Index: libs/libmythtv/avformatdecoder.cpp =================================================================== --- libs/libmythtv/avformatdecoder.cpp (revision 8035) +++ libs/libmythtv/avformatdecoder.cpp (working copy) @@ -29,6 +29,9 @@ #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); + extern pthread_mutex_t avcodeclock; int get_avf_buffer_xvmc(struct AVCodecContext *c, AVFrame *pic); @@ -919,6 +922,12 @@ <<") 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??; + } if (enc->channels > 2) enc->channels = 2; bitrate += enc->bit_rate; @@ -1760,11 +1769,12 @@ // // This function will select the best audio track -// available usgin the following rules: +// available using the following rules: // -// 1. The fist AC3 track found will be select, -// 2. If no AC3 is found then the audio track with -// the most number of channels is selected. +// 1. The first DTS track found will be selected, +// 2. If no DTS is found then the fist AC3 track will be selected, +// 3. If no AC3 is found then the audio track with the most number +// of channels is selected. // // This code has no awareness to language preferences // although I don't think it would be too hard to @@ -1778,32 +1788,37 @@ return false; } - int minChannels = 1; int maxTracks = (audioStreams.size() - 1); int track; - if (do_ac3_passthru) - minChannels = 2; + int firstAC3Track = -1; + int firstDtsTrack = -1; int selectedTrack = -1; - int selectedChannels = -1; - - while ((selectedTrack == -1) && (minChannels >= 0)) + int selectedChannels = 0; + + for (track = 0; track <= maxTracks; track++) { - for (track = 0; track <= maxTracks; track++) + int tempStream = audioStreams[track]; + AVCodecContext *e = ic->streams[tempStream]->codec; + + switch (e->codec_id) { - int tempStream = audioStreams[track]; - AVCodecContext *e = ic->streams[tempStream]->codec; + case CODEC_ID_AC3: + if(firstAC3Track == -1) + { + firstAC3Track = track; + } + break; - if (e->channels > minChannels) - { - //if we find an AC3 codec then we select it - //as the preferred codec. - if (e->codec_id == CODEC_ID_AC3) + case CODEC_ID_DTS: + if(firstDtsTrack == -1) { - selectedTrack = track; - break; + firstDtsTrack = track; } + break; + default: + if (e->channels > selectedChannels) { // this is a candidate with more channels @@ -1812,12 +1827,19 @@ selectedChannels = e->channels; selectedTrack = track; } - } } - if (selectedTrack == -1) + } + + if (!transcoding) + { + if (firstDtsTrack >= 0 && gContext->GetNumSetting("DTSPassThru", false)) { - minChannels--; + selectedTrack = firstDtsTrack; } + else if(firstAC3Track >= 0 && gContext->GetNumSetting("AC3PassThru", false)) + { + selectedTrack = firstAC3Track; + } } if (selectedTrack == -1) @@ -1830,20 +1852,27 @@ wantedAudioStream = audioStreams[currentAudioTrack]; AVCodecContext *e = ic->streams[wantedAudioStream]->codec; - if (e->codec_id == CODEC_ID_AC3) + switch (e->codec_id) { - VERBOSE(VB_AUDIO, LOC + - QString("Auto-selecting AC3 audio track (stream #%2).") - .arg(wantedAudioStream)); + case CODEC_ID_AC3: + VERBOSE(VB_AUDIO, LOC + + QString("Auto-selecting AC3 audio track (stream #%2).") + .arg(wantedAudioStream)); + break; + + case CODEC_ID_DTS: + VERBOSE(VB_AUDIO, LOC + + QString("Auto-selecting DTS audio track (stream #%2).") + .arg(wantedAudioStream)); + break; + + default: + VERBOSE(VB_AUDIO, LOC + + QString("Auto-selecting audio track #%1 (stream #%2).") + .arg(selectedTrack + 1).arg(wantedAudioStream) + "\n\t\t\t" + + QString("It has %1 channels and we needed at least 1") + .arg(selectedChannels)); } - else - { - VERBOSE(VB_AUDIO, LOC + - QString("Auto-selecting audio track #%1 (stream #%2).") - .arg(selectedTrack + 1).arg(wantedAudioStream) + "\n\t\t\t" + - QString("It has %1 channels and we needed at least %2") - .arg(selectedChannels).arg(do_ac3_passthru ? 2 : 1)); - } SetupAudioStream(); CheckAudioParams(e->sample_rate, e->channels, true); return true; @@ -1863,9 +1892,11 @@ GetNVP()->SetEffDsp(curstream->codec->sample_rate * 100); - do_ac3_passthru = curstream->codec->codec_id == CODEC_ID_AC3 && - !transcoding && - gContext->GetNumSetting("AC3PassThru", false); + do_ac3_passthru = !transcoding && + (curstream->codec->codec_id == CODEC_ID_AC3 && + gContext->GetNumSetting("AC3PassThru", false) || + curstream->codec->codec_id == CODEC_ID_DTS && + gContext->GetNumSetting("DTSPassThru", false)); } @@ -2198,11 +2229,13 @@ } pthread_mutex_unlock(&avcodeclock); - ptr += ret; - len -= ret; if (data_size <= 0) + { + ptr += ret; + len -= ret; continue; + } if (!do_ac3_passthru) CheckAudioParams(curstream->codec->sample_rate, @@ -2385,6 +2418,8 @@ { 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 @@ -2395,7 +2430,20 @@ // 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) { @@ -2403,25 +2451,54 @@ 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, @@ -2429,3 +2506,134 @@ { 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/libavformat/mpeg.c =================================================================== --- libs/libavformat/mpeg.c (revision 8035) +++ 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 8035) +++ 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"); @@ -2043,6 +2057,9 @@ addChild(AudioOutputDevice()); addChild(AC3PassThrough()); +#ifdef CONFIG_DTS + addChild(DTSPassThrough()); +#endif addChild(AggressiveBuffer()); Setting* volumeControl = MythControlsVolume();