frodovdr/vdr-plugin-softhddevice
1
0
Fork 0
mirror of https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git synced 2023-10-10 19:16:51 +02:00
Code Issues Releases Wiki Activity

Only a single frame is supported.

Browse Source
This commit is contained in:
Johns 2012-02-24 15:38:04 +01:00
parent 07b426f2b5
commit 762959fbb4
1 changed files with 172 additions and 175 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

347
codec.c
View File

@ -667,6 +667,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
int codec_id) int codec_id)
{ {
AVCodec *audio_codec; AVCodec *audio_codec;
AVDictionary *av_dict;
if (name && (audio_codec = avcodec_find_decoder_by_name(name))) { if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: audio decoder '%s' found\n", 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")); Fatal(_("codec: can't open audio codec\n"));
} }
#else #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); pthread_mutex_unlock(&CodecLockMutex);
Fatal(_("codec: can't open audio codec\n")); Fatal(_("codec: can't open audio codec\n"));
} }
av_dict_free(&av_dict);
#endif #endif
pthread_mutex_unlock(&CodecLockMutex); pthread_mutex_unlock(&CodecLockMutex);
Debug(3, "codec: audio '%s'\n", audio_decoder->AudioCtx->codec_name); 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 + int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16))); FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
int buf_sz;
int l;
AVCodecContext *audio_ctx; AVCodecContext *audio_ctx;
int index;
audio_ctx = audio_decoder->AudioCtx; audio_ctx = audio_decoder->AudioCtx;
index = 0;
while (avpkt->size > index) {
int l;
int buf_sz;
buf_sz = sizeof(buf); buf_sz = sizeof(buf);
l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt); l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
if (avpkt->size != l) {
if (l == AVERROR(EAGAIN)) { if (l == AVERROR(EAGAIN)) {
Error(_("codec: latm\n")); Error(_("codec: latm\n"));
break; return;
} }
if (l < 0) { // no audio frame could be decompressed if (l < 0) { // no audio frame could be decompressed
Error(_("codec: error audio data at %d\n"), index); Error(_("codec: error audio data\n"));
break; return;
} }
Error(_("codec: error more than one frame data\n"));
}
#ifdef notyetFF_API_OLD_DECODE_AUDIO #ifdef notyetFF_API_OLD_DECODE_AUDIO
// FIXME: ffmpeg git comeing // FIXME: ffmpeg git comeing
int got_frame; int got_frame;
avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt); avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
#else #else
#endif #endif
// Update audio clock // Update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) { if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
AudioSetClock(avpkt->pts); 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; audio_decoder->SampleRate = audio_ctx->sample_rate;
// FIXME: use swr_convert from swresample (only in ffmpeg!) audio_decoder->HwSampleRate = audio_ctx->sample_rate;
if (audio_decoder->ReSample) { audio_decoder->Channels = audio_ctx->channels;
audio_resample_close(audio_decoder->ReSample); // SPDIF/HDMI passthrough
audio_decoder->ReSample = NULL; 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; // channels not support?
audio_decoder->HwSampleRate = audio_ctx->sample_rate; if ((err =
audio_decoder->Channels = audio_ctx->channels; AudioSetup(&audio_decoder->HwSampleRate,
// SPDIF/HDMI passthrough &audio_decoder->HwChannels, isAC3))) {
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) { Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
audio_decoder->HwChannels = 2; audio_ctx->sample_rate, audio_ctx->channels,
isAC3 = 1; audio_decoder->HwSampleRate, audio_decoder->HwChannels);
} else {
audio_decoder->HwChannels = audio_ctx->channels;
isAC3 = 0;
}
// channels not support? if (err == 1) {
if ((err = audio_decoder->ReSample =
AudioSetup(&audio_decoder->HwSampleRate, av_audio_resample_init(audio_decoder->HwChannels,
&audio_decoder->HwChannels, isAC3))) { audio_ctx->channels, audio_decoder->HwSampleRate,
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n", audio_ctx->sample_rate, audio_ctx->sample_fmt,
audio_ctx->sample_rate, audio_ctx->channels, audio_ctx->sample_fmt, 16, 10, 0, 0.8);
audio_decoder->HwSampleRate, audio_decoder->HwChannels); // libav-0.8_pre didn't support 6 -> 2 channels
if (!audio_decoder->ReSample) {
if (err == 1) { Error(_("codec/audio: resample setup error\n"));
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
audio_decoder->HwChannels = 0; audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0; audio_decoder->HwSampleRate = 0;
break;
}
}
}
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");
}
#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 { } else {
#ifdef USE_PASSTHROUGH Debug(3, "codec/audio: audio setup error\n");
// SPDIF/HDMI passthrough // FIXME: handle errors
if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) { audio_decoder->HwChannels = 0;
// build SPDIF header and append A52 audio to it audio_decoder->HwSampleRate = 0;
// avpkt is the original data return;
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);
}
#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
}
// DTS HD?
// True HD?
#endif
#endif
CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
AudioEnqueue(buf, buf_sz);
} }
} }
}
if (avpkt->size > l) { if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
Error(_("codec: error more than one frame data\n")); // 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");
}
#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 {
#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"));
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
}
// DTS HD?
// True HD?
#endif
#endif
CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
AudioEnqueue(buf, buf_sz);
} }
index += l;
} }
} }