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Merge pull request mixxxdj#1024 from illuusio/ffmpeg-3.1-update
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FFMpeg 3.1 depricates AVStream->codec to favor AVStream->codecpar
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@daschuer
daschuer authored Apr 10, 2017
2 parents c43a215 + 78bd025 commit 318fead
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Showing 2 changed files with 253 additions and 29 deletions.
166 changes: 138 additions & 28 deletions src/sources/soundsourceffmpeg.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -30,7 +30,7 @@ const SINT kMaxChannelCount = 2;

inline
AVMediaType getMediaTypeOfStream(AVStream* pStream) {
return pStream->codec->codec_type;
return m_pAVStreamWrapper.getMediaTypeOfStream(pStream);
}

AVStream* findFirstAudioStream(AVFormatContext* pFormatCtx) {
Expand All @@ -51,17 +51,17 @@ AVStream* findFirstAudioStream(AVFormatContext* pFormatCtx) {

inline
AVCodec* findDecoderForStream(AVStream* pStream) {
return avcodec_find_decoder(pStream->codec->codec_id);
return m_pAVStreamWrapper.findDecoderForStream(pStream);
}

inline
SINT getChannelCountOfStream(AVStream* pStream) {
return pStream->codec->channels;
return m_pAVStreamWrapper.getChannelCountOfStream(pStream);
}

inline
SINT getSamplingRateOfStream(AVStream* pStream) {
return pStream->codec->sample_rate;
return m_pAVStreamWrapper.getSamplingRateOfStream(pStream);
}

inline
Expand Down Expand Up @@ -107,7 +107,7 @@ bool getFrameCountOfStream(AVStream* pStream, SINT* pFrameCount) {

inline
AVSampleFormat getSampleFormatOfStream(AVStream* pStream) {
return pStream->codec->sample_fmt;
return m_pAVStreamWrapper.getSampleFormatOfStream(pStream);
}

} // anonymous namespace
Expand Down Expand Up @@ -230,16 +230,10 @@ void SoundSourceFFmpeg::ClosableInputAVFormatContextPtr::close() {

//static
SoundSource::OpenResult SoundSourceFFmpeg::openAudioStream(
AVStream* pAudioStream) {
DEBUG_ASSERT(pAudioStream != nullptr);
AVCodec* pDecoder = findDecoderForStream(pAudioStream);
if (pDecoder == nullptr) {
qWarning() << "[SoundSourceFFmpeg]"
<< "Failed to find a decoder for stream"
<< pAudioStream->index;
return SoundSource::OpenResult::ABORTED;
}
const int avcodec_open2_result = avcodec_open2(pAudioStream->codec, pDecoder, nullptr);
AVCodecContext* pCodecContext, AVCodec *pDecoder) {
DEBUG_ASSERT(pCodecContext != nullptr);

const int avcodec_open2_result = avcodec_open2(pCodecContext, pDecoder, nullptr);
if (avcodec_open2_result < 0) {
qWarning() << "[SoundSourceFFmpeg]"
<< "avcodec_open2() failed and returned"
Expand All @@ -260,17 +254,37 @@ void SoundSourceFFmpeg::ClosableAVStreamPtr::take(AVStream** ppClosableStream) {

void SoundSourceFFmpeg::ClosableAVStreamPtr::close() {
if (m_pClosableStream != nullptr) {
#if ! AVSTREAM_FROM_API_VERSION_3_1
const int avcodec_close_result = avcodec_close(m_pClosableStream->codec);
if (avcodec_close_result != 0) {
qWarning() << "[SoundSourceFFmpeg]"
<< "avcodec_close() failed and returned"
<< avcodec_close_result;
// ignore error and continue
}
#endif
m_pClosableStream = nullptr;
}
}

#if AVSTREAM_FROM_API_VERSION_3_1
void SoundSourceFFmpeg::ClosableAVCodecContextPtr::take(AVCodecContext** ppClosableContext) {
DEBUG_ASSERT(ppClosableContext != nullptr);
if (m_pClosableContext != *ppClosableContext) {
close();
m_pClosableContext = *ppClosableContext;
*ppClosableContext = nullptr;
}
}

void SoundSourceFFmpeg::ClosableAVCodecContextPtr::close() {
if (m_pClosableContext != nullptr) {
avcodec_free_context(&m_pClosableContext);
m_pClosableContext = nullptr;
}
}
#endif

SoundSourceFFmpeg::SoundSourceFFmpeg(const QUrl& url)
: SoundSource(url),
m_pResample(nullptr),
Expand Down Expand Up @@ -321,10 +335,52 @@ SoundSource::OpenResult SoundSourceFFmpeg::tryOpen(const AudioSourceConfig& /*au
<< "No audio stream found";
return OpenResult::ABORTED;
}
const OpenResult openAudioStreamResult = openAudioStream(pAudioStream);

// Find codec to decode stream or pass out
AVCodec* pDecoder = findDecoderForStream(pAudioStream);
if (pDecoder == nullptr) {
qWarning() << "[SoundSourceFFmpeg]"
<< "Failed to find a decoder for stream"
<< pAudioStream->index;
return SoundSource::OpenResult::ABORTED;
}

#if AVSTREAM_FROM_API_VERSION_3_1
AVCodecContext *pCodecContext = avcodec_alloc_context3(pDecoder);

if (pCodecContext == nullptr) {
qWarning() << "[SoundSourceFFmpeg]"
<< "Failed to allocate codec context"
<< pAudioStream->index;
return SoundSource::OpenResult::ABORTED;
}

// Add stream parameters to context
if(avcodec_parameters_to_context(pCodecContext,pAudioStream->codecpar)) {
qWarning() << "[SoundSourceFFmpeg]"
<< "Failed to find to set Code parameter for AVCodecContext"
<< pAudioStream->index;
return SoundSource::OpenResult::ABORTED;
}

// Se timebase correct
av_codec_set_pkt_timebase(pCodecContext, pAudioStream->time_base);

// Make sure that Codecs are identical or avcodec_open2 fails.
pCodecContext->codec_id = pDecoder->id;

const OpenResult openAudioStreamResult = openAudioStream(pCodecContext, pDecoder);

m_pAudioContext.take(&pCodecContext);
#else
const OpenResult openAudioStreamResult = openAudioStream(pAudioStream->codec, pDecoder);
#endif


if (openAudioStreamResult != OpenResult::SUCCEEDED) {
return openAudioStreamResult; // early exit on any error
}

// Now set the member, because the audio stream has been opened
// successfully and needs to be closed eventually.
m_pAudioStream.take(&pAudioStream);
Expand Down Expand Up @@ -352,7 +408,7 @@ SoundSource::OpenResult SoundSourceFFmpeg::tryOpen(const AudioSourceConfig& /*au
}

SINT frameCount = getFrameCount();
if (getFrameCountOfStream(m_pAudioStream, &frameCount) && isValidFrameCount(frameCount)) {
if (getFrameCountOfStream(m_pAudioStream, &frameCount) && isValidFrameCount(frameCount) == false) {
qWarning() << "[SoundSourceFFmpeg]"
<< "Stream has invalid number of frames:"
<< frameCount;
Expand All @@ -363,7 +419,11 @@ SoundSource::OpenResult SoundSourceFFmpeg::tryOpen(const AudioSourceConfig& /*au
setSamplingRate(samplingRate);
setFrameCount(frameCount);

#if AVSTREAM_FROM_API_VERSION_3_1
m_pResample = std::make_unique<EncoderFfmpegResample>(m_pAudioContext);
#else
m_pResample = std::make_unique<EncoderFfmpegResample>(m_pAudioStream->codec);
#endif
m_pResample->openMixxx(getSampleFormatOfStream(m_pAudioStream), AV_SAMPLE_FMT_FLT);

return OpenResult::SUCCEEDED;
Expand All @@ -380,6 +440,9 @@ void SoundSourceFFmpeg::close() {
free(l_SRmJmp);
}

#if AVSTREAM_FROM_API_VERSION_3_1
m_pAudioContext.close();
#endif
m_pAudioStream.close();
m_pInputFormatContext.close();
}
Expand All @@ -401,7 +464,9 @@ bool SoundSourceFFmpeg::readFramesToCache(unsigned int count, SINT offset) {
l_SPacket.size = 0;
AVFrame *l_pFrame = nullptr;
bool l_bStop = false;
#if ! AVSTREAM_FROM_API_VERSION_3_1
int l_iFrameFinished = 0;
#endif
struct ffmpegCacheObject *l_SObj = nullptr;
struct ffmpegCacheObject *l_SRmObj = nullptr;
qint64 l_lLastPacketPos = -1;
Expand Down Expand Up @@ -447,8 +512,17 @@ bool SoundSourceFFmpeg::readFramesToCache(unsigned int count, SINT offset) {
// Are we on correct audio stream. Currently we are always
// Using first audio stream but in future there should be
// possibility to choose which to use
// If Pos is -1 it meand FFmpeg/AVConv doesn't know it
// So then we use pts instead
if (l_SPacket.stream_index == m_pAudioStream->index &&
l_SPacket.pos >= 0) {
(l_SPacket.pos >= 0 || l_SPacket.pos == -1)) {

// Codecs like Wavpack does it like this
// They work but you can say about position nothing
if (l_SPacket.pos == -1)
{
l_SPacket.pos = l_SPacket.pts;
}
if (m_lStoredSeekPoint > 0) {
struct ffmpegLocationObject *l_STestObj = nullptr;
if (m_SJumpPoints.size() > 0) {
Expand All @@ -474,14 +548,43 @@ bool SoundSourceFFmpeg::readFramesToCache(unsigned int count, SINT offset) {
m_SStoredJumpPoint = nullptr;
}

#if AVSTREAM_FROM_API_VERSION_3_1
l_iRet = avcodec_send_packet(m_pAudioContext, &l_SPacket);

// AVERROR(EAGAIN) means that we need to feed more
// That we can decode Frame or Packet
if (l_iRet == AVERROR(EAGAIN)) {
qDebug() << "SoundSourceFFmpeg::readFramesToCache: Need more packets to decode!";
continue;
}

if(l_iRet == AVERROR_EOF || l_iRet == AVERROR(EINVAL)) {
qDebug() << "SoundSourceFFmpeg::readFramesToCache: Warning can't decode frame!";
}

l_iRet = avcodec_receive_frame(m_pAudioContext, l_pFrame);

// AVERROR(EAGAIN) means that we need to feed more
// That we can decode Frame or Packet
if (l_iRet == AVERROR(EAGAIN)) {
qDebug() << "SoundSourceFFmpeg::readFramesToCache: Need more packets to decode!";
continue;
}

if(l_iRet == AVERROR_EOF || l_iRet == AVERROR(EINVAL)) {
qDebug() << "SoundSourceFFmpeg::readFramesToCache: Warning can't decode frame!";
}

if (l_iRet == AVERROR_EOF || l_iRet < 0) {
#else
// Decode audio bytes (These can be S16P or FloatP [P is Planar])
l_iRet = avcodec_decode_audio4(m_pAudioStream->codec,l_pFrame,&l_iFrameFinished,
&l_SPacket);

if (l_iRet <= 0) {
#endif
// An error or EOF occurred,index break out and return what
// we have so far.
qDebug() << "EOF!";
qDebug() << "SoundSourceFFmpeg::readFramesToCache: EOF or uncoverable error!";
l_bStop = true;
continue;
} else {
Expand Down Expand Up @@ -637,7 +740,10 @@ bool SoundSourceFFmpeg::getBytesFromCache(CSAMPLE* buffer, SINT offset,
// If cache is empty then retun without crash.
if (m_SCache.isEmpty()) {
qDebug() << "SoundSourceFFmpeg::getBytesFromCache: Cache is empty can't return bytes";
memset(l_pBuffer, 0x00, l_lLeft);
if(l_pBuffer != nullptr)
{
memset(l_pBuffer, 0x00, l_lLeft);
}
return false;
}

Expand Down Expand Up @@ -735,7 +841,10 @@ bool SoundSourceFFmpeg::getBytesFromCache(CSAMPLE* buffer, SINT offset,
} else {
qDebug() <<
"SoundSourceFFmpeg::getBytesFromCache: Buffer run out. Shouldn't happen!";
memset(l_pBuffer, 0x00, l_lLeft);
if(l_pBuffer != nullptr)
{
memset(l_pBuffer, 0x00, l_lLeft);
}
return false;
}
}
Expand Down Expand Up @@ -870,6 +979,12 @@ SINT SoundSourceFFmpeg::seekSampleFrame(SINT frameIndex) {
SINT SoundSourceFFmpeg::readSampleFrames(SINT numberOfFrames,
CSAMPLE* sampleBuffer) {

if (sampleBuffer == nullptr) {
// They are trying to make us skip
seekSampleFrame(m_currentMixxxFrameIndex + numberOfFrames);
return numberOfFrames;
}

if (m_SCache.size() == 0) {
// Make sure we always start at beginning and cache have some
// material that we can consume.
Expand All @@ -879,12 +994,7 @@ SINT SoundSourceFFmpeg::readSampleFrames(SINT numberOfFrames,

getBytesFromCache(sampleBuffer, m_currentMixxxFrameIndex, numberOfFrames);

// As this is also Hack
// If we don't seek like we don't on analyzer.. keep
// place in mind..
if (m_bIsSeeked == false) {
m_currentMixxxFrameIndex += numberOfFrames;
}
m_currentMixxxFrameIndex += numberOfFrames;

m_bIsSeeked = false;

Expand Down
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