Hey there! avcodec_open() has been deprecated for some time, and indeed has been removed in the most recent versions of libavcodec/ffmpeg. You're already using avcodec_open2() elsewhere, but one appears to have been missed. The change is trivial:
[skynet](0) $ svn diff
Index: src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp
===================================================================
--- src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp (revision 13355)
+++ src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp (working copy)
@@ -109,7 +109,7 @@
// m_context->flags |= CODEC_FLAG_TRUNCATED;
// Open codec
- if (avcodec_open(m_context, p_codec) < 0)
+ if (avcodec_open2(m_context, p_codec, NULL) < 0)
throw std::runtime_error("avcodec_open() failed");
}
[skynet](0) $
I've applied similar changes this year to many other packages, including
cheese, blender, linphone, ad nauseam. It's been tested by verifying that
with my patch, OSG builds against the newest libavcodec and ffmpeg, whereas
otherwise it does not.
The modified src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp is attached. It
originates in svn trunk revision 13355. Thanks!
--
nick black http://www.sprezzatech.com -- unix and hpc consulting
to make an apple pie from scratch, you need first invent a universe.
#include "FFmpegDecoderAudio.hpp"
#include <osg/Notify>
#include <stdexcept>
#include <string.h>
//DEBUG
//#include <iostream>
namespace osgFFmpeg {
static int decode_audio(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
const uint8_t *buf, int buf_size)
{
#if LIBAVCODEC_VERSION_MAJOR >= 53 || (LIBAVCODEC_VERSION_MAJOR==52 && LIBAVCODEC_VERSION_MINOR>=32)
// following code segment copied from ffmpeg's avcodec_decode_audio2()
// implementation to avoid warnings about deprecated function usage.
AVPacket avpkt;
av_init_packet(&avpkt);
avpkt.data = const_cast<uint8_t *>(buf);
avpkt.size = buf_size;
return avcodec_decode_audio3(avctx, samples, frame_size_ptr, &avpkt);
#else
// fallback for older versions of ffmpeg that don't have avcodec_decode_audio3.
return avcodec_decode_audio2(avctx, samples, frame_size_ptr, buf, buf_size);
#endif
}
FFmpegDecoderAudio::FFmpegDecoderAudio(PacketQueue & packets, FFmpegClocks & clocks) :
m_packets(packets),
m_clocks(clocks),
m_stream(0),
m_context(0),
m_packet_data(0),
m_bytes_remaining(0),
m_audio_buffer((AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2),
m_audio_buf_size(0),
m_audio_buf_index(0),
m_end_of_stream(false),
m_paused(true),
m_exit(false)
{
}
FFmpegDecoderAudio::~FFmpegDecoderAudio()
{
this->close(true);
}
void FFmpegDecoderAudio::open(AVStream * const stream)
{
try
{
// Sound can be optional (i.e. no audio stream is present)
if (stream == 0)
return;
m_stream = stream;
m_context = stream->codec;
m_frequency = m_context->sample_rate;
m_nb_channels = m_context->channels;
switch (m_context->sample_fmt)
{
case AV_SAMPLE_FMT_NONE:
throw std::runtime_error("invalid audio format AV_SAMPLE_FMT_NONE");
case AV_SAMPLE_FMT_U8:
m_sample_format = osg::AudioStream::SAMPLE_FORMAT_U8;
break;
case AV_SAMPLE_FMT_S16:
m_sample_format = osg::AudioStream::SAMPLE_FORMAT_S16;
break;
case AV_SAMPLE_FMT_S32:
m_sample_format = osg::AudioStream::SAMPLE_FORMAT_S32;
break;
case AV_SAMPLE_FMT_FLT:
m_sample_format = osg::AudioStream::SAMPLE_FORMAT_F32;
break;
case AV_SAMPLE_FMT_DBL:
throw std::runtime_error("unhandled audio format AV_SAMPLE_FMT_DBL");
default:
throw std::runtime_error("unknown audio format");
}
// Check stream sanity
if (m_context->codec_id == CODEC_ID_NONE)
throw std::runtime_error("invalid audio codec");;
// Find the decoder for the audio stream
AVCodec * const p_codec = avcodec_find_decoder(m_context->codec_id);
if (p_codec == 0)
throw std::runtime_error("avcodec_find_decoder() failed");
// Inform the codec that we can handle truncated bitstreams
//if (p_codec->capabilities & CODEC_CAP_TRUNCATED)
// m_context->flags |= CODEC_FLAG_TRUNCATED;
// Open codec
if (avcodec_open2(m_context, p_codec, NULL) < 0)
throw std::runtime_error("avcodec_open() failed");
}
catch (...)
{
m_context = 0;
throw;
}
}
void FFmpegDecoderAudio::pause(bool pause)
{
if (pause != m_paused)
{
m_paused = pause;
if (m_audio_sink.valid())
{
if (m_paused) m_audio_sink->pause();
else m_audio_sink->play();
}
}
}
void FFmpegDecoderAudio::close(bool waitForThreadToExit)
{
if (isRunning())
{
m_exit = true;
if (waitForThreadToExit)
join();
}
}
void FFmpegDecoderAudio::setVolume(float volume)
{
if (m_audio_sink.valid())
{
m_audio_sink->setVolume(volume);
}
}
float FFmpegDecoderAudio::getVolume() const
{
if (m_audio_sink.valid())
{
return m_audio_sink->getVolume();
}
return 0.0f;
}
void FFmpegDecoderAudio::run()
{
try
{
decodeLoop();
}
catch (const std::exception & error)
{
OSG_WARN << "FFmpegDecoderAudio::run : " << error.what() << std::endl;
}
catch (...)
{
OSG_WARN << "FFmpegDecoderAudio::run : unhandled exception" << std::endl;
}
}
void FFmpegDecoderAudio::setAudioSink(osg::ref_ptr<osg::AudioSink> audio_sink)
{
// The FFmpegDecoderAudio object takes the responsability of destroying the audio_sink.
OSG_NOTICE<<"Assigning "<<audio_sink<<std::endl;
m_audio_sink = audio_sink;
}
void FFmpegDecoderAudio::fillBuffer(void * const buffer, size_t size)
{
uint8_t * dst_buffer = reinterpret_cast<uint8_t*>(buffer);
while (size != 0)
{
if (m_audio_buf_index == m_audio_buf_size)
{
m_audio_buf_index = 0;
// Pre-fetch audio buffer is empty, refill it.
const size_t bytes_decoded = decodeFrame(&m_audio_buffer[0], m_audio_buffer.size());
// If nothing could be decoded (e.g. error or no packet available), output a bit of silence
if (bytes_decoded == 0)
{
m_audio_buf_size = std::min(Buffer::size_type(1024), m_audio_buffer.size());
memset(&m_audio_buffer[0], 0, m_audio_buf_size);
}
else
{
m_audio_buf_size = bytes_decoded;
}
}
const size_t fill_size = std::min(m_audio_buf_size - m_audio_buf_index, size);
memcpy(dst_buffer, &m_audio_buffer[m_audio_buf_index], fill_size);
size -= fill_size;
dst_buffer += fill_size;
m_audio_buf_index += fill_size;
adjustBufferEndTps(fill_size);
}
}
void FFmpegDecoderAudio::decodeLoop()
{
const bool skip_audio = ! validContext() || ! m_audio_sink.valid();
if (! skip_audio && ! m_audio_sink->playing())
{
m_clocks.audioSetDelay(m_audio_sink->getDelay());
m_audio_sink->play();
}
else
{
m_clocks.audioDisable();
}
while (! m_exit)
{
if(m_paused)
{
m_clocks.pause(true);
m_pause_timer.setStartTick();
while(m_paused && !m_exit)
{
microSleep(10000);
}
m_clocks.setPauseTime(m_pause_timer.time_s());
m_clocks.pause(false);
}
// If skipping audio, make sure the audio stream is still consumed.
if (skip_audio)
{
bool is_empty;
FFmpegPacket packet = m_packets.timedPop(is_empty, 10);
if (packet.valid())
packet.clear();
}
// Else, just idle in this thread.
// Note: If m_audio_sink has an audio callback, this thread will still be awaken
// from time to time to refill the audio buffer.
else
{
OpenThreads::Thread::microSleep(10000);
}
}
}
void FFmpegDecoderAudio::adjustBufferEndTps(const size_t buffer_size)
{
int sample_size = nbChannels() * frequency();
switch (sampleFormat())
{
case osg::AudioStream::SAMPLE_FORMAT_U8:
sample_size *= 1;
break;
case osg::AudioStream::SAMPLE_FORMAT_S16:
sample_size *= 2;
break;
case osg::AudioStream::SAMPLE_FORMAT_S24:
sample_size *= 3;
break;
case osg::AudioStream::SAMPLE_FORMAT_S32:
sample_size *= 4;
break;
case osg::AudioStream::SAMPLE_FORMAT_F32:
sample_size *= 4;
break;
default:
throw std::runtime_error("unsupported audio sample format");
}
m_clocks.audioAdjustBufferEndPts(double(buffer_size) / double(sample_size));
}
size_t FFmpegDecoderAudio::decodeFrame(void * const buffer, const size_t size)
{
for (;;)
{
// Decode current packet
while (m_bytes_remaining > 0)
{
int data_size = size;
const int bytes_decoded = decode_audio(m_context, reinterpret_cast<int16_t*>(buffer), &data_size, m_packet_data, m_bytes_remaining);
if (bytes_decoded < 0)
{
// if error, skip frame
m_bytes_remaining = 0;
break;
}
m_bytes_remaining -= bytes_decoded;
m_packet_data += bytes_decoded;
// If we have some data, return it and come back for more later.
if (data_size > 0)
return data_size;
}
// Get next packet
if (m_packet.valid())
m_packet.clear();
if (m_exit)
return 0;
bool is_empty = true;
m_packet = m_packets.tryPop(is_empty);
if (is_empty)
return 0;
if (m_packet.type == FFmpegPacket::PACKET_DATA)
{
if (m_packet.packet.pts != int64_t(AV_NOPTS_VALUE))
{
const double pts = av_q2d(m_stream->time_base) * m_packet.packet.pts;
m_clocks.audioSetBufferEndPts(pts);
}
m_bytes_remaining = m_packet.packet.size;
m_packet_data = m_packet.packet.data;
}
else if (m_packet.type == FFmpegPacket::PACKET_END_OF_STREAM)
{
m_end_of_stream = true;
}
else if (m_packet.type == FFmpegPacket::PACKET_FLUSH)
{
avcodec_flush_buffers(m_context);
}
// just output silence when we reached the end of stream
if (m_end_of_stream)
{
memset(buffer, 0, size);
return size;
}
}
}
} // namespace osgFFmpeg
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