diff options
| -rw-r--r-- | ChangeLog | 1 | ||||
| -rw-r--r-- | codec.c | 40 |
2 files changed, 33 insertions, 8 deletions
@@ -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. @@ -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)) { |