mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Only a single frame is supported.
This commit is contained in:
Johns
parent
07b426f2b5
commit
762959fbb4
347
codec.c
347
codec.c
@ -667,6 +667,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
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int codec_id)
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{
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AVCodec *audio_codec;
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AVDictionary *av_dict;
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if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
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Debug(3, "codec: audio decoder '%s' found\n", name);
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@ -693,10 +694,15 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
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Fatal(_("codec: can't open audio codec\n"));
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}
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#else
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if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, NULL) < 0) {
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av_dict = NULL;
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//av_dict_set(&av_dict, "dmix_mode", "0", 0);
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//av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
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//av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
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if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
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pthread_mutex_unlock(&CodecLockMutex);
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Fatal(_("codec: can't open audio codec\n"));
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}
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av_dict_free(&av_dict);
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#endif
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pthread_mutex_unlock(&CodecLockMutex);
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Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name);
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@ -1095,205 +1101,196 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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{
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int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
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int buf_sz;
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int l;
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AVCodecContext *audio_ctx;
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int index;
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audio_ctx = audio_decoder->AudioCtx;
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index = 0;
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while (avpkt->size > index) {
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int l;
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int buf_sz;
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buf_sz = sizeof(buf);
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l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
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buf_sz = sizeof(buf);
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l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
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if (avpkt->size != l) {
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if (l == AVERROR(EAGAIN)) {
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Error(_("codec: latm\n"));
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break;
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return;
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}
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if (l < 0) { // no audio frame could be decompressed
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Error(_("codec: error audio data at %d\n"), index);
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break;
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Error(_("codec: error audio data\n"));
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return;
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}
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Error(_("codec: error more than one frame data\n"));
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}
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#ifdef notyetFF_API_OLD_DECODE_AUDIO
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// FIXME: ffmpeg git comeing
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int got_frame;
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// FIXME: ffmpeg git comeing
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int got_frame;
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avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
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avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
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#else
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#endif
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// Update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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AudioSetClock(avpkt->pts);
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// Update audio clock
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if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
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AudioSetClock(avpkt->pts);
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}
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// FIXME: must first play remainings bytes, than change and play new.
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if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
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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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int err;
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int isAC3;
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audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
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// FIXME: use swr_convert from swresample (only in ffmpeg!)
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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audio_decoder->ReSample = NULL;
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}
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// FIXME: must first play remainings bytes, than change and play new.
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if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
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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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int err;
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int isAC3;
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audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
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// FIXME: use swr_convert from swresample (only in ffmpeg!)
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if (audio_decoder->ReSample) {
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audio_resample_close(audio_decoder->ReSample);
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audio_decoder->ReSample = NULL;
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}
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audio_decoder->SampleRate = audio_ctx->sample_rate;
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audio_decoder->HwSampleRate = audio_ctx->sample_rate;
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audio_decoder->Channels = audio_ctx->channels;
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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audio_decoder->HwChannels = 2;
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isAC3 = 1;
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} else {
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audio_decoder->HwChannels = audio_ctx->channels;
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isAC3 = 0;
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}
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audio_decoder->SampleRate = audio_ctx->sample_rate;
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audio_decoder->HwSampleRate = audio_ctx->sample_rate;
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audio_decoder->Channels = audio_ctx->channels;
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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audio_decoder->HwChannels = 2;
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isAC3 = 1;
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} else {
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audio_decoder->HwChannels = audio_ctx->channels;
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isAC3 = 0;
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}
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// channels not support?
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if ((err =
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AudioSetup(&audio_decoder->HwSampleRate,
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&audio_decoder->HwChannels, isAC3))) {
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Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
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audio_ctx->sample_rate, audio_ctx->channels,
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audio_decoder->HwSampleRate, audio_decoder->HwChannels);
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// channels not support?
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if ((err =
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AudioSetup(&audio_decoder->HwSampleRate,
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&audio_decoder->HwChannels, isAC3))) {
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Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
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audio_ctx->sample_rate, audio_ctx->channels,
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audio_decoder->HwSampleRate, audio_decoder->HwChannels);
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if (err == 1) {
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audio_decoder->ReSample =
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av_audio_resample_init(audio_decoder->HwChannels,
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audio_ctx->channels, audio_decoder->HwSampleRate,
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audio_ctx->sample_rate, audio_ctx->sample_fmt,
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audio_ctx->sample_fmt, 16, 10, 0, 0.8);
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// libav-0.8_pre didn't support 6 -> 2 channels
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if (!audio_decoder->ReSample) {
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Error(_("codec/audio: resample setup error\n"));
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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}
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} else {
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Debug(3, "codec/audio: audio setup error\n");
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// FIXME: handle errors
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if (err == 1) {
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audio_decoder->ReSample =
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av_audio_resample_init(audio_decoder->HwChannels,
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audio_ctx->channels, audio_decoder->HwSampleRate,
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audio_ctx->sample_rate, audio_ctx->sample_fmt,
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audio_ctx->sample_fmt, 16, 10, 0, 0.8);
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// libav-0.8_pre didn't support 6 -> 2 channels
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if (!audio_decoder->ReSample) {
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Error(_("codec/audio: resample setup error\n"));
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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break;
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}
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}
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}
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if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
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// need to resample audio
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if (audio_decoder->ReSample) {
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int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE]
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__attribute__ ((aligned(16)));
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int outlen;
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// FIXME: libav-0.7.2 crash here
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outlen =
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audio_resample(audio_decoder->ReSample, outbuf, buf,
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buf_sz);
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#ifdef DEBUG
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if (outlen != buf_sz) {
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Debug(3, "codec/audio: possible fixed ffmpeg\n");
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}
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#endif
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if (outlen) {
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// outlen seems to be wrong in ffmpeg-0.9
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outlen /= audio_decoder->Channels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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outlen *=
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audio_decoder->HwChannels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
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CodecReorderAudioFrame(outbuf, outlen,
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audio_decoder->HwChannels);
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AudioEnqueue(outbuf, outlen);
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}
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} else {
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#ifdef USE_PASSTHROUGH
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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// build SPDIF header and append A52 audio to it
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// avpkt is the original data
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buf_sz = 6144;
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if (buf_sz < avpkt->size + 8) {
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Error(_
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("codec/audio: decoded data smaller than encoded\n"));
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break;
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}
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// copy original data for output
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// FIXME: not 100% sure, if endian is correct
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buf[0] = htole16(0xF872); // iec 61937 sync word
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buf[1] = htole16(0x4E1F);
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buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
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buf[3] = htole16(avpkt->size * 8);
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swab(avpkt->data, buf + 4, avpkt->size);
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memset(buf + 4 + avpkt->size / 2, 0,
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buf_sz - 8 - avpkt->size);
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}
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#if 0
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//
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// old experimental code
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//
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if (1) {
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// FIXME: need to detect dts
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// copy original data for output
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// FIXME: buf is sint
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buf[0] = 0x72;
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buf[1] = 0xF8;
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buf[2] = 0x1F;
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buf[3] = 0x4E;
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buf[4] = 0x00;
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switch (avpkt->size) {
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case 512:
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buf[5] = 0x0B;
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break;
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case 1024:
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buf[5] = 0x0C;
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break;
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case 2048:
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buf[5] = 0x0D;
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break;
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default:
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Debug(3,
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"codec/audio: dts sample burst not supported\n");
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buf[5] = 0x00;
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break;
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}
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buf[6] = (avpkt->size * 8);
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buf[7] = (avpkt->size * 8) >> 8;
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//buf[8] = 0x0B;
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//buf[9] = 0x77;
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//printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
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// swab?
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memcpy(buf + 8, avpkt->data, avpkt->size);
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memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
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} else if (1) {
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// FIXME: need to detect mp2
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// FIXME: mp2 passthrough
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// see softhddev.c version/layer
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// 0x04 mpeg1 layer1
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// 0x05 mpeg1 layer23
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// 0x06 mpeg2 ext
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// 0x07 mpeg2.5 layer 1
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// 0x08 mpeg2.5 layer 2
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// 0x09 mpeg2.5 layer 3
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}
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// DTS HD?
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// True HD?
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#endif
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#endif
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CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
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AudioEnqueue(buf, buf_sz);
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Debug(3, "codec/audio: audio setup error\n");
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// FIXME: handle errors
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audio_decoder->HwChannels = 0;
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audio_decoder->HwSampleRate = 0;
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return;
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}
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}
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}
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if (avpkt->size > l) {
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Error(_("codec: error more than one frame data\n"));
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if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
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// need to resample audio
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if (audio_decoder->ReSample) {
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int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
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FF_INPUT_BUFFER_PADDING_SIZE]
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__attribute__ ((aligned(16)));
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int outlen;
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// FIXME: libav-0.7.2 crash here
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outlen =
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audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
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#ifdef DEBUG
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if (outlen != buf_sz) {
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Debug(3, "codec/audio: possible fixed ffmpeg\n");
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}
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#endif
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if (outlen) {
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// outlen seems to be wrong in ffmpeg-0.9
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outlen /= audio_decoder->Channels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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outlen *=
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audio_decoder->HwChannels *
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av_get_bytes_per_sample(audio_ctx->sample_fmt);
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Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
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CodecReorderAudioFrame(outbuf, outlen,
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audio_decoder->HwChannels);
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AudioEnqueue(outbuf, outlen);
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}
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} else {
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#ifdef USE_PASSTHROUGH
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// SPDIF/HDMI passthrough
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if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
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// build SPDIF header and append A52 audio to it
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// avpkt is the original data
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buf_sz = 6144;
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if (buf_sz < avpkt->size + 8) {
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Error(_
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("codec/audio: decoded data smaller than encoded\n"));
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return;
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}
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// copy original data for output
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// FIXME: not 100% sure, if endian is correct
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buf[0] = htole16(0xF872); // iec 61937 sync word
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buf[1] = htole16(0x4E1F);
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buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
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buf[3] = htole16(avpkt->size * 8);
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swab(avpkt->data, buf + 4, avpkt->size);
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memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
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}
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#if 0
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//
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// old experimental code
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//
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if (1) {
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// FIXME: need to detect dts
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// copy original data for output
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// FIXME: buf is sint
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buf[0] = 0x72;
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buf[1] = 0xF8;
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buf[2] = 0x1F;
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buf[3] = 0x4E;
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buf[4] = 0x00;
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switch (avpkt->size) {
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case 512:
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buf[5] = 0x0B;
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break;
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case 1024:
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buf[5] = 0x0C;
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break;
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case 2048:
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buf[5] = 0x0D;
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break;
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default:
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Debug(3,
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"codec/audio: dts sample burst not supported\n");
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buf[5] = 0x00;
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break;
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}
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buf[6] = (avpkt->size * 8);
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buf[7] = (avpkt->size * 8) >> 8;
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//buf[8] = 0x0B;
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//buf[9] = 0x77;
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//printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
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// swab?
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memcpy(buf + 8, avpkt->data, avpkt->size);
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memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
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} else if (1) {
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// FIXME: need to detect mp2
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// FIXME: mp2 passthrough
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// see softhddev.c version/layer
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// 0x04 mpeg1 layer1
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// 0x05 mpeg1 layer23
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// 0x06 mpeg2 ext
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// 0x07 mpeg2.5 layer 1
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// 0x08 mpeg2.5 layer 2
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// 0x09 mpeg2.5 layer 3
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}
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// DTS HD?
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// True HD?
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#endif
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#endif
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CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
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AudioEnqueue(buf, buf_sz);
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}
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index += l;
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}
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}
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