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-rw-r--r--codec.c335
1 files changed, 166 insertions, 169 deletions
diff --git a/codec.c b/codec.c
index ed9d858..9000b29 100644
--- a/codec.c
+++ b/codec.c
@@ -667,6 +667,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
int codec_id)
{
AVCodec *audio_codec;
+ AVDictionary *av_dict;
if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: audio decoder '%s' found\n", name);
@@ -693,10 +694,15 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
Fatal(_("codec: can't open audio codec\n"));
}
#else
- if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, NULL) < 0) {
+ av_dict = NULL;
+ //av_dict_set(&av_dict, "dmix_mode", "0", 0);
+ //av_dict_set(&av_dict, "ltrt_cmixlev", "1.414", 0);
+ //av_dict_set(&av_dict, "loro_cmixlev", "1.414", 0);
+ if (avcodec_open2(audio_decoder->AudioCtx, audio_codec, &av_dict) < 0) {
pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open audio codec\n"));
}
+ av_dict_free(&av_dict);
#endif
pthread_mutex_unlock(&CodecLockMutex);
Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name);
@@ -1095,205 +1101,196 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
+ int buf_sz;
+ int l;
AVCodecContext *audio_ctx;
- int index;
audio_ctx = audio_decoder->AudioCtx;
- index = 0;
- while (avpkt->size > index) {
- int l;
- int buf_sz;
- buf_sz = sizeof(buf);
- l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
+ buf_sz = sizeof(buf);
+ l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
+ if (avpkt->size != l) {
if (l == AVERROR(EAGAIN)) {
Error(_("codec: latm\n"));
- break;
+ return;
}
if (l < 0) { // no audio frame could be decompressed
- Error(_("codec: error audio data at %d\n"), index);
- break;
+ Error(_("codec: error audio data\n"));
+ return;
}
+ Error(_("codec: error more than one frame data\n"));
+ }
#ifdef notyetFF_API_OLD_DECODE_AUDIO
- // FIXME: ffmpeg git comeing
- int got_frame;
+ // FIXME: ffmpeg git comeing
+ int got_frame;
- avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
+ avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
#else
#endif
- // Update audio clock
- if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
- AudioSetClock(avpkt->pts);
+ // Update audio clock
+ if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
+ AudioSetClock(avpkt->pts);
+ }
+ // FIXME: must first play remainings bytes, than change and play new.
+ if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+ || audio_decoder->SampleRate != audio_ctx->sample_rate
+ || audio_decoder->Channels != audio_ctx->channels) {
+ int err;
+ int isAC3;
+
+ audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
+ // FIXME: use swr_convert from swresample (only in ffmpeg!)
+ if (audio_decoder->ReSample) {
+ audio_resample_close(audio_decoder->ReSample);
+ audio_decoder->ReSample = NULL;
}
- // FIXME: must first play remainings bytes, than change and play new.
- if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
- || audio_decoder->SampleRate != audio_ctx->sample_rate
- || audio_decoder->Channels != audio_ctx->channels) {
- int err;
- int isAC3;
-
- audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
- // FIXME: use swr_convert from swresample (only in ffmpeg!)
- if (audio_decoder->ReSample) {
- audio_resample_close(audio_decoder->ReSample);
- audio_decoder->ReSample = NULL;
- }
- audio_decoder->SampleRate = audio_ctx->sample_rate;
- audio_decoder->HwSampleRate = audio_ctx->sample_rate;
- audio_decoder->Channels = audio_ctx->channels;
- // SPDIF/HDMI passthrough
- if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
- audio_decoder->HwChannels = 2;
- isAC3 = 1;
- } else {
- audio_decoder->HwChannels = audio_ctx->channels;
- isAC3 = 0;
- }
+ audio_decoder->SampleRate = audio_ctx->sample_rate;
+ audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+ audio_decoder->Channels = audio_ctx->channels;
+ // SPDIF/HDMI passthrough
+ if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+ audio_decoder->HwChannels = 2;
+ isAC3 = 1;
+ } else {
+ audio_decoder->HwChannels = audio_ctx->channels;
+ isAC3 = 0;
+ }
- // channels not support?
- if ((err =
- AudioSetup(&audio_decoder->HwSampleRate,
- &audio_decoder->HwChannels, isAC3))) {
- Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
- audio_ctx->sample_rate, audio_ctx->channels,
- audio_decoder->HwSampleRate, audio_decoder->HwChannels);
-
- if (err == 1) {
- audio_decoder->ReSample =
- av_audio_resample_init(audio_decoder->HwChannels,
- audio_ctx->channels, audio_decoder->HwSampleRate,
- audio_ctx->sample_rate, audio_ctx->sample_fmt,
- audio_ctx->sample_fmt, 16, 10, 0, 0.8);
- // libav-0.8_pre didn't support 6 -> 2 channels
- if (!audio_decoder->ReSample) {
- Error(_("codec/audio: resample setup error\n"));
- audio_decoder->HwChannels = 0;
- audio_decoder->HwSampleRate = 0;
- }
- } else {
- Debug(3, "codec/audio: audio setup error\n");
- // FIXME: handle errors
+ // channels not support?
+ if ((err =
+ AudioSetup(&audio_decoder->HwSampleRate,
+ &audio_decoder->HwChannels, isAC3))) {
+ Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
+ audio_ctx->sample_rate, audio_ctx->channels,
+ audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+
+ if (err == 1) {
+ audio_decoder->ReSample =
+ av_audio_resample_init(audio_decoder->HwChannels,
+ audio_ctx->channels, audio_decoder->HwSampleRate,
+ audio_ctx->sample_rate, audio_ctx->sample_fmt,
+ audio_ctx->sample_fmt, 16, 10, 0, 0.8);
+ // libav-0.8_pre didn't support 6 -> 2 channels
+ if (!audio_decoder->ReSample) {
+ Error(_("codec/audio: resample setup error\n"));
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
- break;
}
+ } else {
+ Debug(3, "codec/audio: audio setup error\n");
+ // FIXME: handle errors
+ audio_decoder->HwChannels = 0;
+ audio_decoder->HwSampleRate = 0;
+ return;
}
}
+ }
- if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
- // need to resample audio
- if (audio_decoder->ReSample) {
- int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
- FF_INPUT_BUFFER_PADDING_SIZE]
- __attribute__ ((aligned(16)));
- int outlen;
-
- // FIXME: libav-0.7.2 crash here
- outlen =
- audio_resample(audio_decoder->ReSample, outbuf, buf,
- buf_sz);
+ if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
+ // need to resample audio
+ if (audio_decoder->ReSample) {
+ int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
+ FF_INPUT_BUFFER_PADDING_SIZE]
+ __attribute__ ((aligned(16)));
+ int outlen;
+
+ // FIXME: libav-0.7.2 crash here
+ outlen =
+ audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
#ifdef DEBUG
- if (outlen != buf_sz) {
- Debug(3, "codec/audio: possible fixed ffmpeg\n");
- }
+ if (outlen != buf_sz) {
+ Debug(3, "codec/audio: possible fixed ffmpeg\n");
+ }
#endif
- if (outlen) {
- // outlen seems to be wrong in ffmpeg-0.9
- outlen /= audio_decoder->Channels *
- av_get_bytes_per_sample(audio_ctx->sample_fmt);
- outlen *=
- audio_decoder->HwChannels *
- av_get_bytes_per_sample(audio_ctx->sample_fmt);
- Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
- CodecReorderAudioFrame(outbuf, outlen,
- audio_decoder->HwChannels);
- AudioEnqueue(outbuf, outlen);
- }
- } else {
+ if (outlen) {
+ // outlen seems to be wrong in ffmpeg-0.9
+ outlen /= audio_decoder->Channels *
+ av_get_bytes_per_sample(audio_ctx->sample_fmt);
+ outlen *=
+ audio_decoder->HwChannels *
+ av_get_bytes_per_sample(audio_ctx->sample_fmt);
+ Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
+ CodecReorderAudioFrame(outbuf, outlen,
+ audio_decoder->HwChannels);
+ AudioEnqueue(outbuf, outlen);
+ }
+ } else {
#ifdef USE_PASSTHROUGH
- // SPDIF/HDMI passthrough
- if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
- // build SPDIF header and append A52 audio to it
- // avpkt is the original data
- buf_sz = 6144;
- if (buf_sz < avpkt->size + 8) {
- Error(_
- ("codec/audio: decoded data smaller than encoded\n"));
- break;
- }
- // copy original data for output
- // FIXME: not 100% sure, if endian is correct
- buf[0] = htole16(0xF872); // iec 61937 sync word
- buf[1] = htole16(0x4E1F);
- buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
- buf[3] = htole16(avpkt->size * 8);
- swab(avpkt->data, buf + 4, avpkt->size);
- memset(buf + 4 + avpkt->size / 2, 0,
- buf_sz - 8 - avpkt->size);
+ // SPDIF/HDMI passthrough
+ if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+ // build SPDIF header and append A52 audio to it
+ // avpkt is the original data
+ buf_sz = 6144;
+ if (buf_sz < avpkt->size + 8) {
+ Error(_
+ ("codec/audio: decoded data smaller than encoded\n"));
+ return;
}
+ // copy original data for output
+ // FIXME: not 100% sure, if endian is correct
+ buf[0] = htole16(0xF872); // iec 61937 sync word
+ buf[1] = htole16(0x4E1F);
+ buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
+ buf[3] = htole16(avpkt->size * 8);
+ swab(avpkt->data, buf + 4, avpkt->size);
+ memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
+ }
#if 0
- //
- // old experimental code
- //
- if (1) {
- // FIXME: need to detect dts
- // copy original data for output
- // FIXME: buf is sint
- buf[0] = 0x72;
- buf[1] = 0xF8;
- buf[2] = 0x1F;
- buf[3] = 0x4E;
- buf[4] = 0x00;
- switch (avpkt->size) {
- case 512:
- buf[5] = 0x0B;
- break;
- case 1024:
- buf[5] = 0x0C;
- break;
- case 2048:
- buf[5] = 0x0D;
- break;
- default:
- Debug(3,
- "codec/audio: dts sample burst not supported\n");
- buf[5] = 0x00;
- break;
- }
- buf[6] = (avpkt->size * 8);
- buf[7] = (avpkt->size * 8) >> 8;
- //buf[8] = 0x0B;
- //buf[9] = 0x77;
- //printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
- // swab?
- memcpy(buf + 8, avpkt->data, avpkt->size);
- memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
- } else if (1) {
- // FIXME: need to detect mp2
- // FIXME: mp2 passthrough
- // see softhddev.c version/layer
- // 0x04 mpeg1 layer1
- // 0x05 mpeg1 layer23
- // 0x06 mpeg2 ext
- // 0x07 mpeg2.5 layer 1
- // 0x08 mpeg2.5 layer 2
- // 0x09 mpeg2.5 layer 3
+ //
+ // old experimental code
+ //
+ if (1) {
+ // FIXME: need to detect dts
+ // copy original data for output
+ // FIXME: buf is sint
+ buf[0] = 0x72;
+ buf[1] = 0xF8;
+ buf[2] = 0x1F;
+ buf[3] = 0x4E;
+ buf[4] = 0x00;
+ switch (avpkt->size) {
+ case 512:
+ buf[5] = 0x0B;
+ break;
+ case 1024:
+ buf[5] = 0x0C;
+ break;
+ case 2048:
+ buf[5] = 0x0D;
+ break;
+ default:
+ Debug(3,
+ "codec/audio: dts sample burst not supported\n");
+ buf[5] = 0x00;
+ break;
}
- // DTS HD?
- // True HD?
+ buf[6] = (avpkt->size * 8);
+ buf[7] = (avpkt->size * 8) >> 8;
+ //buf[8] = 0x0B;
+ //buf[9] = 0x77;
+ //printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
+ // swab?
+ memcpy(buf + 8, avpkt->data, avpkt->size);
+ memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
+ } else if (1) {
+ // FIXME: need to detect mp2
+ // FIXME: mp2 passthrough
+ // see softhddev.c version/layer
+ // 0x04 mpeg1 layer1
+ // 0x05 mpeg1 layer23
+ // 0x06 mpeg2 ext
+ // 0x07 mpeg2.5 layer 1
+ // 0x08 mpeg2.5 layer 2
+ // 0x09 mpeg2.5 layer 3
+ }
+ // DTS HD?
+ // True HD?
#endif
#endif
- CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
- AudioEnqueue(buf, buf_sz);
- }
+ CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
+ AudioEnqueue(buf, buf_sz);
}
-
- if (avpkt->size > l) {
- Error(_("codec: error more than one frame data\n"));
- }
-
- index += l;
}
}
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