndex + 1], input[readIndex + 2]};
sample = sample_hermite_4p_3o(i, samps); // cubic hermite interpolate over 4 samples
output.push_back(sample);
virtualReadIndex += rate;
}
}
enum EncoderError
{
NoError,

3
libnyquist.vcxproj/v140/libnyquist.vcxproj
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@ -36,7 +36,6 @@
<ClCompile Include="..\..\src\ModplugDependencies.cpp" />
<ClCompile Include="..\..\src\MusepackDecoder.cpp" />
<ClCompile Include="..\..\src\MusepackDependencies.c" />
<ClCompile Include="..\..\src\WavPackDependencies.c" />
</ItemGroup>
<ItemGroup>
<ClInclude Include="$(ProjectDir)..\..\include\libnyquist\AudioDecoder.h" />
@ -161,4 +160,4 @@
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
<ImportGroup Label="ExtensionTargets">
</ImportGroup>
</Project>
</Project>

5
libnyquist.vcxproj/v140/libnyquist.vcxproj.filters
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@ -46,9 +46,6 @@
<ClCompile Include="..\..\src\MusepackDecoder.cpp">
<Filter>src</Filter>
</ClCompile>
<ClCompile Include="..\..\src\WavPackDependencies.c">
<Filter>src\deps</Filter>
</ClCompile>
<ClCompile Include="..\..\src\ModplugDependencies.cpp">
<Filter>src\deps</Filter>
</ClCompile>
@ -114,4 +111,4 @@
<UniqueIdentifier>{d839471f-71d1-471e-95e0-c33af9bb64bc}</UniqueIdentifier>
</Filter>
</ItemGroup>
</Project>
</Project>

35
libnyquist.vcxproj/v141/libnyquist.vcxproj
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@ -35,7 +35,32 @@
<ClCompile Include="..\..\src\ModplugDependencies.cpp" />
<ClCompile Include="..\..\src\MusepackDecoder.cpp" />
<ClCompile Include="..\..\src\MusepackDependencies.c" />
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<ClCompile Include="..\..\third_party\wavpack\src\common_utils.c" />
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<ClCompile Include="..\..\third_party\wavpack\src\pack_floats.c" />
<ClCompile Include="..\..\third_party\wavpack\src\pack_utils.c" />
<ClCompile Include="..\..\third_party\wavpack\src\read_words.c" />
<ClCompile Include="..\..\third_party\wavpack\src\tags.c" />
<ClCompile Include="..\..\third_party\wavpack\src\tag_utils.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack3.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack3_open.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack3_seek.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack_dsd.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack_floats.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack_seek.c" />
<ClCompile Include="..\..\third_party\wavpack\src\unpack_utils.c" />
<ClCompile Include="..\..\third_party\wavpack\src\write_words.c" />
</ItemGroup>
<ItemGroup>
<ClInclude Include="$(ProjectDir)..\..\include\libnyquist\AudioDecoder.h" />
@ -106,8 +131,9 @@
<WarningLevel>Level3</WarningLevel>
<Optimization>Disabled</Optimization>
<SDLCheck>true</SDLCheck>
<PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
<PreprocessorDefinitions>DEBUG;_ITERATOR_DEBUG_LEVEL=0;_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
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<RuntimeLibrary>MultiThreadedDebug</RuntimeLibrary>
</ClCompile>
<Link>
<GenerateDebugInformation>true</GenerateDebugInformation>
@ -118,8 +144,9 @@
<WarningLevel>Level3</WarningLevel>
<Optimization>Disabled</Optimization>
<SDLCheck>true</SDLCheck>
<PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
<PreprocessorDefinitions>DEBUG;_ITERATOR_DEBUG_LEVEL=0;_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
<AdditionalIncludeDirectories>$(ProjectDir)..\..\third_party\;$(ProjectDir)..\..\include\libnyquist\;$(ProjectDir)..\..\third_party\libvorbis\include;$(ProjectDir)..\..\third_party\libogg\include;$(ProjectDir)..\..\third_party\wavpack\include;$(ProjectDir)..\..\third_party\flac\src\include;$(ProjectDir)..\..\third_party\opus\celt;$(ProjectDir)..\..\third_party\opus\libopus\include;$(ProjectDir)..\..\third_party\opus\libopus\src;$(ProjectDir)..\..\third_party\opus\opusfile\include;$(ProjectDir)..\..\third_party\opus\opusfile\src;$(ProjectDir)..\..\third_party\opus\opusfile\src\include;$(ProjectDir)..\..\third_party\opus\silk;$(ProjectDir)..\..\third_party\opus\silk\float;$(ProjectDir)..\..\third_party\musepack\include;$(ProjectDir)..\..\third_party\musepack\libmpcenc;$(ProjectDir)..\..\third_party\musepack\libmpcdec;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<RuntimeLibrary>MultiThreadedDebugDLL</RuntimeLibrary>
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<Link>
<GenerateDebugInformation>true</GenerateDebugInformation>
@ -134,6 +161,7 @@
<SDLCheck>true</SDLCheck>
<PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
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<RuntimeLibrary>MultiThreaded</RuntimeLibrary>
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<Link>
<GenerateDebugInformation>true</GenerateDebugInformation>
@ -150,6 +178,7 @@
<SDLCheck>true</SDLCheck>
<PreprocessorDefinitions>_CRT_SECURE_NO_WARNINGS;_MBCS;D_SCL_SECURE_NO_WARNINGS;_SCL_SECURE_NO_WARNING;WIN32;_WIN32;USE_ALLOCA;OPUS_BUILD;%(PreprocessorDefinitions)</PreprocessorDefinitions>
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<RuntimeLibrary>MultiThreadedDLL</RuntimeLibrary>
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<Link>
<GenerateDebugInformation>true</GenerateDebugInformation>

90
libnyquist.vcxproj/v141/libnyquist.vcxproj.filters
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@ -43,20 +43,92 @@
<ClCompile Include="..\..\src\MusepackDecoder.cpp">
<Filter>src</Filter>
</ClCompile>
<ClCompile Include="..\..\src\WavPackDependencies.c">
<Filter>src\deps</Filter>
</ClCompile>
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<Filter>src\deps</Filter>
</ClCompile>
<ClCompile Include="..\..\src\ModplugDecoder.cpp">
<Filter>src</Filter>
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<Filter>src</Filter>
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<Filter>include</Filter>
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@ -93,6 +165,9 @@
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<Filter>include\util</Filter>
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<Filter>include</Filter>
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<Filter Include="include">
@ -107,5 +182,8 @@
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4
libnyquist.xcodeproj/project.pbxproj
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@ -22,7 +22,6 @@
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@ -72,7 +71,6 @@
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@ -92,7 +90,6 @@
isa = PBXGroup;
children = (
08FFC72C1CA702EC005812D6 /* ModplugDependencies.cpp */,
08D0EC741C6DA41300FCDA23 /* WavPackDependencies.c */,
081FFB181ADF803800673073 /* FlacDependencies.c */,
086DADAB1ADF9DF30031F793 /* VorbisDependencies.c */,
0804D13E1AE69F0100F4B1FD /* OpusDependencies.c */,
@ -264,7 +261,6 @@
08FFC72D1CA702EC005812D6 /* ModplugDependencies.cpp in Sources */,
08B91D9E1AC73B8A00335131 /* FlacDecoder.cpp in Sources */,
08FFC72F1CA7038D005812D6 /* ModplugDecoder.cpp in Sources */,
08D0EC751C6DA41300FCDA23 /* WavPackDependencies.c in Sources */,
086DADAD1AE029860031F793 /* VorbisDependencies.c in Sources */,
08B91DA31AC73B8A00335131 /* WavPackDecoder.cpp in Sources */,
08B91D9D1AC73B8A00335131 /* Common.cpp in Sources */,

2
src/AudioDecoder.cpp
Unescape Escape View File

@ -134,4 +134,4 @@ void NyquistIO::BuildDecoderTable()
AddDecoderToTable(std::make_shared<OpusDecoder>());
AddDecoderToTable(std::make_shared<MusepackDecoder>());
AddDecoderToTable(std::make_shared<ModplugDecoder>());
}
}

6
src/Common.cpp
Unescape Escape View File

@ -24,6 +24,7 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "Common.h"
#include <cstring>
using namespace nqr;
@ -104,9 +105,10 @@ void nqr::ConvertToFloat32(float * dst, const uint8_t * src, const size_t N, PCM
else if (f == PCM_FLT)
{
const float * dataPtr = reinterpret_cast<const float *>(src);
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]);
dst[i] = (float) Read32(dataPtr[i]); */
}
else if (f == PCM_DBL)
{

126
src/FlacDecoder.cpp
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@ -24,26 +24,23 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "FlacDecoder.h"
#include "flac/all.h"
#include "flac/stream_decoder.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:
// FLAC is a big-endian format. All values are unsigned.
FlacDecoderInternal(AudioData * d, std::string filepath) : d(d)
FlacDecoderInternal(AudioData * d, const std::string & filepath) : d(d)
{
/////////////////////////////
// Initialize FLAC library //
/////////////////////////////
decoderInternal = FLAC__stream_decoder_new();
FLAC__stream_decoder_set_metadata_respond(decoderInternal, FLAC__METADATA_TYPE_STREAMINFO);
@ -57,10 +54,49 @@ public:
this) == FLAC__STREAM_DECODER_INIT_STATUS_OK;
FLAC__stream_decoder_set_md5_checking(decoderInternal, true);
if (initialized)
{
// Find the size and allocate memory
FLAC__stream_decoder_process_until_end_of_metadata(decoderInternal);
// Read memory out into our temporary internalBuffer
FLAC__stream_decoder_process_until_end_of_stream(decoderInternal);
// Presently unneeded, but useful for reference
// FLAC__ChannelAssignment channelAssignment = FLAC__stream_decoder_get_channel_assignment(decoderInternal);
// Fill out remaining user data
d->lengthSeconds = (float) numSamples / (float) d->sampleRate;
auto totalSamples = numSamples * d->channelCount;
// Next, process internal buffer into the user-visible samples array
ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples, d->sourceFormat);
}
else throw std::runtime_error("Unable to initialize FLAC decoder");
}
FlacDecoderInternal(AudioData * d, const std::vector<uint8_t> & memory) : d(d), data(std::move(memory)), dataPos(0)
{
decoderInternal = FLAC__stream_decoder_new();
//////////////////////
// Read Stream Data //
/////////////////////
FLAC__stream_decoder_set_metadata_respond(decoderInternal, FLAC__METADATA_TYPE_STREAMINFO);
bool initialized = FLAC__stream_decoder_init_stream(
decoderInternal,
read_callback,
seek_callback,
tell_callback,
length_callback,
eof_callback,
s_writeCallback,
s_metadataCallback,
s_errorCallback,
this
) == FLAC__STREAM_DECODER_INIT_STATUS_OK;
FLAC__stream_decoder_set_md5_checking(decoderInternal, true);
if (initialized)
{
@ -81,12 +117,7 @@ public:
// Next, process internal buffer into the user-visible samples array
ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples, d->sourceFormat);
}
else
{
throw std::runtime_error("Unable to initialize FLAC decoder");
}
else throw std::runtime_error("Unable to initialize FLAC decoder");
}
~FlacDecoderInternal()
@ -121,16 +152,14 @@ public:
static FLAC__StreamDecoderWriteStatus s_writeCallback(const FLAC__StreamDecoder *, const FLAC__Frame* frame, const FLAC__int32 * const buffer[], void * userPtr)
{
FlacDecoderInternal * decoder = reinterpret_cast<FlacDecoderInternal *>(userPtr);
const size_t bytesPerSample = GetFormatBitsPerSample(decoder->d->sourceFormat) / 8;
auto dataPtr = decoder->internalBuffer.data();
for (uint32_t i = 0; i < frame->header.blocksize; i++)
{
for(int j = 0; j < decoder->d->channelCount; j++)
for (int j = 0; j < decoder->d->channelCount; j++)
{
memcpy(dataPtr + decoder->bufferPosition, &buffer[j][i], bytesPerSample);
std::memcpy(dataPtr + decoder->bufferPosition, &buffer[j][i], bytesPerSample);
decoder->bufferPosition += bytesPerSample;
}
}
@ -145,7 +174,51 @@ public:
static void s_errorCallback (const FLAC__StreamDecoder *, FLAC__StreamDecoderErrorStatus status, void *)
{
std::cerr << "FLAC Decoder Error: " << FLAC__StreamDecoderErrorStatusString[status] << std::endl;
throw std::runtime_error("FLAC decode exception " + std::string(FLAC__StreamDecoderErrorStatusString[status]));
}
static FLAC__StreamDecoderReadStatus read_callback(const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], size_t *bytes, void *client_data)
{
FlacDecoderInternal *decoderInternal = (FlacDecoderInternal *)client_data;
size_t readLength = std::min<size_t>(*bytes, decoderInternal->data.size() - decoderInternal->dataPos);
if (readLength > 0)
{
std::memcpy(buffer, decoderInternal->data.data(), readLength);
decoderInternal->dataPos += readLength;
*bytes = readLength;
if (decoderInternal->dataPos < decoderInternal->data.size()) return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
else return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
}
else return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
}
static FLAC__StreamDecoderSeekStatus seek_callback(const FLAC__StreamDecoder *decoder, FLAC__uint64 absolute_byte_offset, void *client_data)
{
FlacDecoderInternal *decoderInternal = (FlacDecoderInternal *)client_data;
size_t newPos = std::min<size_t>(absolute_byte_offset, decoderInternal->data.size() - decoderInternal->dataPos);
decoderInternal->dataPos = newPos;
return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
}
static FLAC__StreamDecoderTellStatus tell_callback(const FLAC__StreamDecoder *decoder, FLAC__uint64 *absolute_byte_offset, void *client_data)
{
FlacDecoderInternal *decoderInternal = (FlacDecoderInternal *)client_data;
*absolute_byte_offset = decoderInternal->dataPos;
return FLAC__STREAM_DECODER_TELL_STATUS_OK;
}
static FLAC__StreamDecoderLengthStatus length_callback(const FLAC__StreamDecoder *decoder, FLAC__uint64 *stream_length, void *client_data)
{
FlacDecoderInternal *decoderInternal = (FlacDecoderInternal *)client_data;
*stream_length = decoderInternal->data.size();
return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
}
static FLAC__bool eof_callback(const FLAC__StreamDecoder *decoder, void *client_data)
{
FlacDecoderInternal *decoderInternal = (FlacDecoderInternal *)client_data;
return decoderInternal->dataPos == decoderInternal->data.size();
}
private:
@ -153,7 +226,8 @@ private:
NO_COPY(FlacDecoderInternal);
FLAC__StreamDecoder * decoderInternal;
std::vector<uint8_t> data;
size_t dataPos;
size_t bufferPosition = 0;
size_t numSamples = 0;
@ -173,10 +247,10 @@ void FlacDecoder::LoadFromPath(AudioData * data, const std::string & path)
void FlacDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & memory)
{
throw LoadBufferNotImplEx();
FlacDecoderInternal decoder(data, memory);
}
std::vector<std::string> FlacDecoder::GetSupportedFileExtensions()
{
return {"flac"};
}
}

32
src/FlacDependencies.c
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@ -77,25 +77,25 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#define HAVE_LROUND 1
#include "flac/all.h"
#include "FLAC/all.h"
#if defined(_MSC_VER)
#include "flac/src/win_utf8_io.c"
#include "FLAC/src/win_utf8_io.c"
#endif
#include "flac/src/bitmath.c"
#include "flac/src/bitreader.c"
#include "flac/src/bitwriter.c"
#include "flac/src/cpu.c"
#include "flac/src/crc.c"
#include "flac/src/fixed.c"
#include "flac/src/float.c"
#include "flac/src/format.c"
#include "flac/src/lpc.c"
#include "flac/src/md5.c"
#include "flac/src/memory.c"
#include "flac/src/stream_decoder.c"
#include "flac/src/window.c"
#include "FLAC/src/bitmath.c"
#include "FLAC/src/bitreader.c"
#include "FLAC/src/bitwriter.c"
#include "FLAC/src/cpu.c"
#include "FLAC/src/crc.c"
#include "FLAC/src/fixed.c"
#include "FLAC/src/float.c"
#include "FLAC/src/format.c"
#include "FLAC/src/lpc.c"
#include "FLAC/src/md5.c"
#include "FLAC/src/memory.c"
#include "FLAC/src/stream_decoder.c"
#include "FLAC/src/window.c"
#undef VERSION
@ -105,4 +105,4 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#if (_MSC_VER)
#pragma warning (pop)
#endif
#endif

47
src/ModplugDecoder.cpp
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@ -53,10 +53,7 @@ public:
ModPlug_SetSettings(&mps);
mpf = ModPlug_Load((const void*)fileData.data(), fileData.size());
if (!mpf)
{
throw std::runtime_error("could not load module");
}
if (!mpf) throw std::runtime_error("could not load module");
d->sampleRate = 44100;
d->channelCount = 2;
@ -68,43 +65,46 @@ public:
auto totalSamples = (44100LL * len_ms) / 1000;
d->samples.resize(totalSamples * d->channelCount);
auto readInternal = [&]()
auto read_func = [&]()
{
const float invf = 1 / (float)0x7fffffff;
float *ptr = d->samples.data();
float *end = d->samples.data() + d->samples.size();
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);
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;
}
if (totalSamples < samples_read)
{
samples_read = totalSamples;
}
for( int i = 0; i < samples_read; ++i ) {
*ptr++ = *((int*)ptr) * invf;
*ptr++ = *((int*)ptr) * invf;
}
for (int i = 0; i < samples_read; ++i)
{
*ptr++ = *((int*)ptr) * invf;
*ptr++ = *((int*)ptr) * invf;
}
totalSamples -= samples_read;
}
totalSamples -= samples_read;
}
return ptr >= end;
};
if (!readInternal())
if (!read_func())
{
throw std::runtime_error("could not read any data");
}
ModPlug_Unload(mpf);
}
private:
ModPlugFile* mpf;
ModPlugFile * mpf;
NO_MOVE(ModplugInternal);
AudioData * d;
};
@ -128,4 +128,3 @@ 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"};
}

6
src/MusepackDecoder.cpp
Unescape Escape View File

@ -110,9 +110,7 @@ public:
reader.tell = tell_mem;
mpcDemux = mpc_demux_init(&reader);
if (!mpcDemux)
throw std::runtime_error("could not initialize mpc demuxer");
if (!mpcDemux) throw std::runtime_error("could not initialize mpc demuxer");
mpc_demux_get_info(mpcDemux, &streamInfo);
@ -190,4 +188,4 @@ void MusepackDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t
std::vector<std::string> MusepackDecoder::GetSupportedFileExtensions()
{
return {"mpc", "mpp"};
}
}

2
src/OpusDecoder.cpp
Unescape Escape View File

@ -174,4 +174,4 @@ void nqr::OpusDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_
std::vector<std::string> nqr::OpusDecoder::GetSupportedFileExtensions()
{
return {"opus"};
}
}

3
src/WavDecoder.cpp
Unescape Escape View File

@ -26,6 +26,7 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include "WavDecoder.h"
#include "RiffUtils.h"
#include "IMA4Util.h"
#include <cstring>
using namespace nqr;
@ -200,7 +201,7 @@ void WavDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & m
uint32_t frameOffset = 0;
uint32_t frameCount = DataChunkInfo.size / s.frame_size;
for (int i = 0; i < frameCount; ++i)
for (uint32_t i = 0; i < frameCount; ++i)
{
decode_ima_adpcm(s, adpcm_pcm16.data() + frameOffset, wavHeader.channel_count);
s.inBuffer += s.frame_size;

7
src/WavEncoder.cpp
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@ -79,28 +79,25 @@ int WavEncoder::WriteFile(const EncoderParams p, const AudioData * d, const std:
sampleDataOptionalMix.resize(sampleDataSize * 2);
MonoToStereo(sampleData, sampleDataOptionalMix.data(), sampleDataSize); // Mix
// Re-point data
// Re-point data
sampleData = sampleDataOptionalMix.data();
sampleDataSize = sampleDataOptionalMix.size();
}
// Stereo => Mono
else if (d->channelCount == 2 && p.channelCount == 1)
{
sampleDataOptionalMix.resize(sampleDataSize / 2);
StereoToMono(sampleData, sampleDataOptionalMix.data(), sampleDataSize); // Mix
// Re-point data
// Re-point data
sampleData = sampleDataOptionalMix.data();
sampleDataSize = sampleDataOptionalMix.size();
}
else if (d->channelCount == p.channelCount)
{
// No op
}
else
{
return EncoderError::UnsupportedChannelMix;

182
src/WavPackDecoder.cpp
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@ -25,6 +25,7 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include "WavPackDecoder.h"
#include "wavpack.h"
#include <string.h>
using namespace nqr;
@ -33,11 +34,67 @@ class WavPackInternal
public:
WavPackInternal(AudioData * d, const std::string path) : d(d)
WavPackInternal(AudioData * d, const std::string & path) : d(d)
{
char errorStr[128];
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);
/// 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), 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 = WavpackOpenFileInputEx64(&reader, this, nullptr, errorStr, OPEN_WVC | OPEN_NORMALIZE, 0);
if (!context)
{
throw std::runtime_error("Not a WavPack file");
@ -60,11 +117,6 @@ public:
d->sourceFormat = MakeFormatForBits(bitdepth, isFloatingPoint, false);
/* From the 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)
@ -89,8 +141,6 @@ public:
size_t framesRemaining = requestedFrameCount;
size_t totalFramesRead = 0;
// int frameSize = d->channelCount * WavpackGetBitsPerSample(context);
// The samples returned are handled differently based on the file's mode
int mode = WavpackGetMode(context);
@ -111,8 +161,7 @@ public:
}
// EOF
//if (framesRead == 0)
// break;
//if (framesRead == 0) break;
totalFramesRead += framesRead;
framesRemaining -= framesRead;
@ -120,6 +169,115 @@ public:
return totalFramesRead;
}
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;
}
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 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:
@ -130,6 +288,8 @@ private:
WavpackContext * context; //@todo unique_ptr
AudioData * d;
std::vector<uint8_t> data;
size_t dataPos;
std::vector<int32_t> internalBuffer;
@ -149,7 +309,7 @@ void WavPackDecoder::LoadFromPath(AudioData * data, const std::string & path)
void WavPackDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & memory)
{
throw LoadBufferNotImplEx();
WavPackInternal decoder(data, memory);
}
std::vector<std::string> WavPackDecoder::GetSupportedFileExtensions()

68
src/WavPackDependencies.c
Unescape Escape View File

@ -1,68 +0,0 @@
/*
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.
*/
#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 4703)
#endif
#ifdef __clang__
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wconversion"
#pragma clang diagnostic ignored "-Wshadow"
#pragma clang diagnostic ignored "-Wdeprecated-register"
#endif
#ifdef _WIN32
#ifndef WIN32
#define WIN32
#endif
#endif
#include "wavpack/src/bits.c"
#include "wavpack/src/extra1.c"
#define WavpackExtraInfo WavpackExtraInfo_alt
#define log2overhead log2overhead_alt
#define xtable xtable_alt
#include "wavpack/src/extra2.c"
#include "wavpack/src/float.c"
#include "wavpack/src/metadata.c"
#define decorr_stereo_pass decorr_stereo_pass_alt
#include "wavpack/src/pack.c"
#include "wavpack/src/tags.c"
#undef decorr_stereo_pass
#define decorr_stereo_pass decorr_stereo_pass_alt_2
#include "wavpack/src/unpack.c"
#include "wavpack/src/unpack3.c"
#include "wavpack/src/words.c"
#include "wavpack/src/wputils.c"
#ifdef __clang__
#pragma clang diagnostic pop
#endif
#if (_MSC_VER)
#pragma warning (pop)
#endif

0
third_party/flac/all.h → third_party/FLAC/all.h vendored
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0
third_party/flac/assert.h → third_party/FLAC/assert.h vendored
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0
third_party/flac/callback.h → third_party/FLAC/callback.h vendored
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0
third_party/flac/export.h → third_party/FLAC/export.h vendored
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0
third_party/flac/format.h → third_party/FLAC/format.h vendored
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@ -98,12 +98,10 @@ 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*2];
extern int MixSoundBuffer[MIXBUFFERSIZE*4];
//cextern int MixReverbBuffer[MIXBUFFERSIZE*2];
extern int MixReverbBuffer[MIXBUFFERSIZE*2];
*/
static UINT GetMaskFromSize(UINT len)
//-----------------------------------

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