diff options
Diffstat (limited to 'codec.c')
| -rw-r--r-- | codec.c | 336 |
1 files changed, 278 insertions, 58 deletions
@@ -30,8 +30,10 @@ /// many bugs and incompatiblity in it. Don't use this shit. /// - /// compile with passthrough support (experimental) + /// compile with passthrough support (stable, ac3 only) #define USE_PASSTHROUGH + /// compile audio drift correction support (experimental) +#define noUSE_AUDIO_DRIFT_CORRECTION #include <stdio.h> #include <unistd.h> @@ -606,6 +608,22 @@ struct _audio_decoder_ int HwChannels; ///< hw channels ReSampleContext *ReSample; ///< audio resampling context + + int64_t StartPTS; ///< start PTS + struct timespec StartTime; ///< start time + int64_t LastPTS; ///< last PTS + + int Drift; ///< drift correction value +#define AVERAGE 10 ///< number of average values + int Average[AVERAGE]; ///< average for drift calculation + + struct AVResampleContext *AvResample; ///< second audio resample context +#define MAX_CHANNELS 8 ///< max number of channels supported + int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers + int BufferSize; ///< size of sample buffer + int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples + int RemainSize; ///< size of remain buffer + int RemainCount; ///< number of remaining samples }; #ifdef USE_PASSTHROUGH @@ -710,6 +728,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, audio_decoder->Channels = 0; audio_decoder->HwSampleRate = 0; audio_decoder->HwChannels = 0; + audio_decoder->StartPTS = AV_NOPTS_VALUE; } /** @@ -720,6 +739,21 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, void CodecAudioClose(AudioDecoder * audio_decoder) { // FIXME: output any buffered data + if (audio_decoder->AvResample) { + int ch; + + av_resample_close(audio_decoder->AvResample); + audio_decoder->AvResample = NULL; + audio_decoder->RemainCount = 0; + audio_decoder->BufferSize = 0; + audio_decoder->RemainSize = 0; + for (ch = 0; ch < MAX_CHANNELS; ++ch) { + free(audio_decoder->Buffer[ch]); + audio_decoder->Buffer[ch] = NULL; + free(audio_decoder->Remain[ch]); + audio_decoder->Remain[ch] = NULL; + } + } if (audio_decoder->ReSample) { audio_resample_close(audio_decoder->ReSample); audio_decoder->ReSample = NULL; @@ -810,6 +844,238 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels) } /** +** Set/update audio pts clock. +** +** @param audio_decoder audio decoder data +*/ +static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts) +{ + struct timespec nowtime; + int64_t tim_diff; + int64_t pts_diff; + int64_t drift; + + AudioSetClock(pts); + + // start drift detection + if (audio_decoder->StartPTS == (int64_t) AV_NOPTS_VALUE && AudioGetDelay()) { + audio_decoder->StartPTS = AudioGetClock(); + audio_decoder->LastPTS = audio_decoder->StartPTS; + clock_gettime(CLOCK_REALTIME, &audio_decoder->StartTime); + return; + } + + pts = AudioGetClock(); + clock_gettime(CLOCK_REALTIME, &nowtime); + pts_diff = pts - audio_decoder->StartPTS; + + tim_diff = (nowtime.tv_sec - audio_decoder->StartTime.tv_sec) + * 1000 * 1000 * 1000 + (nowtime.tv_nsec - + audio_decoder->StartTime.tv_nsec); + drift = pts_diff * 1000 * 1000 / 90 - tim_diff; + + if (abs(drift) > 100 * 1000 * 1000) { + // drift too big, pts changed? + audio_decoder->StartPTS = pts; + audio_decoder->LastPTS = audio_decoder->StartPTS; + audio_decoder->StartTime = nowtime; + return; + } + // collect over some time + if (pts - audio_decoder->LastPTS < 10 * 1000 * 90) { + return; + } + audio_decoder->LastPTS = pts; + + audio_decoder->Drift += + (int)((10 * audio_decoder->SampleRate * drift) / tim_diff); + + if (audio_decoder->AvResample) { + av_resample_compensate(audio_decoder->AvResample, audio_decoder->Drift, + 10 * audio_decoder->SampleRate); + } + + Info("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %g\n", + audio_decoder->Drift, drift / (1000 * 1000), drift, + (double)drift / tim_diff); + printf("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %d\n", + audio_decoder->Drift, drift / (1000 * 1000), drift, + (int)((10 * audio_decoder->SampleRate * drift) / tim_diff)); +} + +/** +** Handle audio format changes. +** +** @param audio_decoder audio decoder data +*/ +static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder) +{ + const AVCodecContext *audio_ctx; + int err; + int isAC3; + + audio_ctx = audio_decoder->AudioCtx; + + 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; + } + if (audio_decoder->AvResample) { + av_resample_close(audio_decoder->AvResample); + audio_decoder->AvResample = NULL; + audio_decoder->RemainCount = 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; + return; + } + } else { + Debug(3, "codec/audio: audio setup error\n"); + // FIXME: handle errors + audio_decoder->HwChannels = 0; + audio_decoder->HwSampleRate = 0; + return; + } + } + // prepare audio drift resample + if (!isAC3) { + audio_decoder->AvResample = + av_resample_init(audio_decoder->HwSampleRate, + audio_decoder->HwSampleRate, 16, 10, 0, 0.8); + if (!audio_decoder->AvResample) { + Error(_("codec/audio: AvResample setup error\n")); + } + } +} + +/** +** Codec enqueue audio samples. +** +** @param audio_decoder audio decoder data +** @param data samples data +** @param count number of samples +** +*/ +void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count) +{ +#ifdef USE_AUDIO_DRIFT_CORRECTION + if (audio_decoder->AvResample) { + int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 + + FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16))); + int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4]; + int consumed; + int i; + int n; + int ch; + int bytes_n; + + bytes_n = count / audio_decoder->HwChannels; + // resize sample buffer, if needed + if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) { + audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n; + for (ch = 0; ch < MAX_CHANNELS; ++ch) { + audio_decoder->Buffer[ch] = + realloc(audio_decoder->Buffer[ch], + audio_decoder->BufferSize); + } + } + // copy remaining bytes into sample buffer + for (ch = 0; ch < audio_decoder->HwChannels; ++ch) { + memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch], + audio_decoder->RemainCount); + } + // deinterleave samples into sample buffer + for (i = 0; i < bytes_n / 2; i++) { + for (ch = 0; ch < audio_decoder->HwChannels; ++ch) { + audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i] + = data[i * audio_decoder->HwChannels + ch]; + } + } + + bytes_n += audio_decoder->RemainSize; + n = 0; // keep gcc lucky + // resample the sample buffer into tmp buffer + for (ch = 0; ch < audio_decoder->HwChannels; ++ch) { + n = av_resample(audio_decoder->AvResample, buftmp[ch], + audio_decoder->Buffer[ch], &consumed, bytes_n / 2, + sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1); + //printf("remain%d: %d = %d/%d\n", ch, n, consumed, bytes_n /2); + // fixme remaining channels + if (bytes_n - consumed * 2 > audio_decoder->RemainSize) { + audio_decoder->RemainSize = bytes_n - consumed * 2; + } + audio_decoder->Remain[ch] = + realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize); + memcpy(audio_decoder->Remain[ch], + audio_decoder->Buffer[ch] + consumed, + audio_decoder->RemainSize); + audio_decoder->RemainCount = audio_decoder->RemainSize; + } + + // interleave samples from sample buffer + for (i = 0; i < n; i++) { + for (ch = 0; ch < audio_decoder->HwChannels; ++ch) { + buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i]; + } + } + n *= 2; + +#if 0 + // FIXME: must split channels, filter, join channels + n = av_resample(audio_decoder->AvResample, buf, data, &consumed, count, + sizeof(buf), 1); + if (n < 0) { + Error(_("codec/audio: can't av_resample\n")); + return; + } + if (consumed != count) { + Error(_("codec/audio: av_resample didn't consume all samples\n")); + } +#endif + n *= audio_decoder->HwChannels; + CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels); + AudioEnqueue(buf, n); + return; + } +#endif + CodecReorderAudioFrame(data, count, audio_decoder->HwChannels); + AudioEnqueue(data, count); +} + +/** ** Decode an audio packet. ** ** PTS must be handled self. @@ -847,64 +1113,17 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt); #else #endif - // Update audio clock + + // update audio clock if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) { - AudioSetClock(avpkt->pts); + CodecAudioSetClock(audio_decoder, 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; - } - - 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 - audio_decoder->HwChannels = 0; - audio_decoder->HwSampleRate = 0; - return; - } - } + CodecAudioUpdateFormat(audio_decoder); } if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) { @@ -931,9 +1150,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) 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); + CodecAudioEnqueue(audio_decoder, outbuf, outlen); } } else { #ifdef USE_PASSTHROUGH @@ -955,6 +1172,10 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) 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); + // don't play with the ac-3 samples + AudioEnqueue(buf, buf_sz); + // FIXME: handle drift. + return; } #if 0 // @@ -1008,8 +1229,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) // True HD? #endif #endif - CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels); - AudioEnqueue(buf, buf_sz); + CodecAudioEnqueue(audio_decoder, buf, buf_sz); } } } |