openmw-tes3coop/apps/openmw/mwsound/ffmpeg_decoder.cpp
scrawl b39d69e98c Videoplayer fixes, play/pause & seeking
- Fix rindex overflow
 - Fix audio sample size bugs (was using sample_fmt and channel count of the decoder, instead of the resampled settings). We didn't notice this bug before, because the OpenAL MovieAudioFactory tries to resample to a format of the same byte size.
 - Add support for play/pause and seeking controls (not used by cutscenes in OpenMW)
 - Closing the video when arriving at the stream end is now handled by the user (we may also want to keep the video open and seek back)

The video player now has a standalone demo, at https://github.com/scrawl/ogre-ffmpeg-videoplayer
2014-10-24 21:31:11 +02:00

441 lines
13 KiB
C++

#include "ffmpeg_decoder.hpp"
// auto_ptr
#include <memory>
#include <stdexcept>
extern "C" {
#ifndef HAVE_LIBSWRESAMPLE
// FIXME: remove this section once libswresample is packaged for Debian
int swr_init(AVAudioResampleContext *avr);
void swr_free(AVAudioResampleContext **avr);
int swr_convert( AVAudioResampleContext *avr, uint8_t** output, int out_samples, const uint8_t** input, int in_samples);
AVAudioResampleContext * swr_alloc_set_opts( AVAudioResampleContext *avr, int64_t out_ch_layout, AVSampleFormat out_fmt, int out_rate, int64_t in_ch_layout, AVSampleFormat in_fmt, int in_rate, int o, void* l);
#endif
}
namespace MWSound
{
void FFmpeg_Decoder::fail(const std::string &msg)
{
throw std::runtime_error("FFmpeg exception: "+msg);
}
int FFmpeg_Decoder::readPacket(void *user_data, uint8_t *buf, int buf_size)
{
Ogre::DataStreamPtr stream = static_cast<FFmpeg_Decoder*>(user_data)->mDataStream;
return stream->read(buf, buf_size);
}
int FFmpeg_Decoder::writePacket(void *user_data, uint8_t *buf, int buf_size)
{
Ogre::DataStreamPtr stream = static_cast<FFmpeg_Decoder*>(user_data)->mDataStream;
return stream->write(buf, buf_size);
}
int64_t FFmpeg_Decoder::seek(void *user_data, int64_t offset, int whence)
{
Ogre::DataStreamPtr stream = static_cast<FFmpeg_Decoder*>(user_data)->mDataStream;
whence &= ~AVSEEK_FORCE;
if(whence == AVSEEK_SIZE)
return stream->size();
if(whence == SEEK_SET)
stream->seek(offset);
else if(whence == SEEK_CUR)
stream->seek(stream->tell()+offset);
else if(whence == SEEK_END)
stream->seek(stream->size()+offset);
else
return -1;
return stream->tell();
}
/* Used by getAV*Data to search for more compressed data, and buffer it in the
* correct stream. It won't buffer data for streams that the app doesn't have a
* handle for. */
bool FFmpeg_Decoder::getNextPacket()
{
if(!mStream)
return false;
int stream_idx = mStream - mFormatCtx->streams;
while(av_read_frame(mFormatCtx, &mPacket) >= 0)
{
/* Check if the packet belongs to this stream */
if(stream_idx == mPacket.stream_index)
{
if((uint64_t)mPacket.pts != AV_NOPTS_VALUE)
mNextPts = av_q2d((*mStream)->time_base)*mPacket.pts;
return true;
}
/* Free the packet and look for another */
av_free_packet(&mPacket);
}
return false;
}
bool FFmpeg_Decoder::getAVAudioData()
{
int got_frame;
if((*mStream)->codec->codec_type != AVMEDIA_TYPE_AUDIO)
return false;
do {
if(mPacket.size == 0 && !getNextPacket())
return false;
/* Decode some data, and check for errors */
int len = 0;
if((len=avcodec_decode_audio4((*mStream)->codec, mFrame, &got_frame, &mPacket)) < 0)
return false;
/* Move the unread data to the front and clear the end bits */
int remaining = mPacket.size - len;
if(remaining <= 0)
av_free_packet(&mPacket);
else
{
memmove(mPacket.data, &mPacket.data[len], remaining);
av_shrink_packet(&mPacket, remaining);
}
if(mSwr)
{
if(!mDataBuf || mDataBufLen < mFrame->nb_samples)
{
av_freep(&mDataBuf);
if(av_samples_alloc(&mDataBuf, NULL, (*mStream)->codec->channels,
mFrame->nb_samples, mOutputSampleFormat, 0) < 0)
break;
else
mDataBufLen = mFrame->nb_samples;
}
if(swr_convert(mSwr, (uint8_t**)&mDataBuf, mFrame->nb_samples,
(const uint8_t**)mFrame->extended_data, mFrame->nb_samples) < 0)
{
break;
}
mFrameData = &mDataBuf;
}
else
mFrameData = &mFrame->data[0];
} while(got_frame == 0 || mFrame->nb_samples == 0);
mNextPts += (double)mFrame->nb_samples / (double)(*mStream)->codec->sample_rate;
return true;
}
size_t FFmpeg_Decoder::readAVAudioData(void *data, size_t length)
{
size_t dec = 0;
while(dec < length)
{
/* If there's no decoded data, find some */
if(mFramePos >= mFrameSize)
{
if(!getAVAudioData())
break;
mFramePos = 0;
mFrameSize = mFrame->nb_samples * (*mStream)->codec->channels *
av_get_bytes_per_sample(mOutputSampleFormat);
}
/* Get the amount of bytes remaining to be written, and clamp to
* the amount of decoded data we have */
size_t rem = std::min<size_t>(length-dec, mFrameSize-mFramePos);
/* Copy the data to the app's buffer and increment */
memcpy(data, mFrameData[0]+mFramePos, rem);
data = (char*)data + rem;
dec += rem;
mFramePos += rem;
}
/* Return the number of bytes we were able to get */
return dec;
}
void FFmpeg_Decoder::open(const std::string &fname)
{
close();
mDataStream = mResourceMgr.openResource(fname);
if((mFormatCtx=avformat_alloc_context()) == NULL)
fail("Failed to allocate context");
mFormatCtx->pb = avio_alloc_context(NULL, 0, 0, this, readPacket, writePacket, seek);
if(!mFormatCtx->pb || avformat_open_input(&mFormatCtx, fname.c_str(), NULL, NULL) != 0)
{
// "Note that a user-supplied AVFormatContext will be freed on failure".
if (mFormatCtx)
{
if (mFormatCtx->pb != NULL)
{
if (mFormatCtx->pb->buffer != NULL)
{
av_free(mFormatCtx->pb->buffer);
mFormatCtx->pb->buffer = NULL;
}
av_free(mFormatCtx->pb);
mFormatCtx->pb = NULL;
}
avformat_free_context(mFormatCtx);
}
mFormatCtx = NULL;
fail("Failed to allocate input stream");
}
try
{
if(avformat_find_stream_info(mFormatCtx, NULL) < 0)
fail("Failed to find stream info in "+fname);
for(size_t j = 0;j < mFormatCtx->nb_streams;j++)
{
if(mFormatCtx->streams[j]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
{
mStream = &mFormatCtx->streams[j];
break;
}
}
if(!mStream)
fail("No audio streams in "+fname);
(*mStream)->codec->request_sample_fmt = (*mStream)->codec->sample_fmt;
AVCodec *codec = avcodec_find_decoder((*mStream)->codec->codec_id);
if(!codec)
{
std::stringstream ss("No codec found for id ");
ss << (*mStream)->codec->codec_id;
fail(ss.str());
}
if(avcodec_open2((*mStream)->codec, codec, NULL) < 0)
fail("Failed to open audio codec " + std::string(codec->long_name));
mFrame = av_frame_alloc();
}
catch(std::exception&)
{
if (mFormatCtx->pb->buffer != NULL)
{
av_free(mFormatCtx->pb->buffer);
mFormatCtx->pb->buffer = NULL;
}
av_free(mFormatCtx->pb);
mFormatCtx->pb = NULL;
avformat_close_input(&mFormatCtx);
throw;
}
}
void FFmpeg_Decoder::close()
{
if(mStream)
avcodec_close((*mStream)->codec);
mStream = NULL;
av_free_packet(&mPacket);
av_freep(&mFrame);
swr_free(&mSwr);
av_freep(&mDataBuf);
if(mFormatCtx)
{
if (mFormatCtx->pb != NULL)
{
// mFormatCtx->pb->buffer must be freed by hand,
// if not, valgrind will show memleak, see:
//
// https://trac.ffmpeg.org/ticket/1357
//
if (mFormatCtx->pb->buffer != NULL)
{
av_free(mFormatCtx->pb->buffer);
mFormatCtx->pb->buffer = NULL;
}
av_free(mFormatCtx->pb);
mFormatCtx->pb = NULL;
}
avformat_close_input(&mFormatCtx);
}
mDataStream.setNull();
}
std::string FFmpeg_Decoder::getName()
{
return mFormatCtx->filename;
}
void FFmpeg_Decoder::getInfo(int *samplerate, ChannelConfig *chans, SampleType *type)
{
if(!mStream)
fail("No audio stream info");
if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_U8)
*type = SampleType_UInt8;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_S16)
*type = SampleType_Int16;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_FLT)
*type = SampleType_Float32;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_U8P)
*type = SampleType_UInt8;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_S16P)
*type = SampleType_Int16;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_FLTP)
*type = SampleType_Float32;
else
fail(std::string("Unsupported sample format: ")+
av_get_sample_fmt_name((*mStream)->codec->sample_fmt));
int64_t ch_layout = (*mStream)->codec->channel_layout;
if((*mStream)->codec->channel_layout == AV_CH_LAYOUT_MONO)
*chans = ChannelConfig_Mono;
else if((*mStream)->codec->channel_layout == AV_CH_LAYOUT_STEREO)
*chans = ChannelConfig_Stereo;
else if((*mStream)->codec->channel_layout == AV_CH_LAYOUT_QUAD)
*chans = ChannelConfig_Quad;
else if((*mStream)->codec->channel_layout == AV_CH_LAYOUT_5POINT1)
*chans = ChannelConfig_5point1;
else if((*mStream)->codec->channel_layout == AV_CH_LAYOUT_7POINT1)
*chans = ChannelConfig_7point1;
else if((*mStream)->codec->channel_layout == 0)
{
/* Unknown channel layout. Try to guess. */
if((*mStream)->codec->channels == 1)
{
*chans = ChannelConfig_Mono;
ch_layout = AV_CH_LAYOUT_MONO;
}
else if((*mStream)->codec->channels == 2)
{
*chans = ChannelConfig_Stereo;
ch_layout = AV_CH_LAYOUT_STEREO;
}
else
{
std::stringstream sstr("Unsupported raw channel count: ");
sstr << (*mStream)->codec->channels;
fail(sstr.str());
}
}
else
{
char str[1024];
av_get_channel_layout_string(str, sizeof(str), (*mStream)->codec->channels,
(*mStream)->codec->channel_layout);
fail(std::string("Unsupported channel layout: ")+str);
}
*samplerate = (*mStream)->codec->sample_rate;
if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_U8P)
mOutputSampleFormat = AV_SAMPLE_FMT_U8;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_S16P)
mOutputSampleFormat = AV_SAMPLE_FMT_S16;
else if((*mStream)->codec->sample_fmt == AV_SAMPLE_FMT_FLTP)
mOutputSampleFormat = AV_SAMPLE_FMT_FLT;
if(mOutputSampleFormat != AV_SAMPLE_FMT_NONE)
{
mSwr = swr_alloc_set_opts(mSwr, // SwrContext
ch_layout, // output ch layout
mOutputSampleFormat, // output sample format
(*mStream)->codec->sample_rate, // output sample rate
ch_layout, // input ch layout
(*mStream)->codec->sample_fmt, // input sample format
(*mStream)->codec->sample_rate, // input sample rate
0, // logging level offset
NULL); // log context
if(!mSwr)
fail(std::string("Couldn't allocate SwrContext"));
if(swr_init(mSwr) < 0)
fail(std::string("Couldn't initialize SwrContext"));
}
}
size_t FFmpeg_Decoder::read(char *buffer, size_t bytes)
{
if(!mStream)
fail("No audio stream");
return readAVAudioData(buffer, bytes);
}
void FFmpeg_Decoder::readAll(std::vector<char> &output)
{
if(!mStream)
fail("No audio stream");
while(getAVAudioData())
{
size_t got = mFrame->nb_samples * (*mStream)->codec->channels *
av_get_bytes_per_sample(mOutputSampleFormat);
const char *inbuf = reinterpret_cast<char*>(mFrameData[0]);
output.insert(output.end(), inbuf, inbuf+got);
}
}
void FFmpeg_Decoder::rewind()
{
int stream_idx = mStream - mFormatCtx->streams;
if(av_seek_frame(mFormatCtx, stream_idx, 0, 0) < 0)
fail("Failed to seek in audio stream");
av_free_packet(&mPacket);
mFrameSize = mFramePos = 0;
mNextPts = 0.0;
}
size_t FFmpeg_Decoder::getSampleOffset()
{
int delay = (mFrameSize-mFramePos) / (*mStream)->codec->channels /
av_get_bytes_per_sample(mOutputSampleFormat);
return (int)(mNextPts*(*mStream)->codec->sample_rate) - delay;
}
FFmpeg_Decoder::FFmpeg_Decoder()
: mFormatCtx(NULL)
, mStream(NULL)
, mFrame(NULL)
, mFrameSize(0)
, mFramePos(0)
, mNextPts(0.0)
, mSwr(0)
, mOutputSampleFormat(AV_SAMPLE_FMT_NONE)
, mDataBuf(NULL)
, mFrameData(NULL)
, mDataBufLen(0)
{
memset(&mPacket, 0, sizeof(mPacket));
/* We need to make sure ffmpeg is initialized. Optionally silence warning
* output from the lib */
static bool done_init = false;
if(!done_init)
{
av_register_all();
av_log_set_level(AV_LOG_ERROR);
done_init = true;
}
}
FFmpeg_Decoder::~FFmpeg_Decoder()
{
close();
}
}