Index: libs/libmythtv/avformatdecoder.cpp =================================================================== --- libs/libmythtv/avformatdecoder.cpp (revision 8241) +++ libs/libmythtv/avformatdecoder.cpp (working copy) @@ -35,6 +35,9 @@ #define MAX_AC3_FRAME_SIZE 6144 +static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate); +static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq); + /** Set to zero to allow any number of AC3 channels. */ #define MAX_OUTPUT_CHANNELS 2 @@ -180,6 +183,7 @@ // Audio audioSamples(new short int[AVCODEC_MAX_AUDIO_FRAME_SIZE]), allow_ac3_passthru(false), + allow_dts_passthru(false), // Audio selection wantedAudioStream(), selectedAudioStream(), // Subtitle selection @@ -196,6 +200,7 @@ save_cctc[0] = save_cctc[1] = 0; allow_ac3_passthru = gContext->GetNumSetting("AC3PassThru", false); + allow_dts_passthru = gContext->GetNumSetting("DTSPassThru", false); audioIn.sample_size = -32; // force SetupAudioStream to run once } @@ -946,6 +951,12 @@ <<") already open, leaving it alone."); } assert(enc->codec_id); + if (enc->codec_id == CODEC_ID_DTS) + { + enc->sample_rate = 48000; + enc->channels = 2; + // enc->bit_rate = what??; + } bitrate += enc->bit_rate; break; } @@ -1838,30 +1849,10 @@ return ret; } -int filter_ac3(const AVFormatContext *ic, - const sinfo_vec_t &audioStreams, - const vector &fs) -{ - int selectedTrack = -1; - - vector::const_iterator it = fs.begin(); - for (; it != fs.end(); ++it) - { - const int stream_index = audioStreams[*it].av_stream_index; - const AVCodecContext *ctx = ic->streams[stream_index]->codec; - if (CODEC_ID_AC3 == ctx->codec_id) - { - selectedTrack = *it; - break; - } - } - - return selectedTrack; -} - int filter_max_ch(const AVFormatContext *ic, const sinfo_vec_t &audioStreams, - const vector &fs) + const vector &fs, + enum CodecID codecId = CODEC_ID_NONE) { int selectedTrack = -1, max_seen = -1; @@ -1870,10 +1861,13 @@ { const int stream_index = audioStreams[*it].av_stream_index; const AVCodecContext *ctx = ic->streams[stream_index]->codec; - if (max_seen < ctx->channels) + if (codecId == CODEC_ID_NONE || codecId == ctx->codec_id) { - selectedTrack = *it; - max_seen = ctx->channels; + if (max_seen < ctx->channels) + { + selectedTrack = *it; + max_seen = ctx->channels; + } } } @@ -1883,25 +1877,43 @@ /** \fn AvFormatDecoder::autoSelectAudioTrack(void) * \brief Selects the best audio track. * - * This function will select the best audio track - * available using the following rules: + * This function will select the best audio track available + * using the following criteria, in order of decreasing + * preference: * - * First we try to select the stream last selected by the - * user, which is recalled as the Nth stream in the - * preferred language. If it can not be located we attempt - * to find a stream in the same language. + * 1) The stream last selected by the user, which is + * recalled as the Nth stream in the preferred language. + * If it can not be located we attempt to find a stream + * in the same language. * - * If we can not reselect the last user selected stream, - * then for each preferred language from most preferred to - * least preferred, we try to use the first AC3 track found, - * or if no AC3 audio is found then we try to select the - * audio track with the greatest number of audio channels. + * 2) If we can not reselect the last user selected stream, + * then for each preferred language from most preferred + * to least preferred, we try to find a new stream based + * on the algorithm below. * - * If we can not select a stream in a preferred language - * we try to use the first AC3 track found irrespective - * of language, and if no AC3 audio is found then we try - * to select the audio track with the greatest number of - * audio channels. + * 3) If we can not select a stream in a preferred language + * we try to select a stream irrespective of language + * based on the algorithm below. + * + * When searching for a new stream (ie. options 2 and 3 + * above), the following search is carried out in order: + * + * i) If DTS passthrough is enabled then the DTS track with + * the greatest number of audio channels is selected + * (the first will be chosen if there are several the + * same). If DTS passthrough is not enabled this step + * will be skipped because internal DTS decoding is not + * currently supported. + * + * ii) If no DTS track is chosen, the AC3 track with the + * greatest number of audio channels is selected (the + * first will be chosen if there are several the same). + * Internal decoding of AC3 is supported, so this will + * be used irrespective of whether AC3 passthrough is + * enabled. + * + * iii) Lastly the track with the greatest number of audio + * channels irrespective of type will be selected. * \return true if a track was selected, false otherwise */ bool AvFormatDecoder::autoSelectAudioTrack(void) @@ -1918,8 +1930,9 @@ AVCodecContext *codec_ctx = ic->streams[idx]->codec; AudioInfo item(codec_ctx->codec_id, codec_ctx->sample_rate, codec_ctx->channels, - (allow_ac3_passthru && !transcoding && - (codec_ctx->codec_id == CODEC_ID_AC3))); + !transcoding && + (allow_ac3_passthru && codec_ctx->codec_id == CODEC_ID_AC3 || + allow_dts_passthru && codec_ctx->codec_id == CODEC_ID_DTS)); VERBOSE(VB_AUDIO, LOC + " * " + item.toString()); } #endif @@ -1952,8 +1965,11 @@ for (; it != languagePreference.end(); ++it) { vector flang = filter_lang(audioStreams, *it); - if ((selectedTrack = filter_ac3(ic, audioStreams, flang)) >= 0) + if (allow_dts_passthru && + (selectedTrack = filter_max_ch(ic, audioStreams, flang, CODEC_ID_DTS)) >= 0) break; + if ((selectedTrack = filter_max_ch(ic, audioStreams, flang, CODEC_ID_AC3)) >= 0) + break; if ((selectedTrack = filter_max_ch(ic, audioStreams, flang)) >= 0) break; } @@ -1962,7 +1978,9 @@ { VERBOSE(VB_AUDIO, LOC + "Trying to select audio track (wo/lang)"); vector flang = filter_lang(audioStreams, -1); - if ((selectedTrack = filter_ac3(ic, audioStreams, flang)) < 0) + if ((!allow_dts_passthru || + (selectedTrack = filter_max_ch(ic, audioStreams, flang, CODEC_ID_DTS)) < 0) && + (selectedTrack = filter_max_ch(ic, audioStreams, flang, CODEC_ID_AC3)) < 0) selectedTrack = filter_max_ch(ic, audioStreams, flang); } } @@ -2365,7 +2383,7 @@ pthread_mutex_lock(&avcodeclock); ret = len; data_size = 0; - if (audioOut.do_ac3_passthru) + if (audioOut.do_passthru) { data_size = pkt->size; ret = EncodeAC3Frame(ptr, len, audioSamples, @@ -2397,11 +2415,13 @@ } pthread_mutex_unlock(&avcodeclock); - ptr += ret; - len -= ret; if (data_size <= 0) + { + ptr += ret; + len -= ret; continue; + } long long temppts = lastapts; @@ -2599,12 +2619,13 @@ { assert(curstream); assert(curstream->codec); - codec_ctx = curstream->codec; - bool do_ac3_passthru = (allow_ac3_passthru && !transcoding && - (codec_ctx->codec_id == CODEC_ID_AC3)); + codec_ctx = curstream->codec; + bool do_passthru = !transcoding && + (allow_ac3_passthru && codec_ctx->codec_id == CODEC_ID_AC3 || + allow_dts_passthru && codec_ctx->codec_id == CODEC_ID_DTS); info = AudioInfo(codec_ctx->codec_id, codec_ctx->sample_rate, codec_ctx->channels, - do_ac3_passthru); + do_passthru); } if (info == audioIn) @@ -2614,7 +2635,7 @@ QString("audio track #%1").arg(currentAudioTrack+1)); audioOut = audioIn = info; - if (audioIn.do_ac3_passthru) + if (audioIn.do_passthru) { // A passthru stream looks like a 48KHz 2ch (@ 16bit) to the sound card audioOut.channels = 2; @@ -2651,6 +2672,8 @@ { int enc_len; int flags, sample_rate, bit_rate; + int nr_samples = 0, block_len; + bool dts = false; unsigned char* ucsamples = (unsigned char*) samples; // we don't do any length/crc validation of the AC3 frame here; presumably @@ -2661,7 +2684,20 @@ // ignore, and if so, may as well just assume that it will ignore // anything with a bad CRC... - enc_len = a52_syncinfo(data, &flags, &sample_rate, &bit_rate); + enc_len = dts_syncinfo(data, &flags, &sample_rate, &bit_rate); + if(enc_len >= 10) + { + dts = true; + int rate, sfreq, nblks; + dts_decode_header(data, &rate, &nblks, &sfreq); + nr_samples = nblks * 32; + block_len = nr_samples * 2 * 2; + } + else + { + enc_len = a52_syncinfo(data, &flags, &sample_rate, &bit_rate); + block_len = MAX_AC3_FRAME_SIZE; + } if (enc_len == 0 || enc_len > len) { @@ -2669,25 +2705,54 @@ return len; } - if (enc_len > MAX_AC3_FRAME_SIZE - 8) - enc_len = MAX_AC3_FRAME_SIZE - 8; + if (enc_len > block_len - 8) + { + enc_len = block_len - 8; + } swab(data, ucsamples + 8, enc_len); - // the following values come from ao_hwac3.c in mplayer. + // the following values come from libmpcodecs/ad_hwac3.c in mplayer. // they form a valid IEC958 AC3 header. ucsamples[0] = 0x72; ucsamples[1] = 0xF8; ucsamples[2] = 0x1F; ucsamples[3] = 0x4E; - ucsamples[4] = 0x01; + if(dts) + { + switch(nr_samples) + { + case 512: + ucsamples[4] = 0x0B; /* DTS-1 (512-sample bursts) */ + break; + + case 1024: + ucsamples[4] = 0x0C; /* DTS-2 (1024-sample bursts) */ + break; + + case 2048: + ucsamples[4] = 0x0D; /* DTS-3 (2048-sample bursts) */ + break; + + default: + VERBOSE(VB_IMPORTANT, LOC + + QString("DTS: %1-sample bursts not supported") + .arg(nr_samples)); + ucsamples[4] = 0x00; + break; + } + } + else + { + ucsamples[4] = 0x01; + } ucsamples[5] = 0x00; ucsamples[6] = (enc_len << 3) & 0xFF; ucsamples[7] = (enc_len >> 5) & 0xFF; - memset(ucsamples + 8 + enc_len, 0, MAX_AC3_FRAME_SIZE - 8 - enc_len); - samples_size = MAX_AC3_FRAME_SIZE; + memset(ucsamples + 8 + enc_len, 0, block_len - 8 - enc_len); + samples_size = block_len; - return len; // consume whole frame even if len > enc_len ? + return enc_len; } void AvFormatDecoder::AddTextData(unsigned char *buf, int len, @@ -2695,3 +2760,134 @@ { m_parent->AddTextData((char*)buf, len, timecode, type); } + +static int DTS_SAMPLEFREQS[16] = +{ + 0, + 8000, + 16000, + 32000, + 64000, + 128000, + 11025, + 22050, + 44100, + 88200, + 176400, + 12000, + 24000, + 48000, + 96000, + 192000 +}; + +static int DTS_BITRATES[30] = +{ + 32000, + 56000, + 64000, + 96000, + 112000, + 128000, + 192000, + 224000, + 256000, + 320000, + 384000, + 448000, + 512000, + 576000, + 640000, + 768000, + 896000, + 1024000, + 1152000, + 1280000, + 1344000, + 1408000, + 1411200, + 1472000, + 1536000, + 1920000, + 2048000, + 3072000, + 3840000, + 4096000 +}; + +static int dts_syncinfo(uint8_t *indata_ptr, int */*flags*/, int *sample_rate, int *bit_rate) +{ + int nblks; + int rate; + int sfreq; + + int fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq); + if(fsize >= 0) + { + if(rate >= 0 && rate <= 29) + *bit_rate = DTS_BITRATES[rate]; + else + *bit_rate = 0; + if(sfreq >= 1 && sfreq <= 15) + *sample_rate = DTS_SAMPLEFREQS[sfreq]; + else + *sample_rate = 0; + } + return fsize; +} + +static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq) +{ + if(((indata_ptr[0] << 24) | (indata_ptr[1] << 16) | (indata_ptr[2] << 8) + | (indata_ptr[3])) != 0x7ffe8001) + return -1; + + int ftype = indata_ptr[4] >> 7; + + int surp = (indata_ptr[4] >> 2) & 0x1f; + surp = (surp + 1) % 32; + + *nblks = (indata_ptr[4] & 0x01) << 6 | (indata_ptr[5] >> 2); + *nblks = *nblks + 1; + + int fsize = (indata_ptr[5] & 0x03) << 12 | (indata_ptr[6] << 4) | (indata_ptr[7] >> 4); + fsize = fsize + 1; + + *sfreq = (indata_ptr[8] >> 2) & 0x0f; + *rate = (indata_ptr[8] & 0x03) << 3 | ((indata_ptr[9] >> 5) & 0x07); + + if(ftype != 1) + { + VERBOSE(VB_IMPORTANT, LOC + + QString("DTS: Termination frames not handled (ftype %1)").arg(ftype)); + return -1; + } + + if(*sfreq != 13) + { + VERBOSE(VB_IMPORTANT, LOC + + QString("DTS: Only 48kHz supported (sfreq %1)").arg(*sfreq)); + return -1; + } + + if((fsize > 8192) || (fsize < 96)) + { + VERBOSE(VB_IMPORTANT, LOC + + QString("DTS: fsize: %1 invalid").arg(fsize)); + return -1; + } + + if(*nblks != 8 && + *nblks != 16 && + *nblks != 32 && + *nblks != 64 && + *nblks != 128 && + ftype == 1) + { + VERBOSE(VB_IMPORTANT, LOC + + QString("DTS: nblks %1 not valid for normal frame").arg(*nblks)); + return -1; + } + + return fsize; +} Index: libs/libmythtv/avformatdecoder.h =================================================================== --- libs/libmythtv/avformatdecoder.h (revision 8241) +++ libs/libmythtv/avformatdecoder.h (working copy) @@ -38,32 +38,32 @@ public: AudioInfo() : codec_id(CODEC_ID_NONE), sample_size(-2), sample_rate(-1), - channels(-1), do_ac3_passthru(false) + channels(-1), do_passthru(false) {;} - AudioInfo(CodecID id, int sr, int ch, bool ac3) : + AudioInfo(CodecID id, int sr, int ch, bool passthru) : codec_id(id), sample_size(ch*2), sample_rate(sr), - channels(ch), do_ac3_passthru(ac3) + channels(ch), do_passthru(passthru) {;} CodecID codec_id; int sample_size, sample_rate, channels; - bool do_ac3_passthru; + bool do_passthru; /// \brief Bits per sample. - int bps(void) const { return (8 * sample_size) / channels; } + int bps(void) const { return channels ? (8 * sample_size) / channels : 0; } bool operator==(const AudioInfo &o) const { return (codec_id==o.codec_id && channels==o.channels && sample_size==o.sample_size && sample_rate==o.sample_rate && - do_ac3_passthru==o.do_ac3_passthru); + do_passthru==o.do_passthru); } QString toString() const { return QString("id(%1) %2Hz %3ch %4bps%5") .arg(codec_id_string(codec_id),4).arg(sample_rate,5) .arg(channels,2).arg(bps(),3) - .arg((do_ac3_passthru) ? "pt":"",3); + .arg((do_passthru) ? "pt":"",3); } }; @@ -139,8 +139,8 @@ protected: /// Attempt to find the optimal audio stream to use based on the number of channels, - /// and if we're doing AC3 passthrough. This will select the highest stream number - /// that matches our criteria. + /// and if we're doing AC3/DTS passthrough. This will select the highest stream + /// number that matches our criteria. bool autoSelectAudioTrack(); bool autoSelectSubtitleTrack(); @@ -237,6 +237,7 @@ // Audio short int *audioSamples; bool allow_ac3_passthru; + bool allow_dts_passthru; AudioInfo audioIn; AudioInfo audioOut; Index: libs/libavformat/mpeg.c =================================================================== --- libs/libavformat/mpeg.c (revision 8241) +++ libs/libavformat/mpeg.c (working copy) @@ -88,7 +88,7 @@ #define AUDIO_ID 0xc0 #define VIDEO_ID 0xe0 #define AC3_ID 0x80 -#define DTS_ID 0x8a +#define DTS_ID 0x88 #define LPCM_ID 0xa0 #define SUB_ID 0x20 Index: programs/mythfrontend/globalsettings.cpp =================================================================== --- programs/mythfrontend/globalsettings.cpp (revision 8241) +++ programs/mythfrontend/globalsettings.cpp (working copy) @@ -133,6 +133,20 @@ return gc; } +#ifdef CONFIG_DTS +static HostCheckBox *DTSPassThrough() +{ + HostCheckBox *gc = new HostCheckBox("DTSPassThru"); + gc->setLabel(QObject::tr("Enable DTS to SPDIF passthrough")); + gc->setValue(false); + gc->setHelpText(QObject::tr("Enable sending DTS audio directly to your " + "sound card's SPDIF output, on sources which contain " + "DTS soundtracks (usually DVDs). Requires that the " + "audio output device be set to something suitable.")); + return gc; +} +#endif + static HostCheckBox *Deinterlace() { HostCheckBox *gc = new HostCheckBox("Deinterlace"); @@ -2069,6 +2083,9 @@ addChild(AudioOutputDevice()); addChild(AC3PassThrough()); +#ifdef CONFIG_DTS + addChild(DTSPassThrough()); +#endif addChild(AggressiveBuffer()); Setting* volumeControl = MythControlsVolume();