void AudioEngine::mixAudio(Uint8 *pDestBuffer, int destBufferLen)
{
int numFrames = destBufferLen/(2*getChannels()); // 16 bit samples.
if (m_AudioSources.size() == 0) {
return;
}
if (!m_pTempBuffer || m_pTempBuffer->getNumFrames() < numFrames) {
if (m_pTempBuffer) {
delete[] m_pMixBuffer;
}
m_pTempBuffer = AudioBufferPtr(new AudioBuffer(numFrames, m_AP));
m_pMixBuffer = new float[getChannels()*numFrames];
}
for (int i = 0; i < getChannels()*numFrames; ++i) {
m_pMixBuffer[i]=0;
}
{
lock_guard lock(m_Mutex);
AudioSourceMap::iterator it;
for (it = m_AudioSources.begin(); it != m_AudioSources.end(); it++) {
m_pTempBuffer->clear();
it->second->fillAudioBuffer(m_pTempBuffer);
addBuffers(m_pMixBuffer, m_pTempBuffer);
}
}
calcVolume(m_pMixBuffer, numFrames*getChannels(), getVolume());
for (int i = 0; i < numFrames; ++i) {
m_pLimiter->process(m_pMixBuffer+i*getChannels());
for (int j = 0; j < getChannels(); ++j) {
((short*)pDestBuffer)[i*2+j]=short(m_pMixBuffer[i*2+j]*32768);
}
}
}
// 获取声音缓冲区
/////////////////////////////////////////////////////////////////////////////////
AudioBufferPtr AudioEngine::Buffer(size_t bufID) const
{
AudioBufsConstIter iter(audioBufs_.find(bufID));
if (iter != audioBufs_.end())
{
return iter->second;
}
BOOST_ASSERT(false);
return AudioBufferPtr();
}
void AudioDecoderThread::decodePacket(AVPacket* pPacket)
{
char* pDecodedData = 0;
AVPacket* pTempPacket = new AVPacket;
av_init_packet(pTempPacket);
pTempPacket->data = pPacket->data;
pTempPacket->size = pPacket->size;
AVFrame* pDecodedFrame;
pDecodedFrame = avcodec_alloc_frame();
while (pTempPacket->size > 0) {
int gotFrame = 0;
int bytesDecoded;
int bytesConsumed = avcodec_decode_audio4(m_pStream->codec, pDecodedFrame,
&gotFrame, pTempPacket);
if (gotFrame) {
bytesDecoded = av_samples_get_buffer_size(0, m_pStream->codec->channels,
pDecodedFrame->nb_samples, m_pStream->codec->sample_fmt, 1);
pDecodedData = (char*)(pDecodedFrame->data[0]);
} else {
bytesDecoded = 0;
}
// This is triggered for some strange/broken videos.
// AVG_ASSERT(bytesConsumed != 0);
if (bytesConsumed < 0) {
// Error decoding -> throw away current packet.
bytesDecoded = 0;
pTempPacket->size = 0;
} else {
pTempPacket->data += bytesConsumed;
pTempPacket->size -= bytesConsumed;
}
if (bytesDecoded > 0) {
int framesDecoded = bytesDecoded/(m_pStream->codec->channels*
getBytesPerSample(m_InputSampleFormat));
AudioBufferPtr pBuffer;
bool bNeedsResample = (m_InputSampleRate != m_AP.m_SampleRate ||
m_InputSampleFormat != AV_SAMPLE_FMT_S16 ||
m_pStream->codec->channels != m_AP.m_Channels);
bool bIsPlanar = false;
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51, 27, 0)
bIsPlanar = av_sample_fmt_is_planar((AVSampleFormat)m_InputSampleFormat);
if (bIsPlanar) {
char* pPackedData = (char*)av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE +
FF_INPUT_BUFFER_PADDING_SIZE);
planarToInterleaved(pPackedData, pDecodedFrame, m_pStream->codec->channels,
framesDecoded);
pBuffer = resampleAudio(pPackedData, framesDecoded,
av_get_packed_sample_fmt((AVSampleFormat)m_InputSampleFormat));
av_free(pPackedData);
bNeedsResample = false;
}
#endif
if (bNeedsResample) {
pBuffer = resampleAudio(pDecodedData, framesDecoded,
m_InputSampleFormat);
} else if (!bIsPlanar) {
pBuffer = AudioBufferPtr(new AudioBuffer(framesDecoded, m_AP));
memcpy(pBuffer->getData(), pDecodedData, bytesDecoded);
}
m_LastFrameTime += float(pBuffer->getNumFrames())/m_AP.m_SampleRate;
pushAudioMsg(pBuffer, m_LastFrameTime);
}
}
#if LIBAVCODEC_VERSION_MAJOR > 53
avcodec_free_frame(&pDecodedFrame);
delete pDecodedFrame;
#endif
delete pTempPacket;
}
