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fix random fixed buffer size and add quick and simple planar import support

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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Benjamin Drung <bdrung@debian.org>
This commit is contained in:
benjamin.drung@gmail.com
2014-05-23 20:27:54 +00:00
parent f8be1f9668
commit 5ae8679e3c
3 changed files with 62 additions and 31 deletions
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69
src/FFmpeg.cpp
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@@ -478,10 +478,32 @@ int import_ffmpeg_decode_frame(streamContext *sc, bool flushing)
}
}
AVPacket avpkt;
av_init_packet(&avpkt);
avpkt.data = pDecode;
avpkt.size = nDecodeSiz;
AVFrame *frame = av_frame_alloc();
int got_output = 0;
nBytesDecoded =
avcodec_decode_audio4(sc->m_codecCtx,
frame, // out
&got_output, // out
&avpkt); // in
sc->m_decodedAudioSamplesValidSiz = frame->nb_samples;
if (nBytesDecoded < 0)
{
// Decoding failed. Don't stop.
return -1;
}
sc->m_samplefmt = sc->m_codecCtx->sample_fmt;
sc->m_samplesize = av_get_bits_per_sample_format(sc->m_samplefmt) / 8;
unsigned int newsize = FFMAX(sc->m_pkt.size * sc->m_samplesize, 192000); //FIXME
int channels = sc->m_codecCtx->channels;
unsigned int newsize = sc->m_samplesize * frame->nb_samples * channels;
// Reallocate the audio sample buffer if it's smaller than the frame size.
if (newsize > sc->m_decodedAudioSamplesSiz )
{
@@ -499,38 +521,20 @@ int import_ffmpeg_decode_frame(streamContext *sc, bool flushing)
return -1;
}
}
sc->m_decodedAudioSamplesValidSiz = sc->m_decodedAudioSamplesSiz;
#if LIBAVCODEC_VERSION_INT > AV_VERSION_INT(52, 25, 0)
// avcodec_decode_audio3() expects the size of the output buffer as the 3rd parameter but
// also returns the number of bytes it decoded in the same parameter.
AVPacket avpkt;
av_init_packet(&avpkt);
avpkt.data = pDecode;
avpkt.size = nDecodeSiz;
nBytesDecoded =
avcodec_decode_audio3(sc->m_codecCtx,
(int16_t *)sc->m_decodedAudioSamples, // out
&sc->m_decodedAudioSamplesValidSiz, // in/out
&avpkt); // in
#else
// avcodec_decode_audio2() expects the size of the output buffer as the 3rd parameter but
// also returns the number of bytes it decoded in the same parameter.
nBytesDecoded =
avcodec_decode_audio2(sc->m_codecCtx,
(int16_t *) sc->m_decodedAudioSamples, // out
&sc->m_decodedAudioSamplesValidSiz, // in/out
pDecode, // in
nDecodeSiz); // in
#endif
if (nBytesDecoded < 0)
{
// Decoding failed. Don't stop.
return -1;
if (frame->data[1]) {
for (int i = 0; i<frame->nb_samples; i++) {
for (int ch = 0; ch<channels; ch++) {
memcpy(sc->m_decodedAudioSamples + sc->m_samplesize * (ch + channels*i),
frame->extended_data[ch] + sc->m_samplesize*i,
sc->m_samplesize);
}
}
} else {
memcpy(sc->m_decodedAudioSamples, frame->data[0], newsize);
}
av_frame_free(&frame);
// We may not have read all of the data from this packet. If so, the user can call again.
// Whether or not they do depends on if m_pktRemainingSiz == 0 (they can check).
sc->m_pktDataPtr += nBytesDecoded;
@@ -1015,6 +1019,7 @@ bool FFmpegLibs::InitLibs(wxString libpath_format, bool WXUNUSED(showerr))
FFMPEG_INITDYN(avcodec, avcodec_find_encoder_by_name);
FFMPEG_INITDYN(avcodec, avcodec_find_decoder);
FFMPEG_INITDYN(avcodec, avcodec_open2);
FFMPEG_INITDYN(avcodec, avcodec_decode_audio4);
#if LIBAVCODEC_VERSION_INT > AV_VERSION_INT(52, 25, 0)
FFMPEG_INITDYN(avcodec, avcodec_decode_audio3);
#else
@@ -1047,6 +1052,8 @@ bool FFmpegLibs::InitLibs(wxString libpath_format, bool WXUNUSED(showerr))
FFMPEG_INITDYN(avutil, av_freep);
FFMPEG_INITDYN(avutil, av_rescale_q);
FFMPEG_INITDYN(avutil, avutil_version);
FFMPEG_INITDYN(avutil, av_frame_alloc);
FFMPEG_INITDYN(avutil, av_frame_free);
wxLogMessage(wxT("All symbols loaded successfully. Initializing the library."));
#endif