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vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Support ffmpeg new AVFrame API in the audio codec.

This commit is contained in:
Johns 2014-08-13 12:04:39 +02:00
parent 4a4de36878
commit 8b7402a397
2 changed files with 33 additions and 8 deletions
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1
ChangeLog
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@ -1,6 +1,7 @@
User johns
Date:
Support ffmpeg new AVFrame API in the audio codec.
Config for automatic AES parameters.
Use GCC built-in functions for atomic operations.

40
codec.c
View File

@ -681,6 +681,10 @@ struct _audio_decoder_
int HwSampleRate; ///< hw sample rate
int HwChannels; ///< hw channels
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58,28,1)
AVFrame *Frame; ///< decoded audio frame buffer
#endif
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
ReSampleContext *ReSample; ///< old resampling context
#endif
@ -757,6 +761,11 @@ AudioDecoder *CodecAudioNewDecoder(void)
if (!(audio_decoder = calloc(1, sizeof(*audio_decoder)))) {
Fatal(_("codec: can't allocate audio decoder\n"));
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58,28,1)
if (!(audio_decoder->Frame = av_frame_alloc())) {
Fatal(_("codec: can't allocate audio decoder frame buffer\n"));
}
#endif
return audio_decoder;
}
@ -768,6 +777,9 @@ AudioDecoder *CodecAudioNewDecoder(void)
*/
void CodecAudioDelDecoder(AudioDecoder * decoder)
{
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(58,28,1)
av_frame_free(&decoder->Frame); // callee does checks
#endif
free(decoder);
}
@ -1790,18 +1802,30 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
AVCodecContext *audio_ctx;
AVFrame frame;
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58,28,1)
AVFrame frame[1];
#else
AVFrame *frame;
#endif
int got_frame;
int n;
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
// libav needs memset, frame.data[0] = NULL;
memset(&frame, 0, sizeof(frame));
// new AVFrame API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58,28,1)
avcodec_get_frame_defaults(frame);
#else
frame = audio_decoder->Frame;
av_frame_unref(frame);
#endif
got_frame = 0;
n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame,
n = avcodec_decode_audio4(audio_ctx, frame, &got_frame,
(AVPacket *) avpkt);
if (n != avpkt->size) {
if (n == AVERROR(EAGAIN)) {
Error(_("codec/audio: latm\n"));
@ -1843,7 +1867,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
data_sz =
av_samples_get_buffer_size(&plane_sz, audio_ctx->channels,
frame.nb_samples, audio_ctx->sample_fmt, 1);
frame->nb_samples, audio_ctx->sample_fmt, 1);
fprintf(stderr, "codec/audio: sample_fmt %s\n",
av_get_sample_fmt_name(audio_ctx->sample_fmt));
av_get_channel_layout_string(strbuf, 32, audio_ctx->channels,
@ -1851,7 +1875,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
fprintf(stderr, "codec/audio: layout %s\n", strbuf);
fprintf(stderr,
"codec/audio: channels %d samples %d plane %d data %d\n",
audio_ctx->channels, frame.nb_samples, plane_sz, data_sz);
audio_ctx->channels, frame->nb_samples, plane_sz, data_sz);
}
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample) {
@ -1861,7 +1885,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
out[0] = outbuf;
n = swr_convert(audio_decoder->Resample, out,
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(const uint8_t **)frame.extended_data, frame.nb_samples);
(const uint8_t **)frame->extended_data, frame->nb_samples);
if (n > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf,
@ -1882,7 +1906,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
out[0] = outbuf;
n = avresample_convert(audio_decoder->Resample, out, 0,
sizeof(outbuf) / (2 * audio_decoder->HwChannels),
(uint8_t **) frame.extended_data, 0, frame.nb_samples);
(uint8_t **) frame->extended_data, 0, frame->nb_samples);
// FIXME: set out_linesize, in_linesize correct
if (n > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {