diff options
Diffstat (limited to 'codec.c')
| -rw-r--r-- | codec.c | 337 |
1 files changed, 327 insertions, 10 deletions
@@ -36,6 +36,8 @@ #define USE_AUDIO_DRIFT_CORRECTION /// compile AC3 audio drift correction support (experimental) #define USE_AC3_DRIFT_CORRECTION + /// use ffmpeg libswresample API +#define noUSE_SWRESAMPLE #include <stdio.h> #include <unistd.h> @@ -58,6 +60,9 @@ #ifdef USE_VDPAU #include <libavcodec/vdpau.h> #endif +#ifdef USE_SWRESAMPLE +#include <libswresample/swresample.h> +#endif #ifndef __USE_GNU #define __USE_GNU @@ -629,7 +634,16 @@ struct _audio_decoder_ int HwSampleRate; ///< hw sample rate int HwChannels; ///< hw channels +#ifndef USE_SWRESAMPLE ReSampleContext *ReSample; ///< audio resampling context +#endif +#ifdef USE_SWRESAMPLE +#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100) + struct SwrContext *Resample; ///< audio software resample context +#else + SwrContext *Resample; ///< audio software resample context +#endif +#endif int64_t LastDelay; ///< last delay struct timespec LastTime; ///< last time @@ -639,6 +653,7 @@ struct _audio_decoder_ int DriftCorr; ///< audio drift correction value int DriftFrac; ///< audio drift fraction for ac3 +#ifndef USE_SWRESAMPLE struct AVResampleContext *AvResample; ///< second audio resample context #define MAX_CHANNELS 8 ///< max number of channels supported int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers @@ -646,9 +661,12 @@ struct _audio_decoder_ int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples int RemainSize; ///< size of remain buffer int RemainCount; ///< number of remaining samples +#endif }; #ifdef USE_AUDIO_DRIFT_CORRECTION +#define CORRECT_PCM 1 ///< do PCM audio-drift correction +#define CORRECT_AC3 2 ///< do AC3§ audio-drift correction static char CodecAudioDrift; ///< flag: enable audio-drift correction #else static const int CodecAudioDrift = 0; @@ -718,10 +736,16 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, } if (CodecDownmix) { +#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,61,100) audio_decoder->AudioCtx->request_channels = 2; +#endif audio_decoder->AudioCtx->request_channel_layout = AV_CH_LAYOUT_STEREO_DOWNMIX; } +#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,61,100) + // this has no effect + // audio_decoder->AudioCtx->request_sample_fmt = AV_SAMPLE_FMT_S16; +#endif pthread_mutex_lock(&CodecLockMutex); // open codec #if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,5,0) @@ -768,6 +792,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, void CodecAudioClose(AudioDecoder * audio_decoder) { // FIXME: output any buffered data +#ifndef USE_SWRESAMPLE if (audio_decoder->AvResample) { int ch; @@ -787,6 +812,12 @@ void CodecAudioClose(AudioDecoder * audio_decoder) audio_resample_close(audio_decoder->ReSample); audio_decoder->ReSample = NULL; } +#endif +#ifdef USE_SWRESAMPLE + if (audio_decoder->Resample) { + swr_free(&audio_decoder->Resample); + } +#endif if (audio_decoder->AudioCtx) { pthread_mutex_lock(&CodecLockMutex); avcodec_close(audio_decoder->AudioCtx); @@ -895,6 +926,8 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels) } } +#ifndef USE_SWRESAMPLE + /** ** Set/update audio pts clock. ** @@ -916,14 +949,15 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts) if (!delay) { return; } - clock_gettime(CLOCK_REALTIME, &nowtime); + clock_gettime(CLOCK_MONOTONIC, &nowtime); if (!audio_decoder->LastDelay) { audio_decoder->LastTime = nowtime; audio_decoder->LastPTS = pts; audio_decoder->LastDelay = delay; audio_decoder->Drift = 0; audio_decoder->DriftFrac = 0; - Debug(3, "codec/audio: inital delay %" PRId64 "ms\n", delay / 90); + Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n", + delay / 90); return; } // collect over some time @@ -1205,14 +1239,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) } Error(_("codec: error more than one frame data\n")); } -#ifdef notyetFF_API_OLD_DECODE_AUDIO - // FIXME: ffmpeg git comeing - int got_frame; - - avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt); -#else -#endif - // update audio clock if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) { CodecAudioSetClock(audio_decoder, avpkt->pts); @@ -1352,6 +1378,297 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) } } +#endif + +#ifdef USE_SWRESAMPLE + +/** +** Set/update audio pts clock. +** +** @param audio_decoder audio decoder data +** @param pts presentation timestamp +*/ +static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts) +{ + struct timespec nowtime; + int64_t delay; + int64_t tim_diff; + int64_t pts_diff; + int drift; + int corr; + + AudioSetClock(pts); + + delay = AudioGetDelay(); + if (!delay) { + return; + } + clock_gettime(CLOCK_MONOTONIC, &nowtime); + if (!audio_decoder->LastDelay) { + audio_decoder->LastTime = nowtime; + audio_decoder->LastPTS = pts; + audio_decoder->LastDelay = delay; + audio_decoder->Drift = 0; + audio_decoder->DriftFrac = 0; + Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n", + delay / 90); + return; + } + // collect over some time + pts_diff = pts - audio_decoder->LastPTS; + if (pts_diff < 10 * 1000 * 90) { + return; + } + + tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec) + * 1000 * 1000 * 1000 + (nowtime.tv_nsec - + audio_decoder->LastTime.tv_nsec); + + drift = + (tim_diff * 90) / (1000 * 1000) - pts_diff + delay - + audio_decoder->LastDelay; + + // adjust rounding error + nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90); + audio_decoder->LastTime = nowtime; + audio_decoder->LastPTS = pts; + audio_decoder->LastDelay = delay; + + if (0) { + Debug(3, + "codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4" + PRId64 "ms %f %d\n", pts_diff / 90, tim_diff / (1000 * 1000), + delay / 90, drift / 90.0, audio_decoder->DriftCorr); + } + // underruns and av_resample have the same time :((( + if (abs(drift) > 10 * 90) { + // drift too big, pts changed? + Debug(3, "codec/audio: drift(%6d) %3dms reset\n", + audio_decoder->DriftCorr, drift / 90); + audio_decoder->LastDelay = 0; +#ifdef DEBUG + corr = 0; // keep gcc happy +#endif + } else { + + drift += audio_decoder->Drift; + audio_decoder->Drift = drift; + corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000); + // SPDIF/HDMI passthrough + if ((CodecAudioDrift & 2) && (!CodecPassthroughAC3 + || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)) { + audio_decoder->DriftCorr = -corr; + } + + if (audio_decoder->DriftCorr < -20000) { // limit correction + audio_decoder->DriftCorr = -20000; + } else if (audio_decoder->DriftCorr > 20000) { + audio_decoder->DriftCorr = 20000; + } + } + + if (audio_decoder->Resample && audio_decoder->DriftCorr) { + int distance; + + // try workaround for buggy ffmpeg 0.10 + if (abs(audio_decoder->DriftCorr) < 2000) { + distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000); + } else { + distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000); + } + if (swr_set_compensation(audio_decoder->Resample, + audio_decoder->DriftCorr / 10, distance)) { + Debug(3, "codec/audio: swr_set_compensation failed\n"); + } + } + if (1) { + static int c; + + if (!(c++ % 10)) { + Debug(3, "codec/audio: drift(%6d) %8dus %5d\n", + audio_decoder->DriftCorr, drift * 1000 / 90, corr); + } + } +} + +/** +** 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; + Debug(3, "codec/audio: format change %s %dHz *%d channels %s\n", + av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate, + audio_ctx->channels, CodecPassthroughAC3 ? "pass-through" : ""); + + audio_decoder->SampleRate = audio_ctx->sample_rate; + audio_decoder->HwSampleRate = audio_ctx->sample_rate; + audio_decoder->Channels = audio_ctx->channels; + audio_decoder->PassthroughAC3 = CodecPassthroughAC3; + + // 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: audio setup error\n"); + // FIXME: handle errors + audio_decoder->HwChannels = 0; + audio_decoder->HwSampleRate = 0; + return; + } + + if (isAC3) { // no AC3 conversion allowed + return; + } + + Debug(3, "codec/audio: resample %s %dHz *%d -> %s %dHz *%d\n", + av_get_sample_fmt_name(audio_ctx->sample_fmt), audio_ctx->sample_rate, + audio_ctx->channels, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), + audio_decoder->HwSampleRate, audio_decoder->HwChannels); + + audio_decoder->Resample = + swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout, + AV_SAMPLE_FMT_S16, audio_decoder->HwSampleRate, + audio_ctx->channel_layout, audio_ctx->sample_fmt, + audio_ctx->sample_rate, 0, NULL); + if (audio_decoder->Resample) { + swr_init(audio_decoder->Resample); + } else { + Error(_("codec/audio: can't setup resample\n")); + } +} + +/** +** Decode an audio packet. +** +** PTS must be handled self. +** +** @note the caller has not aligned avpkt and not cleared the end. +** +** @param audio_decoder audio decoder data +** @param avpkt audio packet +*/ +void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) +{ + AVCodecContext *audio_ctx; + AVFrame frame; + int got_frame; + int n; + + audio_ctx = audio_decoder->AudioCtx; + + frame.data[0] = NULL; + n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame, + (AVPacket *) avpkt); + if (n != avpkt->size) { + if (n == AVERROR(EAGAIN)) { + Error(_("codec/audio: latm\n")); + return; + } + if (n < 0) { // no audio frame could be decompressed + Error(_("codec/audio: bad audio frame\n")); + return; + } + Error(_("codec/audio: error more than one frame data\n")); + } + if (!got_frame) { + Error(_("codec/audio: no frame\n")); + return; + } + // update audio clock + if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) { + CodecAudioSetClock(audio_decoder, avpkt->pts); + } + // format change + if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3 + || audio_decoder->SampleRate != audio_ctx->sample_rate + || audio_decoder->Channels != audio_ctx->channels) { + CodecAudioUpdateFormat(audio_decoder); + + } + + if (!audio_decoder->HwSampleRate || !audio_decoder->HwChannels) { + return; // unsupported sample format + } +#ifdef USE_PASSTHROUGH + // SPDIF/HDMI passthrough + if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) { + int16_t spdif[6144 / 2]; + int spdif_sz; + + // build SPDIF header and append A52 audio to it + // avpkt is the original data + spdif_sz = 6144; + if (spdif_sz < avpkt->size + 8) { + Error(_("codec/audio: decoded data smaller than encoded\n")); + return; + } + // copy original data for output + spdif[0] = htole16(0xF872); // iec 61937 sync word + spdif[1] = htole16(0x4E1F); + spdif[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8); + spdif[3] = htole16(avpkt->size * 8); + // FIXME: not 100% sure, if endian is correct on not intel hardware + swab(avpkt->data, spdif + 4, avpkt->size); + memset(spdif + 4 + avpkt->size / 2, 0, spdif_sz - 8 - avpkt->size); + // don't play with the ac-3 samples + AudioEnqueue(spdif, spdif_sz); + return; + } +#endif + if (0) { + char strbuf[32]; + int data_sz; + int plane_sz; + + data_sz = + av_samples_get_buffer_size(&plane_sz, audio_ctx->channels, + frame.nb_samples, audio_ctx->sample_fmt, 1); + fprintf(stderr, "codec/audio: sample_fmt %s\n", + av_get_sample_fmt_name(audio_ctx->sample_fmt)); + av_get_channel_layout_string(strbuf, 32, audio_ctx->channels, + audio_ctx->channel_layout); + fprintf(stderr, "codec/audio: layout %s\n", strbuf); + fprintf(stderr, + "codec/audio: channels %d samples %d plane %d data %d\n", + audio_ctx->channels, frame.nb_samples, plane_sz, data_sz); + } + + if (audio_decoder->Resample) { + uint8_t outbuf[8192 * 2 * 8]; + uint8_t *out[1]; + + out[0] = outbuf; + n = swr_convert(audio_decoder->Resample, out, + sizeof(outbuf) / (2 * audio_decoder->HwChannels), + (const uint8_t **)frame.extended_data, frame.nb_samples); + if (n > 0) { + CodecReorderAudioFrame((int16_t *) outbuf, + n * 2 * audio_decoder->HwChannels, audio_decoder->HwChannels); + AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels); + } + return; + } +} + +#endif + /** ** Flush the audio decoder. ** |