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https://github.com/jojo61/vdr-plugin-softhdcuvid.git
synced 2023-10-10 13:37:41 +02:00
optimized CodecAudioDecode
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jojo61
parent
33a3316344
commit
24ccfefff3
192
codec.c
192
codec.c
@ -1687,156 +1687,62 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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** @param audio_decoder audio decoder data
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** @param avpkt audio packet
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*/
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void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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{
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AVCodecContext *audio_ctx;
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AVCodecContext *audio_ctx = audio_decoder->AudioCtx;
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(56,28,1)
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AVFrame frame[1];
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#else
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AVFrame *frame;
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#endif
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int got_frame;
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int n,ret;
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if (audio_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
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int ret;
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AVPacket pkt[1];
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AVFrame *frame = audio_decoder->Frame;
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audio_ctx = audio_decoder->AudioCtx;
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// FIXME: don't need to decode pass-through codecs
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// new AVFrame API
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(56,28,1)
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avcodec_get_frame_defaults(frame);
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#else
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frame = audio_decoder->Frame;
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av_frame_unref(frame);
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#endif
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got_frame = 0;
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#if 0
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n = avcodec_decode_audio4(audio_ctx, frame, &got_frame, (AVPacket *) avpkt);
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#else
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// SUGGESTION
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// Now that avcodec_decode_audio4 is deprecated and replaced
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// by 2 calls (receive frame and send packet), this could be optimized
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// into separate routines or separate threads.
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// Also now that it always consumes a whole buffer some code
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// in the caller may be able to be optimized.
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ret = avcodec_receive_frame(audio_ctx,frame);
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if (ret == 0)
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got_frame = 1;
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if (ret == AVERROR(EAGAIN))
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ret = 0;
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if (ret == 0)
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ret = avcodec_send_packet(audio_ctx, avpkt);
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if (ret == AVERROR(EAGAIN))
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ret = 0;
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else if (ret < 0)
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{
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// Debug(3, "codec/audio: audio decode error: %1 (%2)\n",av_make_error_string(error, sizeof(error), ret),got_frame);
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return;
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}
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else
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ret = avpkt->size;
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n = ret; //FIXME: why n and not ret??
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#endif
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if (n != avpkt->size) {
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if (n == AVERROR(EAGAIN)) {
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Error(_("codec/audio: latm\n"));
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return;
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}
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if (n < 0) { // no audio frame could be decompressed
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Error(_("codec/audio: bad audio frame\n"));
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return;
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}
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Error(_("codec/audio: error more than one frame data\n"));
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}
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if (!got_frame) {
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Error(_("codec/audio: no frame\n"));
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return;
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}
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// update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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CodecAudioSetClock(audio_decoder, avpkt->pts);
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}
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// format change
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if (audio_decoder->Passthrough != CodecPassthrough
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|| audio_decoder->SampleRate != audio_ctx->sample_rate
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|| audio_decoder->Channels != audio_ctx->channels) {
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CodecAudioUpdateFormat(audio_decoder);
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}
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if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
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return; // unsupported sample format
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}
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if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
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return;
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}
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if (0) {
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char strbuf[32];
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int data_sz;
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int plane_sz;
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data_sz =
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av_samples_get_buffer_size(&plane_sz, audio_ctx->channels,
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frame->nb_samples, audio_ctx->sample_fmt, 1);
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fprintf(stderr, "codec/audio: sample_fmt %s\n",
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av_get_sample_fmt_name(audio_ctx->sample_fmt));
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av_get_channel_layout_string(strbuf, 32, audio_ctx->channels,
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audio_ctx->channel_layout);
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fprintf(stderr, "codec/audio: layout %s\n", strbuf);
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fprintf(stderr,
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"codec/audio: channels %d samples %d plane %d data %d\n",
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audio_ctx->channels, frame->nb_samples, plane_sz, data_sz);
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}
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#ifdef USE_SWRESAMPLE
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if (audio_decoder->Resample) {
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uint8_t outbuf[8192 * 2 * 8];
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uint8_t *out[1];
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out[0] = outbuf;
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n = swr_convert(audio_decoder->Resample, out,
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sizeof(outbuf) / (2 * audio_decoder->HwChannels),
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(const uint8_t **)frame->extended_data, frame->nb_samples);
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if (n > 0) {
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if (!(audio_decoder->Passthrough & CodecPCM)) {
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CodecReorderAudioFrame((int16_t *) outbuf,
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n * 2 * audio_decoder->HwChannels,
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audio_decoder->HwChannels);
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}
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AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
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av_frame_unref(frame);
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*pkt = *avpkt; // use copy
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ret = avcodec_send_packet(audio_ctx, pkt);
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if (ret < 0) {
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Debug(3, "codec: sending audio packet failed");
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return;
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}
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return;
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}
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#endif
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#ifdef USE_AVRESAMPLE
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if (audio_decoder->Resample) {
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uint8_t outbuf[8192 * 2 * 8];
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uint8_t *out[1];
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out[0] = outbuf;
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n = avresample_convert(audio_decoder->Resample, out, 0,
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sizeof(outbuf) / (2 * audio_decoder->HwChannels),
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(uint8_t **) frame->extended_data, 0, frame->nb_samples);
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// FIXME: set out_linesize, in_linesize correct
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if (n > 0) {
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if (!(audio_decoder->Passthrough & CodecPCM)) {
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CodecReorderAudioFrame((int16_t *) outbuf,
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n * 2 * audio_decoder->HwChannels,
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audio_decoder->HwChannels);
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}
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AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels);
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ret = avcodec_receive_frame(audio_ctx, frame);
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if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
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Debug(3, "codec: receiving audio frame failed");
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return;
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}
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return;
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}
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#endif
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if (ret >= 0) {
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// update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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CodecAudioSetClock(audio_decoder, avpkt->pts);
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}
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// format change
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if (audio_decoder->Passthrough != CodecPassthrough || audio_decoder->SampleRate != audio_ctx->sample_rate
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|| audio_decoder->Channels != audio_ctx->channels) {
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CodecAudioUpdateFormat(audio_decoder);
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}
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if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) {
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return; // unsupported sample format
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}
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if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
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return;
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}
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if (audio_decoder->Resample) {
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uint8_t outbuf[8192 * 2 * 8];
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uint8_t *out[1];
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#ifdef DEBUG
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// should be never reached
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fprintf(stderr, "oops\n");
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#endif
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out[0] = outbuf;
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ret = swr_convert(audio_decoder->Resample, out, sizeof(outbuf) / (2 * audio_decoder->HwChannels),
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(const uint8_t **)frame->extended_data, frame->nb_samples);
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if (ret > 0) {
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if (!(audio_decoder->Passthrough & CodecPCM)) {
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CodecReorderAudioFrame((int16_t *) outbuf, ret * 2 * audio_decoder->HwChannels,
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audio_decoder->HwChannels);
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}
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AudioEnqueue(outbuf, ret * 2 * audio_decoder->HwChannels);
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}
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return;
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}
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}
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}
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}
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#endif
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