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audiodecodermediafoundation.cpp
761 lines (676 loc) · 26.9 KB
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audiodecodermediafoundation.cpp
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/*
* libaudiodecoder - Native Portable Audio Decoder Library
* libaudiodecoder API Header File
* Latest version available at: http://www.oscillicious.com/libaudiodecoder
*
* Copyright (c) 2010-2012 Albert Santoni, Bill Good, RJ Ryan
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire libaudiodecoder license; however,
* the Oscillicious community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
/**
* \file soundsourcemediafoundation.cpp
* \author Bill Good <bkgood at gmail dot com>
* \author Albert Santoni <alberts at mixxx dot org>
* \date Jan 10, 2011
* \note This file uses COM interfaces defined in Windows 7 and later added to
* Vista and Server 2008 via the "Platform Update Supplement for Windows Vista
* and for Windows Server 2008" (http://support.microsoft.com/kb/2117917).
* Earlier versions of Vista (and possibly Server 2008) have some Media
* Foundation interfaces but not the required IMFSourceReader, and are missing
* the Microsoft-provided AAC decoder. XP does not include Media Foundation.
*/
#include <iostream>
#include <string.h>
#include <windows.h>
#include <mfapi.h>
#include <mfidl.h>
#include <mfreadwrite.h>
#include <mferror.h>
#include <propvarutil.h>
#include <assert.h>
#include "audiodecodermediafoundation.h"
const int kBitsPerSample = 16;
const int kNumChannels = 2;
const int kSampleRate = 44100;
const int kLeftoverSize = 4096; // in int16's, this seems to be the size MF AAC
// decoder likes to give
const static bool sDebug = false;
/** Microsoft examples use this snippet often. */
template<class T> static void safeRelease(T **ppT)
{
if (*ppT) {
(*ppT)->Release();
*ppT = NULL;
}
}
std::wstring s2ws(const std::string& s)
{
int len;
int slength = (int)s.length() + 1;
len = MultiByteToWideChar(CP_ACP, 0, s.c_str(), slength, 0, 0);
wchar_t* buf = new wchar_t[len];
MultiByteToWideChar(CP_ACP, 0, s.c_str(), slength, buf, len);
std::wstring r(buf);
delete[] buf;
return r;
}
AudioDecoderMediaFoundation::AudioDecoderMediaFoundation(const std::string filename)
: AudioDecoderBase(filename)
, m_pReader(NULL)
, m_pAudioType(NULL)
, m_wcFilename(NULL)
, m_nextFrame(0)
, m_leftoverBuffer(NULL)
, m_leftoverBufferSize(0)
, m_leftoverBufferLength(0)
, m_leftoverBufferPosition(0)
, m_mfDuration(0)
, m_iCurrentPosition(0)
, m_dead(false)
, m_seeking(false)
{
//Defaults
m_iChannels = kNumChannels;
m_iSampleRate = kSampleRate;
m_iBitsPerSample = kBitsPerSample;
// http://social.msdn.microsoft.com/Forums/en/netfxbcl/thread/35c6a451-3507-40c8-9d1c-8d4edde7c0cc
// gives maximum path + file length as 248 + 260, using that -bkgood
m_wcFilename = new wchar_t[248 + 260];
}
AudioDecoderMediaFoundation::~AudioDecoderMediaFoundation()
{
delete [] m_wcFilename;
delete [] m_leftoverBuffer;
safeRelease(&m_pReader);
safeRelease(&m_pAudioType);
MFShutdown();
CoUninitialize();
}
int AudioDecoderMediaFoundation::open()
{
if (sDebug) {
std::cout << "open() " << m_filename << std::endl;
}
//Assumes m_filename is ASCII and converts it to UTF-16 (wide char).
/*
int wcFilenameLength = m_filename.size();
for(std::wstring::size_type i=0; i < m_filename.size(); ++i)
{
m_wcFilename[i] = m_filename[i];
}
m_wcFilename[wcFilenameLength] = (wchar_t)'\0';
std::string s;
std::wstring stemp = s2ws(m_filename); // Temporary buffer is required
LPCWSTR result = (LPCWSTR)stemp.c_str();
*/
//LPCWSTR result;
const char* utf8Str = m_filename.c_str();
MultiByteToWideChar(CP_UTF8,
0,
utf8Str,
-1, //assume utf8Str is NULL terminated and give us back a NULL terminated string
(LPWSTR)m_wcFilename,
512);
LPCWSTR result = m_wcFilename;
HRESULT hr(S_OK);
// Initialize the COM library.
hr = CoInitializeEx(NULL, COINIT_APARTMENTTHREADED | COINIT_DISABLE_OLE1DDE);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to initialize COM" << std::endl;
return AUDIODECODER_ERROR;
}
// Initialize the Media Foundation platform.
hr = MFStartup(MF_VERSION);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to initialize Media Foundation" << std::endl;
return AUDIODECODER_ERROR;
}
// Create the source reader to read the input file.
hr = MFCreateSourceReaderFromURL(/*m_wcFilename*/result, NULL, &m_pReader);
if (FAILED(hr)) {
std::cerr << "SSMF: Error opening input file:" << m_filename << ", with error: " << HRESULT_CODE(hr) << std::endl;
return AUDIODECODER_ERROR;
}
if (!configureAudioStream()) {
std::cerr << "SSMF: Error configuring audio stream." << std::endl;
return AUDIODECODER_ERROR;
}
if (!readProperties()) {
std::cerr << "SSMF::readProperties failed" << std::endl;
return AUDIODECODER_ERROR;
}
//Seek to position 0, which forces us to skip over all the header frames.
//This makes sure we're ready to just let the Analyser rip and it'll
//get the number of samples it expects (ie. no header frames).
seek(0);
return AUDIODECODER_OK;
}
int AudioDecoderMediaFoundation::seek(int sampleIdx)
{
if (sDebug) { std::cout << "seek() " << sampleIdx << std::endl; }
PROPVARIANT prop;
HRESULT hr(S_OK);
__int64 seekTarget(sampleIdx / m_iChannels);
__int64 mfSeekTarget(mfFromFrame(seekTarget) - 1);
// minus 1 here seems to make our seeking work properly, otherwise we will
// (more often than not, maybe always) seek a bit too far (although not
// enough for our calculatedFrameFromMF <= nextFrame assertion in ::read).
// Has something to do with 100ns MF units being much smaller than most
// frame offsets (in seconds) -bkgood
long result = m_iCurrentPosition;
if (m_dead) {
return result;
}
// this doesn't fail, see MS's implementation
hr = InitPropVariantFromInt64(mfSeekTarget < 0 ? 0 : mfSeekTarget, &prop);
hr = m_pReader->Flush(MF_SOURCE_READER_FIRST_AUDIO_STREAM);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to flush before seek";
}
// http://msdn.microsoft.com/en-us/library/dd374668(v=VS.85).aspx
hr = m_pReader->SetCurrentPosition(GUID_NULL, prop);
if (FAILED(hr)) {
// nothing we can do here as we can't fail (no facility to other than
// crashing mixxx)
std::cerr << "SSMF: failed to seek" << (
hr == MF_E_INVALIDREQUEST ? "Sample requests still pending" : "");
} else {
result = sampleIdx;
}
PropVariantClear(&prop);
// record the next frame so that we can make sure we're there the next
// time we get a buffer from MFSourceReader
m_nextFrame = seekTarget;
m_seeking = true;
m_iCurrentPosition = result;
return result;
}
int AudioDecoderMediaFoundation::read(int size, const SAMPLE *destination)
{
assert(size < sizeof(m_destBufferShort));
if (sDebug) { std::cout << "read() " << size << std::endl; }
//TODO: Change this up if we want to support just short samples again -- Albert
SHORT_SAMPLE *destBuffer = m_destBufferShort;
size_t framesRequested(size / m_iChannels);
size_t framesNeeded(framesRequested);
// first, copy frames from leftover buffer IF the leftover buffer is at
// the correct frame
if (m_leftoverBufferLength > 0 && m_leftoverBufferPosition == m_nextFrame) {
copyFrames(destBuffer, &framesNeeded, m_leftoverBuffer,
m_leftoverBufferLength);
if (m_leftoverBufferLength > 0) {
if (framesNeeded != 0) {
std::cerr << __FILE__ << __LINE__
<< "WARNING: Expected frames needed to be 0. Abandoning this file.";
m_dead = true;
}
m_leftoverBufferPosition += framesRequested;
}
} else {
// leftoverBuffer already empty or in the wrong position, clear it
m_leftoverBufferLength = 0;
}
while (!m_dead && framesNeeded > 0) {
HRESULT hr(S_OK);
DWORD dwFlags(0);
__int64 timestamp(0);
IMFSample *pSample(NULL);
bool error(false); // set to true to break after releasing
hr = m_pReader->ReadSample(
MF_SOURCE_READER_FIRST_AUDIO_STREAM, // [in] DWORD dwStreamIndex,
0, // [in] DWORD dwControlFlags,
NULL, // [out] DWORD *pdwActualStreamIndex,
&dwFlags, // [out] DWORD *pdwStreamFlags,
×tamp, // [out] LONGLONG *pllTimestamp,
&pSample); // [out] IMFSample **ppSample
if (FAILED(hr)) {
if (sDebug) { std::cout << "ReadSample failed." << std::endl; }
break;
}
if (sDebug) {
std::cout << "ReadSample timestamp: " << timestamp
<< "frame: " << frameFromMF(timestamp)
<< "dwflags: " << dwFlags
<< std::endl;
}
if (dwFlags & MF_SOURCE_READERF_ERROR) {
// our source reader is now dead, according to the docs
std::cerr << "SSMF: ReadSample set ERROR, SourceReader is now dead";
m_dead = true;
break;
} else if (dwFlags & MF_SOURCE_READERF_ENDOFSTREAM) {
std::cout << "SSMF: End of input file." << std::endl;
break;
} else if (dwFlags & MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED) {
std::cerr << "SSMF: Type change";
break;
} else if (pSample == NULL) {
// generally this will happen when dwFlags contains ENDOFSTREAM,
// so it'll be caught before now -bkgood
std::cerr << "SSMF: No sample";
continue;
} // we now own a ref to the instance at pSample
IMFMediaBuffer *pMBuffer(NULL);
// I know this does at least a memcopy and maybe a malloc, if we have
// xrun issues with this we might want to look into using
// IMFSample::GetBufferByIndex (although MS doesn't recommend this)
if (FAILED(hr = pSample->ConvertToContiguousBuffer(&pMBuffer))) {
error = true;
goto releaseSample;
}
short *buffer(NULL);
size_t bufferLength(0);
hr = pMBuffer->Lock(reinterpret_cast<unsigned __int8**>(&buffer), NULL,
reinterpret_cast<DWORD*>(&bufferLength));
if (FAILED(hr)) {
error = true;
goto releaseMBuffer;
}
bufferLength /= (m_iBitsPerSample / 8 * m_iChannels); // now in frames
if (m_seeking) {
__int64 bufferPosition(frameFromMF(timestamp));
if (sDebug) {
std::cout << "While seeking to "
<< m_nextFrame << "WMF put us at " << bufferPosition
<< std::endl;
}
if (m_nextFrame < bufferPosition) {
// Uh oh. We are farther forward than our seek target. Emit
// silence? We can't seek backwards here.
SHORT_SAMPLE* pBufferCurpos = destBuffer +
(size - framesNeeded * m_iChannels);
__int64 offshootFrames = bufferPosition - m_nextFrame;
// If we can correct this immediately, write zeros and adjust
// m_nextFrame to pretend it never happened.
if (offshootFrames <= framesNeeded) {
std::cerr << __FILE__ << __LINE__
<< "Working around inaccurate seeking. Writing silence for"
<< offshootFrames << "frames";
// Set offshootFrames * m_iChannels samples to zero.
memset(pBufferCurpos, 0,
sizeof(*pBufferCurpos) * offshootFrames *
m_iChannels);
// Now m_nextFrame == bufferPosition
m_nextFrame += offshootFrames;
framesNeeded -= offshootFrames;
} else {
// It's more complicated. The buffer we have just decoded is
// more than framesNeeded frames away from us. It's too hard
// for us to handle this correctly currently, so let's just
// try to get on with our lives.
m_seeking = false;
m_nextFrame = bufferPosition;
std::cerr << __FILE__ << __LINE__
<< "Seek offshoot is too drastic. Cutting losses and pretending the current decoded audio buffer is the right seek point.";
}
}
if (m_nextFrame >= bufferPosition &&
m_nextFrame < bufferPosition + bufferLength) {
// m_nextFrame is in this buffer.
buffer += (m_nextFrame - bufferPosition) * m_iChannels;
bufferLength -= m_nextFrame - bufferPosition;
m_seeking = false;
} else {
// we need to keep going forward
goto releaseRawBuffer;
}
}
// If the bufferLength is larger than the leftover buffer, re-allocate
// it with 2x the space.
if (bufferLength * m_iChannels > m_leftoverBufferSize) {
int newSize = m_leftoverBufferSize;
while (newSize < bufferLength * m_iChannels) {
newSize *= 2;
}
SHORT_SAMPLE* newBuffer = new SHORT_SAMPLE[newSize];
memcpy(newBuffer, m_leftoverBuffer,
sizeof(m_leftoverBuffer[0]) * m_leftoverBufferSize);
delete [] m_leftoverBuffer;
m_leftoverBuffer = newBuffer;
m_leftoverBufferSize = newSize;
}
copyFrames(destBuffer + (size - framesNeeded * m_iChannels),
&framesNeeded, buffer, bufferLength);
releaseRawBuffer:
hr = pMBuffer->Unlock();
// I'm ignoring this, MSDN for IMFMediaBuffer::Unlock stipulates
// nothing about the state of the instance if this fails so might as
// well just let it be released.
//if (FAILED(hr)) break;
releaseMBuffer:
safeRelease(&pMBuffer);
releaseSample:
safeRelease(&pSample);
if (error) break;
}
m_nextFrame += framesRequested - framesNeeded;
if (m_leftoverBufferLength > 0) {
if (framesNeeded != 0) {
std::cerr << __FILE__ << __LINE__
<< "WARNING: Expected frames needed to be 0. Abandoning this file." << std::endl;
m_dead = true;
}
m_leftoverBufferPosition = m_nextFrame;
}
long samples_read = size - framesNeeded * m_iChannels;
m_iCurrentPosition += samples_read;
if (sDebug) { std::cout << "read() " << size << " returning " << samples_read << std::endl; }
const int sampleMax = 1 << (m_iBitsPerSample-1);
//Convert to float samples
if (m_iChannels == 2)
{
SAMPLE *destBufferFloat(const_cast<SAMPLE*>(destination));
for (unsigned long i = 0; i < samples_read; i++)
{
destBufferFloat[i] = destBuffer[i] / (float)sampleMax;
}
}
else //Assuming mono, duplicate into stereo frames...
{
SAMPLE *destBufferFloat(const_cast<SAMPLE*>(destination));
for (unsigned long i = 0; i < samples_read; i++)
{
destBufferFloat[i] = destBuffer[i] / (float)sampleMax;
}
}
return samples_read;
}
inline int AudioDecoderMediaFoundation::numSamples()
{
int len(secondsFromMF(m_mfDuration) * m_iSampleRate * m_iChannels);
return len % m_iChannels == 0 ? len : len + 1;
}
std::vector<std::string> AudioDecoderMediaFoundation::supportedFileExtensions()
{
std::vector<std::string> list;
list.push_back("m4a");
list.push_back("mp4");
list.push_back("wma");
list.push_back("mp3");
list.push_back("wav");
list.push_back("aif");
list.push_back("aiff");
return list;
}
//-------------------------------------------------------------------
// configureAudioStream
//
// Selects an audio stream from the source file, and configures the
// stream to deliver decoded PCM audio.
//-------------------------------------------------------------------
/** Cobbled together from:
http://msdn.microsoft.com/en-us/library/dd757929(v=vs.85).aspx
and http://msdn.microsoft.com/en-us/library/dd317928(VS.85).aspx
-- Albert
If anything in here fails, just bail. I'm not going to decode HRESULTS.
-- Bill
*/
bool AudioDecoderMediaFoundation::configureAudioStream()
{
HRESULT hr(S_OK);
// deselect all streams, we only want the first
hr = m_pReader->SetStreamSelection(MF_SOURCE_READER_ALL_STREAMS, false);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to deselect all streams";
return false;
}
hr = m_pReader->SetStreamSelection(MF_SOURCE_READER_FIRST_AUDIO_STREAM, true);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to select first audio stream";
return false;
}
//Debugging:
//Let's get some info
// Get the complete uncompressed format.
//hr = m_pReader->GetCurrentMediaType(
// MF_SOURCE_READER_FIRST_AUDIO_STREAM,
// &m_pAudioType);
hr = m_pReader->GetNativeMediaType(MF_SOURCE_READER_FIRST_AUDIO_STREAM,
0, //Index of the media type to retreive... (what does that even mean?)
&m_pAudioType);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to retrieve completed media type";
return false;
}
UINT32 allSamplesIndependent = 0;
UINT32 fixedSizeSamples = 0;
UINT32 sampleSize = 0;
UINT32 bitsPerSample = 0;
UINT32 blockAlignment = 0;
UINT32 numChannels = 0;
UINT32 samplesPerSecond = 0;
hr = m_pAudioType->GetUINT32(MF_MT_ALL_SAMPLES_INDEPENDENT, &allSamplesIndependent);
hr = m_pAudioType->GetUINT32(MF_MT_FIXED_SIZE_SAMPLES, &fixedSizeSamples);
hr = m_pAudioType->GetUINT32(MF_MT_SAMPLE_SIZE, &sampleSize);
hr = m_pAudioType->GetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, &bitsPerSample);
hr = m_pAudioType->GetUINT32(MF_MT_AUDIO_BLOCK_ALIGNMENT, &blockAlignment);
hr = m_pAudioType->GetUINT32(MF_MT_AUDIO_NUM_CHANNELS, &numChannels);
hr = m_pAudioType->GetUINT32(MF_MT_AUDIO_SAMPLES_PER_SECOND, &samplesPerSecond);
std::cout << "bitsPerSample: " << bitsPerSample << std::endl;
std::cout << "allSamplesIndependent: " << allSamplesIndependent << std::endl;
std::cout << "fixedSizeSamples: " << fixedSizeSamples << std::endl;
std::cout << "sampleSize: " << sampleSize << std::endl;
std::cout << "bitsPerSample: " << bitsPerSample << std::endl;
std::cout << "blockAlignment: " << blockAlignment << std::endl;
std::cout << "numChannels: " << numChannels << std::endl;
std::cout << "samplesPerSecond: " << samplesPerSecond << std::endl;
m_iChannels = numChannels;
m_iSampleRate = samplesPerSecond;
m_iBitsPerSample = bitsPerSample;
//For compressed files, the bits per sample is undefined, so by convention we're
//going to get 16-bit integers out.
if (m_iBitsPerSample == 0)
{
m_iBitsPerSample = kBitsPerSample;
}
hr = MFCreateMediaType(&m_pAudioType);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to create media type";
return false;
}
hr = m_pAudioType->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Audio);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set major type";
return false;
}
hr = m_pAudioType->SetGUID(MF_MT_SUBTYPE, MFAudioFormat_PCM);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set subtype";
return false;
}
/*
hr = m_pAudioType->SetUINT32(MF_MT_ALL_SAMPLES_INDEPENDENT, true);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set samples independent";
return false;
}
hr = m_pAudioType->SetUINT32(MF_MT_FIXED_SIZE_SAMPLES, true);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set fixed size samples";
return false;
}
hr = m_pAudioType->SetUINT32(MF_MT_SAMPLE_SIZE, kLeftoverSize);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set sample size";
return false;
}
// MSDN for this attribute says that if bps is 8, samples are unsigned.
// Otherwise, they're signed (so they're signed for us as 16 bps). Why
// chose to hide this rather useful tidbit here is beyond me -bkgood
hr = m_pAudioType->SetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, kBitsPerSample);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set bits per sample";
return false;
}
hr = m_pAudioType->SetUINT32(MF_MT_AUDIO_BLOCK_ALIGNMENT,
numChannels * (kBitsPerSample / 8));
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set block alignment";
return false;
}
*/
/*
//MediaFoundation will not convert between mono and stereo without a transform!
hr = m_pAudioType->SetUINT32(MF_MT_AUDIO_NUM_CHANNELS, kNumChannels);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set number of channels";
return false;
}
//MediaFoundation will not do samplerate conversion without a transform in the pipeline.
hr = m_pAudioType->SetUINT32(MF_MT_AUDIO_SAMPLES_PER_SECOND, kSampleRate);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set sample rate";
return false;
}
*/
// Set this type on the source reader. The source reader will
// load the necessary decoder.
hr = m_pReader->SetCurrentMediaType(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
NULL, m_pAudioType);
// the reader has the media type now, free our reference so we can use our
// pointer for other purposes. Do this before checking for failure so we
// don't dangle.
safeRelease(&m_pAudioType);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to set media type";
return false;
}
// Get the complete uncompressed format.
hr = m_pReader->GetCurrentMediaType(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
&m_pAudioType);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to retrieve completed media type";
return false;
}
// Ensure the stream is selected.
hr = m_pReader->SetStreamSelection(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
true);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to select first audio stream (again)";
return false;
}
// this may not be safe on all platforms as m_leftoverBufferSize is a
// size_t and this function is writing a uint32. However, on 32-bit
// Windows 7, size_t is defined as uint which is 32-bits, so we're safe
// for all supported platforms -bkgood
UINT32 leftoverBufferSize = 0;
hr = m_pAudioType->GetUINT32(MF_MT_SAMPLE_SIZE, &leftoverBufferSize);
if (FAILED(hr)) {
std::cerr << "SSMF: failed to get buffer size";
leftoverBufferSize = 32;
// return false;
}
m_leftoverBufferSize = static_cast<size_t>(leftoverBufferSize);
m_leftoverBufferSize /= 2; // convert size in bytes to size in int16s
m_leftoverBuffer = new short[m_leftoverBufferSize];
return true;
}
bool AudioDecoderMediaFoundation::readProperties()
{
PROPVARIANT prop;
HRESULT hr = S_OK;
//Get the duration, provided as a 64-bit integer of 100-nanosecond units
hr = m_pReader->GetPresentationAttribute(MF_SOURCE_READER_MEDIASOURCE,
MF_PD_DURATION, &prop);
if (FAILED(hr)) {
std::cerr << "SSMF: error getting duration";
return false;
}
// QuadPart isn't available on compilers that don't support _int64. Visual
// Studio 6.0 introduced the type in 1998, so I think we're safe here
// -bkgood
m_fDuration = secondsFromMF(prop.hVal.QuadPart);
m_mfDuration = prop.hVal.QuadPart;
std::cout << "SSMF: Duration: " << m_fDuration << std::endl;
PropVariantClear(&prop);
// presentation attribute MF_PD_AUDIO_ENCODING_BITRATE only exists for
// presentation descriptors, one of which MFSourceReader is not.
// Therefore, we calculate it ourselves.
//m_iBitrate = m_iBitsPerSample * m_iSampleRate * m_iChannels;
//XXX: Should we implement bitrate in libaudiodecoder? Just enable that line...
return true;
}
/**
* Copies min(destFrames, srcFrames) frames to dest from src. Anything leftover
* is moved to the beginning of m_leftoverBuffer, so empty it first (possibly
* with this method). If src and dest overlap, I'll hurt you.
*/
void AudioDecoderMediaFoundation::copyFrames(
short *dest, size_t *destFrames, const short *src, size_t srcFrames)
{
if (srcFrames > *destFrames) {
int samplesToCopy(*destFrames * m_iChannels);
memcpy(dest, src, samplesToCopy * sizeof(*src));
srcFrames -= *destFrames;
memmove(m_leftoverBuffer,
src + samplesToCopy,
srcFrames * m_iChannels * sizeof(*src));
*destFrames = 0;
m_leftoverBufferLength = srcFrames;
} else {
int samplesToCopy(srcFrames * m_iChannels);
memcpy(dest, src, samplesToCopy * sizeof(*src));
*destFrames -= srcFrames;
if (src == m_leftoverBuffer) {
m_leftoverBufferLength = 0;
}
}
}
/**
* Convert a 100ns Media Foundation value to a number of seconds.
*/
inline double AudioDecoderMediaFoundation::secondsFromMF(__int64 mf)
{
return static_cast<double>(mf) / 1e7;
}
/**
* Convert a number of seconds to a 100ns Media Foundation value.
*/
inline __int64 AudioDecoderMediaFoundation::mfFromSeconds(double sec)
{
return sec * 1e7;
}
/**
* Convert a 100ns Media Foundation value to a frame offset.
*/
inline __int64 AudioDecoderMediaFoundation::frameFromMF(__int64 mf)
{
return static_cast<double>(mf) * m_iSampleRate / 1e7;
}
/**
* Convert a frame offset to a 100ns Media Foundation value.
*/
inline __int64 AudioDecoderMediaFoundation::mfFromFrame(__int64 frame)
{
return static_cast<double>(frame) / m_iSampleRate * 1e7;
}