diff --git a/mythplugins/mythmusic/mythmusic/aacdecoder.cpp b/mythplugins/mythmusic/mythmusic/aacdecoder.cpp index 0e5b1ed..968c211 100644 --- a/mythplugins/mythmusic/mythmusic/aacdecoder.cpp +++ b/mythplugins/mythmusic/mythmusic/aacdecoder.cpp @@ -29,6 +29,7 @@ // MythTV headers #include #include +#include // MythMusic headers #include "aacdecoder.h" @@ -401,7 +402,7 @@ bool aacDecoder::initializeMP4() if (output()) { const AudioSettings settings( - 16 /*bits*/, channels, sample_rate, + 16 /*bits*/, channels, CODEC_ID_AAC, sample_rate, false /* AC3/DTS pass through */); output()->Reconfigure(settings); output()->SetSourceBitrate(bitrate); diff --git a/mythplugins/mythmusic/mythmusic/avfdecoder.cpp b/mythplugins/mythmusic/mythmusic/avfdecoder.cpp index 260612f..c7e2ace 100644 --- a/mythplugins/mythmusic/mythmusic/avfdecoder.cpp +++ b/mythplugins/mythmusic/mythmusic/avfdecoder.cpp @@ -253,8 +253,8 @@ bool avfDecoder::initialize() if (output()) { const AudioSettings settings( - 16 /*bits*/, m_audioDec->channels, m_audioDec->sample_rate, - false /* AC3/DTS pass through */); + 16 /*bits*/, m_audioDec->channels, m_audioDec->codec_id, + m_audioDec->sample_rate, false /* AC3/DTS pass through */); output()->Reconfigure(settings); output()->SetSourceBitrate(m_audioDec->bit_rate); } diff --git a/mythplugins/mythmusic/mythmusic/cddecoder.cpp b/mythplugins/mythmusic/mythmusic/cddecoder.cpp index 9eb0d74..9caaf24 100644 --- a/mythplugins/mythmusic/mythmusic/cddecoder.cpp +++ b/mythplugins/mythmusic/mythmusic/cddecoder.cpp @@ -14,6 +14,7 @@ using namespace std; #include #include +#include CdDecoder::CdDecoder(const QString &file, DecoderFactory *d, QIODevice *i, AudioOutput *o) : @@ -149,7 +150,8 @@ bool CdDecoder::initialize() if (output()) { const AudioSettings settings( - 16 /*bits*/, chan, freq, false /* AC3/DTS passthru */); + 16 /*bits*/, chan, CODEC_ID_PCM_S16LE, freq, + false /* AC3/DTS passthru */); output()->Reconfigure(settings); output()->SetSourceBitrate(44100 * 2 * 16); } diff --git a/mythplugins/mythmusic/mythmusic/main.cpp b/mythplugins/mythmusic/mythmusic/main.cpp index cde9575..c520219 100644 --- a/mythplugins/mythmusic/mythmusic/main.cpp +++ b/mythplugins/mythmusic/mythmusic/main.cpp @@ -367,6 +367,7 @@ void setupKeys(void) REG_KEY("Music", "VOLUMEDOWN", "Volume down", "[,{,F10,Volume Down"); REG_KEY("Music", "VOLUMEUP", "Volume up", "],},F11,Volume Up"); REG_KEY("Music", "MUTE", "Mute", "|,\\,F9,Volume Mute"); + REG_KEY("Music", "TOGGLEUPMIX","Toggle upmixer", "Ctrl+U"); REG_KEY("Music", "CYCLEVIS", "Cycle visualizer mode", "6"); REG_KEY("Music", "BLANKSCR", "Blank screen", "5"); REG_KEY("Music", "THMBUP", "Increase rating", "9"); diff --git a/mythplugins/mythmusic/mythmusic/musicplayer.cpp b/mythplugins/mythmusic/mythmusic/musicplayer.cpp index fad20ee..088f88b 100644 --- a/mythplugins/mythmusic/mythmusic/musicplayer.cpp +++ b/mythplugins/mythmusic/mythmusic/musicplayer.cpp @@ -348,16 +348,18 @@ void MusicPlayer::stopDecoder(void) void MusicPlayer::openOutputDevice(void) { - QString adevice; + QString adevice, pdevice; if (gContext->GetSetting("MusicAudioDevice") == "default") adevice = gContext->GetSetting("AudioOutputDevice"); else adevice = gContext->GetSetting("MusicAudioDevice"); + pdevice = gContext->GetSetting("PassThruOutputDevice"); + // TODO: Error checking that device is opened correctly! - m_output = AudioOutput::OpenAudio(adevice, "default", 16, 2, 44100, - AUDIOOUTPUT_MUSIC, true, false); + m_output = AudioOutput::OpenAudio(adevice, pdevice, 16, 2, 0, 44100, + AUDIOOUTPUT_MUSIC, true, false); m_output->setBufferSize(256 * 1024); m_output->SetBlocking(false); diff --git a/mythplugins/mythmusic/mythmusic/playbackbox.cpp b/mythplugins/mythmusic/mythmusic/playbackbox.cpp index f8a2bc7..0a6a061 100644 --- a/mythplugins/mythmusic/mythmusic/playbackbox.cpp +++ b/mythplugins/mythmusic/mythmusic/playbackbox.cpp @@ -367,6 +367,8 @@ void PlaybackBoxMusic::keyPressEvent(QKeyEvent *e) changeSpeed(true); else if (action == "MUTE") toggleMute(); + else if (action == "TOGGLEUPMIX") + toggleUpmix(); else if (action == "MENU" && visualizer_status != 2) { menufilters = false; @@ -1202,6 +1204,13 @@ void PlaybackBoxMusic::toggleMute() } } +void PlaybackBoxMusic::toggleUpmix() +{ + if (gPlayer->getOutput()) + gPlayer->getOutput()->ToggleUpmix(); +} + + void PlaybackBoxMusic::showProgressBar() { if (progress_bar && visualizer_status != 2) diff --git a/mythplugins/mythmusic/mythmusic/playbackbox.h b/mythplugins/mythmusic/mythmusic/playbackbox.h index ecc0207..b22c6ce 100644 --- a/mythplugins/mythmusic/mythmusic/playbackbox.h +++ b/mythplugins/mythmusic/mythmusic/playbackbox.h @@ -69,6 +69,7 @@ class PlaybackBoxMusic : public MythThemedDialog void changeVolume(bool up_or_down); void changeSpeed(bool up_or_down); void toggleMute(); + void toggleUpmix(); void resetTimer(); void hideVolume(){showVolume(false);} void showVolume(bool on_or_off); diff --git a/mythtv/libs/libavcodec/ac3dec.c b/mythtv/libs/libavcodec/ac3dec.c index 99cd4e4..99c99be 100644 --- a/mythtv/libs/libavcodec/ac3dec.c +++ b/mythtv/libs/libavcodec/ac3dec.c @@ -115,6 +115,8 @@ static const uint8_t ac3_default_coeffs[8][5][2] = { { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, }, }; +static const int channel_map[6] = { 0, 2, 3, 4, 1, 5 }; + /** * Symmetrical Dequantization * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization @@ -1225,6 +1227,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, int blk, ch, err; const uint8_t *channel_map; const float *output[AC3_MAX_CHANNELS]; + int16_t tmp; /* initialize the GetBitContext with the start of valid AC-3 Frame */ if (s->input_buffer) { @@ -1327,6 +1330,15 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, out_samples += 256 * s->out_channels; } *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t); + if (avctx->channels == 6) { + for (int i = 0; i < *data_size / sizeof(int16_t); i += avctx->channels) { + tmp = out_samples[i+1]; + out_samples[i+1] = out_samples[i+2]; + out_samples[i+2] = out_samples[i+3]; + out_samples[i+3] = out_samples[i+4]; + out_samples[i+4] = tmp; + } + } return s->frame_size; } diff --git a/mythtv/libs/libavcodec/dca.c b/mythtv/libs/libavcodec/dca.c index b42d3df..2f571d2 100644 --- a/mythtv/libs/libavcodec/dca.c +++ b/mythtv/libs/libavcodec/dca.c @@ -1241,6 +1241,7 @@ static int dca_decode_frame(AVCodecContext * avctx, int16_t *samples = data; DCAContext *s = avctx->priv_data; int channels; + int16_t tmp; s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); @@ -1297,7 +1298,18 @@ static int dca_decode_frame(AVCodecContext * avctx, s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); samples += 256 * channels; } - + + if (channels == 6) { + for (i = 0; i < *data_size / sizeof(int16_t); i += channels) { + tmp = samples[i]; + samples[i] = samples[i+1]; + samples[i+1] = samples[i+2]; + samples[i+2] = samples[i+3]; + samples[i+3] = samples[i+4]; + samples[i+4] = tmp; + } + } + return buf_size; } diff --git a/mythtv/libs/libmyth/audiooutput.cpp b/mythtv/libs/libmyth/audiooutput.cpp index 17f7c64..f97fe44 100644 --- a/mythtv/libs/libmyth/audiooutput.cpp +++ b/mythtv/libs/libmyth/audiooutput.cpp @@ -33,13 +33,14 @@ using namespace std; AudioOutput *AudioOutput::OpenAudio( const QString &main_device, const QString &passthru_device, - int audio_bits, int audio_channels, int audio_samplerate, + int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, AudioOutputSource source, bool set_initial_vol, bool audio_passthru) { AudioSettings settings( main_device, passthru_device, audio_bits, - audio_channels, audio_samplerate, source, + audio_channels, audio_codec, audio_samplerate, source, set_initial_vol, audio_passthru); settings.FixPassThrough(); diff --git a/mythtv/libs/libmyth/audiooutput.h b/mythtv/libs/libmyth/audiooutput.h index 943bd77..7c55ac0 100644 --- a/mythtv/libs/libmyth/audiooutput.h +++ b/mythtv/libs/libmyth/audiooutput.h @@ -15,7 +15,8 @@ class MPUBLIC AudioOutput : public VolumeBase, public OutputListeners static AudioOutput *OpenAudio( const QString &audiodevice, const QString &passthrudevice, - int audio_bits, int audio_channels, int audio_samplerate, + int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, AudioOutputSource source, bool set_initial_vol, bool audio_passthru); @@ -68,6 +69,7 @@ class MPUBLIC AudioOutput : public VolumeBase, public OutputListeners virtual void bufferOutputData(bool y) = 0; virtual int readOutputData(unsigned char *read_buffer, int max_length) = 0; + virtual bool ToggleUpmix(void) = 0; protected: void Error(const QString &msg); diff --git a/mythtv/libs/libmyth/audiooutputalsa.cpp b/mythtv/libs/libmyth/audiooutputalsa.cpp index 5bd6445..788ae18 100644 --- a/mythtv/libs/libmyth/audiooutputalsa.cpp +++ b/mythtv/libs/libmyth/audiooutputalsa.cpp @@ -32,6 +32,107 @@ AudioOutputALSA::AudioOutputALSA(const AudioSettings &settings) : AudioOutputALSA::~AudioOutputALSA() { KillAudio(); + SetIECStatus(true); +} + +void AudioOutputALSA::SetIECStatus(bool audio) { + + snd_ctl_t *ctl; + const char *spdif_str = SND_CTL_NAME_IEC958("", PLAYBACK, DEFAULT); + int spdif_index = -1; + snd_ctl_elem_list_t *clist; + snd_ctl_elem_id_t *cid; + snd_ctl_elem_value_t *cval; + snd_aes_iec958_t iec958; + int cidx, controls; + + VERBOSE(VB_AUDIO, QString("Setting IEC958 status: %1") + .arg(audio ? "audio" : "non-audio")); + + snd_ctl_open(&ctl, "default", 0); + snd_ctl_elem_list_alloca(&clist); + snd_ctl_elem_list(ctl, clist); + snd_ctl_elem_list_alloc_space(clist, snd_ctl_elem_list_get_count(clist)); + snd_ctl_elem_list(ctl, clist); + controls = snd_ctl_elem_list_get_used(clist); + for (cidx = 0; cidx < controls; cidx++) { + if (!strcmp(snd_ctl_elem_list_get_name(clist, cidx), spdif_str)) + if (spdif_index < 0 || + snd_ctl_elem_list_get_index(clist, cidx) == (uint)spdif_index) + break; + } + + if (cidx >= controls) + return; + + snd_ctl_elem_id_alloca(&cid); + snd_ctl_elem_list_get_id(clist, cidx, cid); + snd_ctl_elem_value_alloca(&cval); + snd_ctl_elem_value_set_id(cval, cid); + snd_ctl_elem_read(ctl,cval); + snd_ctl_elem_value_get_iec958(cval, &iec958); + + if (!audio) + iec958.status[0] |= IEC958_AES0_NONAUDIO; + else + iec958.status[0] &= ~IEC958_AES0_NONAUDIO; + + snd_ctl_elem_value_set_iec958(cval, &iec958); + snd_ctl_elem_write(ctl, cval); + +} + +vector AudioOutputALSA::GetSupportedRates() +{ + snd_pcm_hw_params_t *params; + int err; + const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; + vector rates(srates, srates + sizeof(srates) / sizeof(int) ); + QString real_device; + + if (audio_passthru || audio_enc) + real_device = audio_passthru_device; + else + real_device = audio_main_device; + + if((err = snd_pcm_open(&pcm_handle, real_device.toAscii(), + SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK|SND_PCM_NO_AUTO_RESAMPLE)) < 0) + { + Error(QString("snd_pcm_open(%1): %2") + .arg(real_device).arg(snd_strerror(err))); + + if (pcm_handle) + { + snd_pcm_close(pcm_handle); + pcm_handle = NULL; + } + rates.clear(); + return rates; + } + + snd_pcm_hw_params_alloca(¶ms); + + if ((err = snd_pcm_hw_params_any(pcm_handle, params)) < 0) + { + Error(QString("Broken configuration for playback; no configurations" + " available: %1").arg(snd_strerror(err))); + snd_pcm_close(pcm_handle); + pcm_handle = NULL; + rates.clear(); + return rates; + } + + vector::iterator it; + + for (it = rates.begin(); it < rates.end(); it++) + if(snd_pcm_hw_params_test_rate(pcm_handle, params, *it, 0) < 0) + rates.erase(it--); + + snd_pcm_close(pcm_handle); + pcm_handle = NULL; + + return rates; } bool AudioOutputALSA::OpenDevice() @@ -39,15 +140,24 @@ bool AudioOutputALSA::OpenDevice() snd_pcm_format_t format; unsigned int buffer_time, period_time; int err; + QString real_device; if (pcm_handle != NULL) CloseDevice(); pcm_handle = NULL; numbadioctls = 0; - - QString real_device = (audio_passthru) ? - audio_passthru_device : audio_main_device; + + if (audio_passthru || audio_enc) + { + real_device = audio_passthru_device; + SetIECStatus(false); + } + else + { + real_device = audio_main_device; + SetIECStatus(true); + } VERBOSE(VB_GENERAL, QString("Opening ALSA audio device '%1'.") .arg(real_device)); diff --git a/mythtv/libs/libmyth/audiooutputalsa.h b/mythtv/libs/libmyth/audiooutputalsa.h index a156edd..c260982 100644 --- a/mythtv/libs/libmyth/audiooutputalsa.h +++ b/mythtv/libs/libmyth/audiooutputalsa.h @@ -65,8 +65,10 @@ class AudioOutputALSA : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; + virtual vector GetSupportedRates(void); private: + void SetIECStatus(bool audio); inline int SetParameters(snd_pcm_t *handle, snd_pcm_format_t format, unsigned int channels, unsigned int rate, unsigned int buffer_time, diff --git a/mythtv/libs/libmyth/audiooutputarts.h b/mythtv/libs/libmyth/audiooutputarts.h index f985ef8..35ec39f 100644 --- a/mythtv/libs/libmyth/audiooutputarts.h +++ b/mythtv/libs/libmyth/audiooutputarts.h @@ -27,8 +27,8 @@ class AudioOutputARTS : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; - - + virtual vector GetSupportedRates(void) + { vector rates; return rates; } private: arts_stream_t pcm_handle; diff --git a/mythtv/libs/libmyth/audiooutputbase.cpp b/mythtv/libs/libmyth/audiooutputbase.cpp index ef2f3d5..63f90a4 100644 --- a/mythtv/libs/libmyth/audiooutputbase.cpp +++ b/mythtv/libs/libmyth/audiooutputbase.cpp @@ -21,16 +21,16 @@ AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : // protected effdsp(0), effdspstretched(0), - audio_channels(-1), audio_bytes_per_sample(0), - audio_bits(-1), audio_samplerate(-1), - audio_buffer_unused(0), + audio_channels(-1), audio_codec(CODEC_ID_NONE), + audio_bytes_per_sample(0), audio_bits(-1), + audio_samplerate(-1), audio_buffer_unused(0), fragment_size(0), soundcard_buffer_size(0), audio_main_device(settings.GetMainDevice()), audio_passthru_device(settings.GetPassthruDevice()), - audio_passthru(false), audio_stretchfactor(1.0f), + audio_passthru(false), audio_enc(false), + audio_reenc(false), audio_stretchfactor(1.0f), - audio_codec(NULL), source(settings.source), killaudio(false), pauseaudio(false), audio_actually_paused(false), @@ -48,9 +48,11 @@ AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : encoder(NULL), upmixer(NULL), source_audio_channels(-1), + source_audio_samplerate(0), source_audio_bytes_per_sample(0), needs_upmix(false), surround_mode(FreeSurround::SurroundModePassive), + old_audio_stretchfactor(1.0), blocking(false), @@ -79,6 +81,9 @@ AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : memset(&audiotime_updated, 0, sizeof(audiotime_updated)); memset(audiobuffer, 0, sizeof(char) * kAudioRingBufferSize); configured_audio_channels = gContext->GetNumSetting("MaxChannels", 2); + orig_config_channels = configured_audio_channels; + allow_ac3_passthru = gContext->GetNumSetting("AC3PassThru", false); + src_quality = gContext->GetNumSetting("SRCQuality", 3); // You need to call Reconfigure from your concrete class. // Reconfigure(laudio_bits, laudio_channels, @@ -124,40 +129,9 @@ void AudioOutputBase::SetStretchFactorLocked(float laudio_stretchfactor) VERBOSE(VB_GENERAL, LOC + QString("Using time stretch %1") .arg(audio_stretchfactor)); pSoundStretch = new soundtouch::SoundTouch(); - if (audio_codec) - { - if (!encoder) - { - VERBOSE(VB_AUDIO, LOC + - QString("Creating Encoder for codec %1 origfs %2") - .arg(audio_codec->codec_id) - .arg(audio_codec->frame_size)); - - encoder = new AudioOutputDigitalEncoder(); - if (!encoder->Init(audio_codec->codec_id, - audio_codec->bit_rate, - audio_codec->sample_rate, - audio_codec->channels - )) - { - // eeks - delete encoder; - encoder = NULL; - VERBOSE(VB_AUDIO, LOC + - QString("Failed to Create Encoder")); - } - } - } - if (audio_codec && encoder) - { - pSoundStretch->setSampleRate(audio_codec->sample_rate); - pSoundStretch->setChannels(audio_codec->channels); - } - else - { - pSoundStretch->setSampleRate(audio_samplerate); - pSoundStretch->setChannels(audio_channels); - } + pSoundStretch->setSampleRate(audio_samplerate); + pSoundStretch->setChannels(upmixer ? + configured_audio_channels : source_audio_channels); pSoundStretch->setTempo(audio_stretchfactor); pSoundStretch->setSetting(SETTING_SEQUENCE_MS, 35); @@ -165,7 +139,6 @@ void AudioOutputBase::SetStretchFactorLocked(float laudio_stretchfactor) // dont need these with only tempo change //pSoundStretch->setPitch(1.0); //pSoundStretch->setRate(1.0); - //pSoundStretch->setSetting(SETTING_USE_QUICKSEEK, true); //pSoundStretch->setSetting(SETTING_USE_AA_FILTER, false); } @@ -183,35 +156,43 @@ float AudioOutputBase::GetStretchFactor(void) const return audio_stretchfactor; } +bool AudioOutputBase::ToggleUpmix(void) +{ + if (orig_config_channels == 2 || source_audio_channels > 2 || + audio_passthru) + return false; + if (configured_audio_channels == 6) + configured_audio_channels = 2; + else + configured_audio_channels = 6; + + const AudioSettings settings(audio_bits, source_audio_channels, + audio_codec, source_audio_samplerate, + audio_passthru); + Reconfigure(settings); + return (configured_audio_channels == 6); +} + + void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) { AudioSettings settings = orig_settings; - int codec_id = CODEC_ID_NONE; - int lcodec_id = CODEC_ID_NONE; - int lcchannels = 0; - int cchannels = 0; int lsource_audio_channels = settings.channels; bool lneeds_upmix = false; + bool laudio_reenc = false; - if (settings.codec) + // Are we reencoding a (previously) timestretched bitstream? + if ((settings.codec == CODEC_ID_AC3 || settings.codec == CODEC_ID_DTS) && + !settings.use_passthru && allow_ac3_passthru) { - lcodec_id = ((AVCodecContext*)settings.codec)->codec_id; - settings.bits = 16; - settings.channels = 2; - lsource_audio_channels = settings.channels; - settings.samplerate = 48000; - lcchannels = ((AVCodecContext*)settings.codec)->channels; + laudio_reenc = true; + VERBOSE(VB_AUDIO, LOC + "Reencoding decoded AC3/DTS to AC3"); } - if (audio_codec) - { - codec_id = audio_codec->codec_id; - cchannels = ((AVCodecContext*)audio_codec)->channels; - } - - if ((configured_audio_channels == 6) && - !(settings.codec || audio_codec)) + // Enough channels? Upmix if not + if (settings.channels < configured_audio_channels && + !settings.use_passthru) { settings.channels = configured_audio_channels; lneeds_upmix = true; @@ -224,7 +205,7 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) settings.samplerate == audio_samplerate && !need_resampler && settings.use_passthru == audio_passthru && lneeds_upmix == needs_upmix && - lcodec_id == codec_id && lcchannels == cchannels); + laudio_reenc == audio_reenc); bool upmix_deps = (lsource_audio_channels == source_audio_channels); if (general_deps && upmix_deps) @@ -251,12 +232,11 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) waud = raud = 0; audio_actually_paused = false; - bool redo_stretch = (pSoundStretch && audio_channels != settings.channels); audio_channels = settings.channels; source_audio_channels = lsource_audio_channels; audio_bits = settings.bits; - audio_samplerate = settings.samplerate; - audio_codec = (AVCodecContext*)settings.codec; + source_audio_samplerate = audio_samplerate = settings.samplerate; + audio_reenc = laudio_reenc; audio_passthru = settings.use_passthru; needs_upmix = lneeds_upmix; @@ -265,8 +245,9 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) Error("AudioOutput only supports 8 or 16bit audio."); return; } - audio_bytes_per_sample = audio_channels * audio_bits / 8; - source_audio_bytes_per_sample = source_audio_channels * audio_bits / 8; + + VERBOSE(VB_AUDIO, LOC + QString("Original audio codec was %1") + .arg(codec_id_string((CodecID)audio_codec))); need_resampler = false; killaudio = false; @@ -275,7 +256,73 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) internal_vol = gContext->GetNumSetting("MythControlsVolume", 0); numlowbuffer = 0; + + // Find out what sample rates we can output (if output layer supports it) + vector rates = GetSupportedRates(); + vector::iterator it; + bool resample = true; + + for (it = rates.begin(); it < rates.end(); it++) + { + VERBOSE(VB_AUDIO, LOC + QString("Sample rate %1 is supported") + .arg(*it)); + if (*it == audio_samplerate) + resample = false; + } + + // Assume 48k if we can't get supported rates + if (rates.empty()) + rates.push_back(48000); + + if (resample) + { + int error; + audio_samplerate = *(rates.end()); + VERBOSE(VB_GENERAL, LOC + QString("Using resampler. From: %1 to %2") + .arg(settings.samplerate).arg(audio_samplerate)); + src_ctx = src_new(3-src_quality, source_audio_channels, &error); + if (error) + { + Error(QString("Error creating resampler, the error was: %1") + .arg(src_strerror(error)) ); + src_ctx = NULL; + return; + } + src_data.src_ratio = (double) audio_samplerate / settings.samplerate; + src_data.data_in = src_in; + src_data.data_out = src_out; + src_data.output_frames = 16384*6; + need_resampler = true; + } + + // Encode to AC-3 if not passing thru , there's > 2 channels + // and a passthru device is defined + if ( + !audio_passthru && allow_ac3_passthru && + (audio_channels > 2 || audio_reenc) + ) + { + VERBOSE(VB_AUDIO, LOC + "Creating AC-3 Encoder"); + encoder = new AudioOutputDigitalEncoder(); + if (!encoder->Init(CODEC_ID_AC3, 448000, audio_samplerate, + audio_channels)) + { + VERBOSE(VB_AUDIO, LOC + "Can't create AC-3 encoder"); + delete encoder; + encoder = NULL; + } + + audio_enc = true; + } + + if(audio_passthru || audio_enc) + // AC-3 output - soundcard expects a 2ch 48k stream + audio_channels = 2; + + audio_bytes_per_sample = audio_channels * audio_bits / 8; + source_audio_bytes_per_sample = source_audio_channels * audio_bits / 8; + VERBOSE(VB_GENERAL, QString("Opening audio device '%1'. ch %2(%3) sr %4") .arg(audio_main_device).arg(audio_channels) .arg(source_audio_channels).arg(audio_samplerate)); @@ -309,27 +356,6 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) current_seconds = -1; source_bitrate = -1; - // NOTE: this won't do anything as above samplerate vars are set equal - // Check if we need the resampler - if (audio_samplerate != settings.samplerate) - { - int error; - VERBOSE(VB_GENERAL, LOC + QString("Using resampler. From: %1 to %2") - .arg(settings.samplerate).arg(audio_samplerate)); - src_ctx = src_new (SRC_SINC_BEST_QUALITY, audio_channels, &error); - if (error) - { - Error(QString("Error creating resampler, the error was: %1") - .arg(src_strerror(error)) ); - return; - } - src_data.src_ratio = (double) audio_samplerate / settings.samplerate; - src_data.data_in = src_in; - src_data.data_out = src_out; - src_data.output_frames = 16384*6; - need_resampler = true; - } - if (needs_upmix) { VERBOSE(VB_AUDIO, LOC + QString("create upmixer")); @@ -344,63 +370,15 @@ void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) (FreeSurround::SurroundMode)surround_mode); VERBOSE(VB_AUDIO, LOC + - QString("create upmixer done with surround mode %1") + QString("Create upmixer done with surround mode %1") .arg(surround_mode)); } VERBOSE(VB_AUDIO, LOC + QString("Audio Stretch Factor: %1") .arg(audio_stretchfactor)); - VERBOSE(VB_AUDIO, QString("Audio Codec Used: %1") - .arg((audio_codec) ? - codec_id_string(audio_codec->codec_id) : "not set")); - - if (redo_stretch) - { - delete pSoundStretch; - pSoundStretch = NULL; - SetStretchFactorLocked(audio_stretchfactor); - } - else - { - SetStretchFactorLocked(audio_stretchfactor); - if (pSoundStretch) - { - // if its passthru then we need to reencode - if (audio_codec) - { - if (!encoder) - { - VERBOSE(VB_AUDIO, LOC + - QString("Creating Encoder for codec %1") - .arg(audio_codec->codec_id)); - - encoder = new AudioOutputDigitalEncoder(); - if (!encoder->Init(audio_codec->codec_id, - audio_codec->bit_rate, - audio_codec->sample_rate, - audio_codec->channels - )) - { - // eeks - delete encoder; - encoder = NULL; - VERBOSE(VB_AUDIO, LOC + "Failed to Create Encoder"); - } - } - } - if (audio_codec && encoder) - { - pSoundStretch->setSampleRate(audio_codec->sample_rate); - pSoundStretch->setChannels(audio_codec->channels); - } - else - { - pSoundStretch->setSampleRate(audio_samplerate); - pSoundStretch->setChannels(audio_channels); - } - } - } + SetStretchFactorLocked(old_audio_stretchfactor); + // Setup visualisations, zero the visualisations buffers prepareVisuals(); @@ -436,10 +414,15 @@ void AudioOutputBase::KillAudio() VERBOSE(VB_AUDIO, LOC + "Killing AudioOutputDSP"); killaudio = true; StopOutputThread(); + QMutexLocker lock1(&audio_buflock); // Close resampler? if (src_ctx) + { src_delete(src_ctx); + src_ctx = NULL; + } + need_resampler = false; // close sound stretcher @@ -447,6 +430,8 @@ void AudioOutputBase::KillAudio() { delete pSoundStretch; pSoundStretch = NULL; + old_audio_stretchfactor = audio_stretchfactor; + audio_stretchfactor = 1.0; } if (encoder) @@ -461,6 +446,7 @@ void AudioOutputBase::KillAudio() upmixer = NULL; } needs_upmix = false; + audio_enc = false; CloseDevice(); @@ -562,7 +548,6 @@ int AudioOutputBase::GetAudiotime(void) ret += (now.tv_usec - audiotime_updated.tv_usec) / 1000; ret = (long long)(ret * audio_stretchfactor); -#if 1 VERBOSE(VB_AUDIO+VB_TIMESTAMP, QString("GetAudiotime now=%1.%2, set=%3.%4, ret=%5, audt=%6 sf=%7") .arg(now.tv_sec).arg(now.tv_usec) @@ -571,7 +556,6 @@ int AudioOutputBase::GetAudiotime(void) .arg(audiotime) .arg(audio_stretchfactor) ); -#endif ret += audiotime; @@ -611,26 +595,20 @@ void AudioOutputBase::SetAudiotime(void) // include algorithmic latencies if (pSoundStretch) - { - // add the effect of any unused but processed samples, - // AC3 reencode does this - totalbuffer += (int)(pSoundStretch->numSamples() * - audio_bytes_per_sample); - // add the effect of unprocessed samples in time stretch algo totalbuffer += (int)((pSoundStretch->numUnprocessedSamples() * audio_bytes_per_sample) / audio_stretchfactor); - } if (upmixer && needs_upmix) - { totalbuffer += upmixer->sampleLatency() * audio_bytes_per_sample; - } + + if (encoder) + totalbuffer += encoder->Buffered(); audiotime = audbuf_timecode - (int)(totalbuffer * 100000.0 / (audio_bytes_per_sample * effdspstretched)); gettimeofday(&audiotime_updated, NULL); -#if 1 + VERBOSE(VB_AUDIO+VB_TIMESTAMP, QString("SetAudiotime set=%1.%2, audt=%3 atc=%4 " "tb=%5 sb=%6 eds=%7 abps=%8 sf=%9") @@ -642,7 +620,6 @@ void AudioOutputBase::SetAudiotime(void) .arg(effdspstretched) .arg(audio_bytes_per_sample) .arg(audio_stretchfactor)); -#endif } int AudioOutputBase::GetAudioBufferedTime(void) @@ -681,6 +658,8 @@ bool AudioOutputBase::AddSamples(char *buffers[], int samples, return false; // would overflow } + QMutexLocker lock1(&audio_buflock); + // resample input if necessary if (need_resampler && src_ctx) { @@ -725,6 +704,10 @@ bool AudioOutputBase::AddSamples(char *buffer, int samples, long long timecode) int abps = (encoder) ? encoder->audio_bytes_per_sample : audio_bytes_per_sample; int len = samples * abps; + + // Give original samples to mythmusic visualisation + dispatchVisual((unsigned char *)buffer, len, timecode, + source_audio_channels, audio_bits); // Check we have enough space to write the data if (need_resampler && src_ctx) @@ -749,6 +732,8 @@ bool AudioOutputBase::AddSamples(char *buffer, int samples, long long timecode) return false; // would overflow } + QMutexLocker lock1(&audio_buflock); + // resample input if necessary if (need_resampler && src_ctx) { @@ -808,10 +793,13 @@ int AudioOutputBase::WaitForFreeSpace(int samples) if (src_ctx) { int error = src_reset(src_ctx); - if (error) + if (error) + { VERBOSE(VB_IMPORTANT, LOC_ERR + QString( "Error occured while resetting resampler: %1") .arg(src_strerror(error))); + src_ctx = NULL; + } } } } @@ -821,8 +809,6 @@ int AudioOutputBase::WaitForFreeSpace(int samples) void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, long long timecode) { - audio_buflock.lock(); - int len; // = samples * audio_bytes_per_sample; int audio_bytes = audio_bits / 8; int org_waud = waud; @@ -839,17 +825,17 @@ void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, .arg(samples * abps) .arg(kAudioRingBufferSize-afree).arg(afree).arg(timecode) .arg(needs_upmix)); - + + len = WaitForFreeSpace(samples); + if (upmixer && needs_upmix) { int out_samples = 0; + org_waud = waud; int step = (interleaved)?source_audio_channels:1; - len = WaitForFreeSpace(samples); // test + for (int itemp = 0; itemp < samples; ) { - // just in case it does a processing cycle, release the lock - // to allow the output loop to do output - audio_buflock.unlock(); if (audio_bytes == 2) { itemp += upmixer->putSamples( @@ -866,7 +852,6 @@ void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, source_audio_channels, (interleaved) ? 0 : samples); } - audio_buflock.lock(); int copy_samples = upmixer->numSamples(); if (copy_samples) @@ -900,8 +885,6 @@ void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, } else { - len = WaitForFreeSpace(samples); - if (interleaved) { char *mybuf = (char*)buffer; @@ -936,138 +919,106 @@ void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, } } - if (samples > 0) + if (samples <= 0) + return; + + if (pSoundStretch) { - if (pSoundStretch) + // does not change the timecode, only the number of samples + // back to orig pos + org_waud = waud; + int bdiff = kAudioRingBufferSize - org_waud; + int nSamplesToEnd = bdiff/abps; + if (bdiff < len) { + pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) + (audiobuffer + + org_waud), nSamplesToEnd); + pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer, + (len - bdiff) / abps); + } + else + { + pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) + (audiobuffer + org_waud), + len / abps); + } - // does not change the timecode, only the number of samples - // back to orig pos - org_waud = waud; - int bdiff = kAudioRingBufferSize - org_waud; - int nSamplesToEnd = bdiff/abps; - if (bdiff < len) - { - pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) - (audiobuffer + - org_waud), nSamplesToEnd); - pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer, - (len - bdiff) / abps); + int nSamples = pSoundStretch->numSamples(); + len = WaitForFreeSpace(nSamples); + + while ((nSamples = pSoundStretch->numSamples())) + { + if (nSamples > nSamplesToEnd) + nSamples = nSamplesToEnd; + + nSamples = pSoundStretch->receiveSamples( + (soundtouch::SAMPLETYPE*) + (audiobuffer + org_waud), nSamples + ); + + if (nSamples == nSamplesToEnd) { + org_waud = 0; + nSamplesToEnd = kAudioRingBufferSize/abps; } - else - { - pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) - (audiobuffer + org_waud), - len / abps); + else { + org_waud += nSamples * abps; + nSamplesToEnd -= nSamples; } + + } + + } - if (encoder) - { - // pull out a packet's worth and reencode it until we - // don't have enough for any more packets - soundtouch::SAMPLETYPE *temp_buff = - (soundtouch::SAMPLETYPE*)encoder->GetFrameBuffer(); - size_t frameSize = encoder->FrameSize()/abps; + // Encode to AC-3? + if (encoder) + { + + org_waud = waud; + int bdiff = kAudioRingBufferSize - org_waud; + int to_get = 0; - VERBOSE(VB_AUDIO+VB_TIMESTAMP, - QString("_AddSamples Enc sfs=%1 bfs=%2 sss=%3") - .arg(frameSize) - .arg(encoder->FrameSize()) - .arg(pSoundStretch->numSamples())); - - // process the same number of samples as it creates - // a full encoded buffer just like before - while (pSoundStretch->numSamples() >= frameSize) - { - int got = pSoundStretch->receiveSamples( - temp_buff, frameSize); - int amount = encoder->Encode(temp_buff); - - VERBOSE(VB_AUDIO+VB_TIMESTAMP, - QString("_AddSamples Enc bytes=%1 got=%2 left=%3") - .arg(amount) - .arg(got) - .arg(pSoundStretch->numSamples())); - - if (!amount) - continue; - - //len = WaitForFreeSpace(amount); - char *ob = encoder->GetOutBuff(); - if (amount >= bdiff) - { - memcpy(audiobuffer + org_waud, ob, bdiff); - ob += bdiff; - amount -= bdiff; - org_waud = 0; - } - if (amount > 0) - memcpy(audiobuffer + org_waud, ob, amount); - - bdiff = kAudioRingBufferSize - amount; - org_waud = (org_waud + amount) % kAudioRingBufferSize; - } - } - else + if (bdiff < len) + { + encoder->Encode(audiobuffer + org_waud, bdiff); + to_get = encoder->Encode(audiobuffer, len - bdiff); + } + else + to_get = encoder->Encode(audiobuffer + org_waud, len); + + if (to_get > 0) + { + + if (to_get >= bdiff) { - int newLen = 0; - int nSamples; - len = WaitForFreeSpace(pSoundStretch->numSamples() * - audio_bytes_per_sample); - do - { - int samplesToGet = len/audio_bytes_per_sample; - if (samplesToGet > nSamplesToEnd) - { - samplesToGet = nSamplesToEnd; - } - - nSamples = pSoundStretch->receiveSamples( - (soundtouch::SAMPLETYPE*) - (audiobuffer + org_waud), samplesToGet); - if (nSamples == nSamplesToEnd) - { - org_waud = 0; - nSamplesToEnd = kAudioRingBufferSize/audio_bytes_per_sample; - } - else - { - int bufsz = nSamples * audio_bytes_per_sample; - org_waud = (org_waud + bufsz) % kAudioRingBufferSize; - nSamplesToEnd -= nSamples; - } - - newLen += nSamples * audio_bytes_per_sample; - len -= nSamples * audio_bytes_per_sample; - } while (nSamples > 0); + encoder->GetFrames(audiobuffer + org_waud, bdiff); + to_get -= bdiff; + org_waud = 0; } - } + if (to_get > 0) + encoder->GetFrames(audiobuffer + org_waud, to_get); - waud = org_waud; - lastaudiolen = audiolen(false); + org_waud += to_get; - if (timecode < 0) - { - // mythmusic doesn't give timestamps.. - timecode = (int)((samples_buffered * 100000.0) / effdsp); } - - samples_buffered += samples; - - /* we want the time at the end -- but the file format stores - time at the start of the chunk. */ - // even with timestretch, timecode is still calculated from original - // sample count - audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp); - if (interleaved) - { - dispatchVisual((unsigned char *)buffer, len, timecode, - source_audio_channels, audio_bits); - } } - audio_buflock.unlock(); + waud = org_waud; + lastaudiolen = audiolen(false); + + if (timecode < 0) + // mythmusic doesn't give timestamps.. + timecode = (int)((samples_buffered * 100000.0) / effdsp); + + samples_buffered += samples; + + /* we want the time at the end -- but the file format stores + time at the start of the chunk. */ + // even with timestretch, timecode is still calculated from original + // sample count + audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp); + } void AudioOutputBase::Status() diff --git a/mythtv/libs/libmyth/audiooutputbase.h b/mythtv/libs/libmyth/audiooutputbase.h index 1f636e2..4bcaaa4 100644 --- a/mythtv/libs/libmyth/audiooutputbase.h +++ b/mythtv/libs/libmyth/audiooutputbase.h @@ -43,6 +43,7 @@ class AudioOutputBase : public AudioOutput, public QThread virtual void SetStretchFactor(float factor); virtual float GetStretchFactor(void) const; + virtual bool ToggleUpmix(void); virtual void Reset(void); @@ -85,6 +86,8 @@ class AudioOutputBase : public AudioOutput, public QThread virtual void WriteAudio(unsigned char *aubuf, int size) = 0; virtual int GetSpaceOnSoundcard(void) const = 0; virtual int GetBufferedOnSoundcard(void) const = 0; + virtual vector GetSupportedRates(void) = 0; + /// You need to call this from any implementation in the dtor. void KillAudio(void); @@ -122,6 +125,7 @@ class AudioOutputBase : public AudioOutput, public QThread // Basic details about the audio stream int audio_channels; + int audio_codec; int audio_bytes_per_sample; int audio_bits; int audio_samplerate; @@ -132,9 +136,10 @@ class AudioOutputBase : public AudioOutput, public QThread QString audio_passthru_device; bool audio_passthru; + bool audio_enc; + bool audio_reenc; float audio_stretchfactor; - AVCodecContext *audio_codec; AudioOutputSource source; bool killaudio; @@ -144,6 +149,8 @@ class AudioOutputBase : public AudioOutput, public QThread bool buffer_output_data_for_use; // used by AudioOutputNULL int configured_audio_channels; + int orig_config_channels; + int src_quality; private: // resampler @@ -156,9 +163,12 @@ class AudioOutputBase : public AudioOutput, public QThread FreeSurround *upmixer; int source_audio_channels; + int source_audio_samplerate; int source_audio_bytes_per_sample; bool needs_upmix; int surround_mode; + bool allow_ac3_passthru; + float old_audio_stretchfactor; bool blocking; // do AddSamples calls block? diff --git a/mythtv/libs/libmyth/audiooutputca.h b/mythtv/libs/libmyth/audiooutputca.h index 5afe3e0..14192bb 100644 --- a/mythtv/libs/libmyth/audiooutputca.h +++ b/mythtv/libs/libmyth/audiooutputca.h @@ -49,6 +49,8 @@ protected: virtual bool StartOutputThread(void) { return true; } virtual void StopOutputThread(void) {} + virtual vector GetSupportedRates(void) + { vector rates; return rates; } private: diff --git a/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp b/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp index a9688c9..f3bce4c 100644 --- a/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp +++ b/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp @@ -32,9 +32,8 @@ extern "C" { AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) : audio_bytes_per_sample(0), av_context(NULL), - outbuf(NULL), - outbuf_size(0), - frame_buffer(NULL), + outbuflen(0), + inbuflen(0), one_frame_bytes(0) { } @@ -52,20 +51,6 @@ void AudioOutputDigitalEncoder::Dispose() av_free(av_context); av_context = NULL; } - - if (outbuf) - { - delete [] outbuf; - outbuf = NULL; - outbuf_size = 0; - } - - if (frame_buffer) - { - delete [] frame_buffer; - frame_buffer = NULL; - one_frame_bytes = 0; - } } //CODEC_ID_AC3 @@ -80,8 +65,10 @@ bool AudioOutputDigitalEncoder::Init( .arg(bitrate) .arg(samplerate) .arg(channels)); - - //codec = avcodec_find_encoder(codec_id); + + // We need to do this when called from mythmusic + avcodec_init(); + avcodec_register_all(); // always AC3 as there is no DTS encoder at the moment 2005/1/9 codec = avcodec_find_encoder(CODEC_ID_AC3); if (!codec) @@ -110,8 +97,6 @@ bool AudioOutputDigitalEncoder::Init( audio_bytes_per_sample = bytes_per_frame; one_frame_bytes = bytes_per_frame * av_context->frame_size; - outbuf_size = 16384; // ok for AC3 but DTS? - outbuf = new char [outbuf_size]; VERBOSE(VB_AUDIO, QString("DigitalEncoder::Init fs=%1, bpf=%2 ofb=%3") .arg(av_context->frame_size) .arg(bytes_per_frame) @@ -256,13 +241,26 @@ typedef struct { } AESHeader; +void reorder_6ch_ac3(void *buf, unsigned int len) { + unsigned short *src = (unsigned short *)buf; + unsigned short tmp; + unsigned int samples = len >> 1; + + for (uint i = 0; i < samples; i += 6) { + tmp = src[i+4]; + src[i+4] = src[i+3]; + src[i+3] = src[i+2]; + src[i+2] = src[i+1]; + src[i+1] = tmp; + } +} + static int encode_frame( bool dts, unsigned char *data, - size_t &len) + size_t enc_len) { unsigned char *payload = data + 8; // skip header, currently 52 or 54bits - size_t enc_len; int flags, sample_rate, bit_rate; // we don't do any length/crc validation of the AC3 frame here; presumably @@ -273,7 +271,8 @@ static int encode_frame( // ignore, and if so, may as well just assume that it will ignore // anything with a bad CRC... - uint nr_samples = 0, block_len; + uint nr_samples = 0, block_len = 0; + if (dts) { enc_len = dts_syncinfo(payload, &flags, &sample_rate, &bit_rate); @@ -302,13 +301,6 @@ static int encode_frame( } } - if (enc_len == 0 || enc_len > len) - { - int l = len; - len = 0; - return l; - } - enc_len = std::min((uint)enc_len, block_len - 8); //uint32_t x = *(uint32_t*)payload; @@ -361,31 +353,45 @@ static int encode_frame( data[6] = (enc_len << 3) & 0xFF; data[7] = (enc_len >> 5) & 0xFF; memset(payload + enc_len, 0, block_len - 8 - enc_len); - len = block_len; return enc_len; } // must have exactly 1 frames worth of data -size_t AudioOutputDigitalEncoder::Encode(short *buff) +size_t AudioOutputDigitalEncoder::Encode(void *buf, int len) { - int encsize = 0; size_t outsize = 0; - // put data in the correct spot for encode frame - outsize = avcodec_encode_audio( - av_context, ((uchar*)outbuf) + 8, outbuf_size - 8, buff); - - size_t tmpsize = outsize; - - outsize = MAX_AC3_FRAME_SIZE; - encsize = encode_frame( - /*av_context->codec_id==CODEC_ID_DTS*/ false, - (unsigned char*)outbuf, outsize); + int fs = FrameSize(); + memcpy(inbuf+inbuflen, buf, len); + inbuflen += len; + int frames = inbuflen / fs; - VERBOSE(VB_AUDIO+VB_TIMESTAMP, - QString("DigitalEncoder::Encode len1=%1 len2=%2 finallen=%3") - .arg(tmpsize).arg(encsize).arg(outsize)); + while (frames--) + { + reorder_6ch_ac3(inbuf, fs); + + // put data in the correct spot for encode frame + outsize = avcodec_encode_audio( + av_context, ((uchar*)outbuf) + outbuflen + 8, OUTBUFSIZE - 8, (short int *)inbuf); + + encode_frame( + /*av_context->codec_id==CODEC_ID_DTS*/ false, + (unsigned char*)outbuf + outbuflen, outsize + ); + + outbuflen += MAX_AC3_FRAME_SIZE; + inbuflen -= fs; + memmove(inbuf, inbuf+fs, inbuflen); + } + + return outbuflen; +} - return outsize; +void AudioOutputDigitalEncoder::GetFrames(void *ptr, int maxlen) +{ + int len = (maxlen < outbuflen ? maxlen : outbuflen); + memcpy(ptr, outbuf, len); + outbuflen -= len; + memmove(outbuf, outbuf+len, outbuflen); } diff --git a/mythtv/libs/libmyth/audiooutputdigitalencoder.h b/mythtv/libs/libmyth/audiooutputdigitalencoder.h index 8a4689a..8d81298 100644 --- a/mythtv/libs/libmyth/audiooutputdigitalencoder.h +++ b/mythtv/libs/libmyth/audiooutputdigitalencoder.h @@ -5,6 +5,9 @@ extern "C" { #include "libavcodec/avcodec.h" }; +#define INBUFSIZE 131072 +#define OUTBUFSIZE 98304 + class AudioOutputDigitalEncoder { public: @@ -13,29 +16,21 @@ class AudioOutputDigitalEncoder bool Init(CodecID codec_id, int bitrate, int samplerate, int channels); void Dispose(void); - size_t Encode(short * buff); - - inline char *GetFrameBuffer(void); + size_t Encode(void *buf, int len); + void GetFrames(void *ptr, int maxlen); size_t FrameSize(void) const { return one_frame_bytes; } - char *GetOutBuff(void) const { return outbuf; } + int Buffered(void) const { return inbuflen; } public: size_t audio_bytes_per_sample; private: AVCodecContext *av_context; - char *outbuf; - int outbuf_size; - char *frame_buffer; + char outbuf[OUTBUFSIZE]; + char inbuf[INBUFSIZE]; + int outbuflen; + int inbuflen; size_t one_frame_bytes; }; -inline char *AudioOutputDigitalEncoder::GetFrameBuffer(void) -{ - if (!frame_buffer && av_context) - frame_buffer = new char [one_frame_bytes]; - - return frame_buffer; -} - #endif diff --git a/mythtv/libs/libmyth/audiooutputdx.h b/mythtv/libs/libmyth/audiooutputdx.h index e0d3e24..485adb1 100644 --- a/mythtv/libs/libmyth/audiooutputdx.h +++ b/mythtv/libs/libmyth/audiooutputdx.h @@ -45,6 +45,8 @@ public: // Volume control virtual int GetVolumeChannel(int channel) const; // Returns 0-100 virtual void SetVolumeChannel(int channel, int volume); // range 0-100 for vol + virtual vector GetSupportedRates(void) + { vector rates; return rates; } private: HINSTANCE dsound_dll; /* handle of the opened dsound dll */ diff --git a/mythtv/libs/libmyth/audiooutputjack.cpp b/mythtv/libs/libmyth/audiooutputjack.cpp index 7e0ad0d..f9a5848 100644 --- a/mythtv/libs/libmyth/audiooutputjack.cpp +++ b/mythtv/libs/libmyth/audiooutputjack.cpp @@ -30,6 +30,42 @@ AudioOutputJACK::AudioOutputJACK(const AudioSettings &settings) : Reconfigure(settings); } +vector AudioOutputJACK::GetSupportedRates() +{ + const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; + vector rates(srates, srates + sizeof(srates) / sizeof(int) ); + unsigned long jack_port_flags = 0; + unsigned int jack_port_name_count = 1; + const char *jack_port_name = audio_main_device.ascii(); + int err = -1; + audioid = -1; + vector::iterator it; + + for (it = rates.begin(); it < rates.end(); it++) + { + err = JACK_OpenEx(&audioid, 16, (unsigned long *)it, + 2, 2, &jack_port_name, jack_port_name_count, + jack_port_flags); + + if (err == 1) + { + Error(QString("Error connecting to jackd: %1. Is it running?") + .arg(audio_main_device)); + rates.clear(); + return rates; + } + else if (err == 2) + rates.erase(it--); + + JACK_Close(audioid); + audioid = -1; + + } + + return rates; +} + + AudioOutputJACK::~AudioOutputJACK() { // Close down all audio stuff diff --git a/mythtv/libs/libmyth/audiooutputjack.h b/mythtv/libs/libmyth/audiooutputjack.h index f87538e..259c202 100644 --- a/mythtv/libs/libmyth/audiooutputjack.h +++ b/mythtv/libs/libmyth/audiooutputjack.h @@ -23,6 +23,7 @@ class AudioOutputJACK : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; + virtual vector GetSupportedRates(void); private: diff --git a/mythtv/libs/libmyth/audiooutputnull.h b/mythtv/libs/libmyth/audiooutputnull.h index 7eb247d..78a0f54 100644 --- a/mythtv/libs/libmyth/audiooutputnull.h +++ b/mythtv/libs/libmyth/audiooutputnull.h @@ -41,6 +41,8 @@ class AudioOutputNULL : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; + virtual vector GetSupportedRates(void) + { vector rates; return rates; } private: QMutex pcm_output_buffer_mutex; diff --git a/mythtv/libs/libmyth/audiooutputoss.cpp b/mythtv/libs/libmyth/audiooutputoss.cpp index 127bf6e..8d3b135 100644 --- a/mythtv/libs/libmyth/audiooutputoss.cpp +++ b/mythtv/libs/libmyth/audiooutputoss.cpp @@ -42,6 +42,32 @@ AudioOutputOSS::~AudioOutputOSS() KillAudio(); } +vector AudioOutputOSS::GetSupportedRates() +{ + const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; + vector rates(srates, srates + sizeof(srates) / sizeof(int) ); + audiofd = open(audio_main_device.toAscii(), O_WRONLY | O_NONBLOCK); + + if (audiofd < 0) + { + VERBOSE(VB_IMPORTANT, QString("Error opening audio device (%1), the" + " error was: %2").arg(audio_main_device).arg(strerror(errno))); + rates.clear(); + return rates; + } + + vector::iterator it; + + for (it = rates.begin(); it < rates.end(); it++) + if(ioctl(audiofd, SNDCTL_DSP_SPEED, &audio_samplerate) < 0) + rates.erase(it--); + + close(audiofd); + audiofd = -1; + + return rates; +} + bool AudioOutputOSS::OpenDevice() { numbadioctls = 0; diff --git a/mythtv/libs/libmyth/audiooutputoss.h b/mythtv/libs/libmyth/audiooutputoss.h index d5105a2..e218e31 100644 --- a/mythtv/libs/libmyth/audiooutputoss.h +++ b/mythtv/libs/libmyth/audiooutputoss.h @@ -23,6 +23,7 @@ class AudioOutputOSS : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; + virtual vector GetSupportedRates(void); private: void VolumeInit(void); diff --git a/mythtv/libs/libmyth/audiooutputwin.h b/mythtv/libs/libmyth/audiooutputwin.h index 8ba37e4..5fd5945 100644 --- a/mythtv/libs/libmyth/audiooutputwin.h +++ b/mythtv/libs/libmyth/audiooutputwin.h @@ -23,6 +23,8 @@ class AudioOutputWin : public AudioOutputBase virtual void WriteAudio(unsigned char *aubuf, int size); virtual int GetSpaceOnSoundcard(void) const; virtual int GetBufferedOnSoundcard(void) const; + virtual vector GetSupportedRates(void) + { vector rates; return rates; } protected: AudioOutputWinPrivate *m_priv; diff --git a/mythtv/libs/libmyth/audiosettings.cpp b/mythtv/libs/libmyth/audiosettings.cpp index 68e515b..1f4ef18 100644 --- a/mythtv/libs/libmyth/audiosettings.cpp +++ b/mythtv/libs/libmyth/audiosettings.cpp @@ -12,10 +12,10 @@ AudioSettings::AudioSettings() : passthru_device(QString::null), bits(-1), channels(-1), + codec(0), samplerate(-1), set_initial_vol(false), use_passthru(false), - codec(NULL), source(AUDIOOUTPUT_UNKNOWN) { } @@ -25,10 +25,10 @@ AudioSettings::AudioSettings(const AudioSettings &other) : passthru_device(other.passthru_device), bits(other.bits), channels(other.channels), + codec(other.codec), samplerate(other.samplerate), set_initial_vol(other.set_initial_vol), use_passthru(other.use_passthru), - codec(other.codec), source(other.source) { } @@ -38,19 +38,19 @@ AudioSettings::AudioSettings( const QString &audio_passthru_device, int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, AudioOutputSource audio_source, bool audio_set_initial_vol, - bool audio_use_passthru, - void *audio_codec) : + bool audio_use_passthru) : main_device(audio_main_device), passthru_device(audio_passthru_device), bits(audio_bits), channels(audio_channels), + codec(audio_codec), samplerate(audio_samplerate), set_initial_vol(audio_set_initial_vol), use_passthru(audio_use_passthru), - codec(audio_codec), source(audio_source) { } @@ -58,17 +58,17 @@ AudioSettings::AudioSettings( AudioSettings::AudioSettings( int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, - bool audio_use_passthru, - void *audio_codec) : + bool audio_use_passthru) : main_device(QString::null), passthru_device(QString::null), bits(audio_bits), channels(audio_channels), + codec(audio_codec), samplerate(audio_samplerate), set_initial_vol(false), use_passthru(audio_use_passthru), - codec(audio_codec), source(AUDIOOUTPUT_UNKNOWN) { } diff --git a/mythtv/libs/libmyth/audiosettings.h b/mythtv/libs/libmyth/audiosettings.h index f9349f4..ba38f51 100644 --- a/mythtv/libs/libmyth/audiosettings.h +++ b/mythtv/libs/libmyth/audiosettings.h @@ -29,17 +29,17 @@ class MPUBLIC AudioSettings const QString &audio_passthru_device, int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, AudioOutputSource audio_source, bool audio_set_initial_vol, - bool audio_use_passthru, - void *audio_codec = NULL); + bool audio_use_passthru); AudioSettings(int audio_bits, int audio_channels, + int audio_codec, int audio_samplerate, - bool audio_use_passthru, - void *audio_codec = NULL); + bool audio_use_passthru); void FixPassThrough(void); void TrimDeviceType(void); @@ -54,10 +54,10 @@ class MPUBLIC AudioSettings public: int bits; int channels; + int codec; int samplerate; bool set_initial_vol; bool use_passthru; - void *codec; AudioOutputSource source; }; diff --git a/mythtv/libs/libmythfreesurround/el_processor.cpp b/mythtv/libs/libmythfreesurround/el_processor.cpp index 8f24737..1cf0dc8 100644 --- a/mythtv/libs/libmythfreesurround/el_processor.cpp +++ b/mythtv/libs/libmythfreesurround/el_processor.cpp @@ -40,17 +40,7 @@ typedef std::complex cfloat; const float PI = 3.141592654; const float epsilon = 0.000001; -//const float center_level = 0.5*sqrt(0.5); // gain of the center channel -//const float center_level = sqrt(0.5); // gain of the center channel -const float center_level = 1.0; // gain of the center channel -//const float center_level = 0.5; // gain of the center channel - -// should be .6-.7 -// but with centerlevel 2x what its supposed to be, we halve 0.68 -// to keep center from clipping -//const float window_gain = 0.34; -//const float window_gain = 0.68; -const float window_gain = 0.95; // to prive a bit of margin +const float center_level = 0.5*sqrt(0.5); // private implementation of the surround decoder class decoder_impl { @@ -98,19 +88,11 @@ public: outbuf[c].resize(N); filter[c].resize(N); } - // DC component of filters is always 0 - for (unsigned c=0;c<5;c++) - { - filter[c][0] = 0.0; - filter[c][1] = 0.0; - filter[c][halfN] = 0.0; - } sample_rate(48000); // generate the window function (square root of hann, b/c it is applied before and after the transform) wnd.resize(N); - // dft normalization included in the window for zero cost scaling - // also add a gain factor of *2 due to processing gain in algo (see center_level) - surround_gain(1.0); + for (unsigned k=0;k=2) && (f= -1 && x1.real() <= 1) - x = x1.real(); - else if (x2.real() >= -1 && x2.real() <= 1) - x = x2.real();*/ - - //cfloat yp = (rt - (x*E+H))/(F+x*G); - //cfloat xp = (lt - (y*B+D))/(A+y*C); - - /*xfs[f] = x; - yfs[f] = y.real();*/ - // --- this is the fancy new linear mode --- // get sound field x/y position @@ -597,7 +538,6 @@ private: float surround_high,surround_low; // high and low surround mixing coefficient (e.g. 0.8165/0.5774) float surround_balance; // the xfs balance that follows from the coeffs float surround_level; // gain for the surround channels (follows from the coeffs - float master_gain; // gain for all channels float phase_offsetL, phase_offsetR;// phase shifts to be applied to the rear channels float front_separation; // front stereo separation float rear_separation; // rear stereo separation @@ -625,8 +565,6 @@ void fsurround_decoder::flush() { impl->flush(); } void fsurround_decoder::surround_coefficients(float a, float b) { impl->surround_coefficients(a,b); } -void fsurround_decoder::gain(float gain) { impl->surround_gain(gain); } - void fsurround_decoder::phase_mode(unsigned mode) { impl->phase_mode(mode); } void fsurround_decoder::steering_mode(bool mode) { impl->steering_mode(mode); } diff --git a/mythtv/libs/libmythfreesurround/el_processor.h b/mythtv/libs/libmythfreesurround/el_processor.h index 021786a..26452f6 100644 --- a/mythtv/libs/libmythfreesurround/el_processor.h +++ b/mythtv/libs/libmythfreesurround/el_processor.h @@ -47,9 +47,6 @@ public: // a is the coefficient of left rear in left total, b is the coefficient of left rear in right total; the same is true for right. void surround_coefficients(float a, float b); - // override for master surround gain - void gain(float gain); - // set the phase shifting mode for decoding // 0 = (+0°,+0°) - music mode // 1 = (+0°,+180°) - PowerDVD compatibility diff --git a/mythtv/libs/libmythfreesurround/freesurround.cpp b/mythtv/libs/libmythfreesurround/freesurround.cpp index fc7ed8f..bf93d28 100644 --- a/mythtv/libs/libmythfreesurround/freesurround.cpp +++ b/mythtv/libs/libmythfreesurround/freesurround.cpp @@ -63,10 +63,9 @@ using namespace std; const unsigned default_block_size = 8192; // there will be a slider for this in the future //const float master_gain = 1.0; -//#define MASTER_GAIN * master_gain +//#define MASTER_GAIN * master_gain #define MASTER_GAIN -//const float master_gain = 1.0/(1<<15); -//const float inv_master_gain = (1<<15); +//const float inv_master_gain = 1.0; //#define INV_MASTER_GAIN * inv_master_gain #define INV_MASTER_GAIN @@ -192,15 +191,13 @@ FreeSurround::FreeSurround(uint srate, bool moviemode, SurroundMode smode) : if (moviemode) { params.phasemode = 1; - params.center_width = 0; - params.gain = 1.0; + params.center_width = 25; + params.dimension = 0.5; } else { - params.center_width = 70; - // for 50, gain should be about 1.9, c/lr about 2.7 - // for 70, gain should be about 3.1, c/lr about 1.5 - params.gain = 3.1; + params.center_width = 65; + params.dimension = 0.3; } switch (surround_mode) { @@ -236,7 +233,6 @@ void FreeSurround::SetParams() decoder->phase_mode(params.phasemode); decoder->surround_coefficients(params.coeff_a, params.coeff_b); decoder->separation(params.front_sep/100.0,params.rear_sep/100.0); - decoder->gain(params.gain); } } @@ -250,8 +246,7 @@ FreeSurround::fsurround_params::fsurround_params( phasemode(0), steering(1), front_sep(100), - rear_sep(100), - gain(1.0) + rear_sep(100) { } @@ -655,16 +650,6 @@ void FreeSurround::process_block() { if (decoder) { - // actually these params need only be set when they change... but it doesn't hurt -#if 0 - decoder->steering_mode(params.steering); - decoder->phase_mode(params.phasemode); - decoder->surround_coefficients(params.coeff_a, params.coeff_b); - decoder->separation(params.front_sep/100.0,params.rear_sep/100.0); -#endif - // decode the bufs->block - //decoder->decode(input,output,params.center_width/100.0,params.dimension/100.0); - //decoder->decode(output,params.center_width/100.0,params.dimension/100.0); decoder->decode(params.center_width/100.0,params.dimension/100.0); } } diff --git a/mythtv/libs/libmythsamplerate/samplerate.c b/mythtv/libs/libmythsamplerate/samplerate.c index adaccf0..a53a942 100644 --- a/mythtv/libs/libmythsamplerate/samplerate.c +++ b/mythtv/libs/libmythsamplerate/samplerate.c @@ -452,11 +452,11 @@ src_float_to_short_array (const float *in, short *out, int len) { len -- ; scaled_value = in [len] * (8.0 * 0x10000000) ; - if (CPU_CLIPS_POSITIVE == 0 && scaled_value >= (1.0 * 0x7FFFFFFF)) + if (scaled_value >= (1.0 * 0x7FFFFFFF)) { out [len] = 32767 ; continue ; } ; - if (CPU_CLIPS_NEGATIVE == 0 && scaled_value <= (-8.0 * 0x10000000)) + if (scaled_value <= (-8.0 * 0x10000000)) { out [len] = -32768 ; continue ; } ; diff --git a/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp b/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp index 7ebedfe..d45c010 100644 --- a/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp +++ b/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp @@ -218,9 +218,9 @@ NuppelVideoPlayer::NuppelVideoPlayer() audioOutput(NULL), audio_main_device(QString::null), audio_passthru_device(QString::null), - audio_channels(2), audio_bits(-1), - audio_samplerate(44100), audio_stretchfactor(1.0f), - audio_codec(NULL), + audio_channels(2), audio_codec(0), + audio_bits(-1), audio_samplerate(44100), + audio_stretchfactor(1.0f), // Picture-in-Picture stuff pip_active(false), pip_visible(true), // Preview window support @@ -899,8 +899,8 @@ QString NuppelVideoPlayer::ReinitAudio(void) bool setVolume = gContext->GetNumSetting("MythControlsVolume", 1); audioOutput = AudioOutput::OpenAudio(audio_main_device, audio_passthru_device, - audio_bits, audio_channels, - audio_samplerate, + audio_bits, audio_channels, + audio_codec, audio_samplerate, AUDIOOUTPUT_VIDEO, setVolume, audio_passthru); if (!audioOutput) @@ -928,10 +928,11 @@ QString NuppelVideoPlayer::ReinitAudio(void) if (audioOutput) { - const AudioSettings settings( - audio_bits, audio_channels, audio_samplerate, - audio_passthru, audio_codec); + const AudioSettings settings(audio_bits, audio_channels, audio_codec, + audio_samplerate, audio_passthru); audioOutput->Reconfigure(settings); + if (audio_passthru) + audio_channels = 2; errMsg = audioOutput->GetError(); if (!errMsg.isEmpty()) audioOutput->SetStretchFactor(audio_stretchfactor); @@ -3862,20 +3863,16 @@ bool NuppelVideoPlayer::StartPlaying(bool openfile) return !IsErrored(); } -void NuppelVideoPlayer::SetAudioParams(int bps, int channels, +void NuppelVideoPlayer::SetAudioParams(int bps, int channels, int codec, int samplerate, bool passthru) { audio_bits = bps; audio_channels = channels; + audio_codec = codec; audio_samplerate = samplerate; audio_passthru = passthru; } -void NuppelVideoPlayer::SetAudioCodec(void *ac) -{ - audio_codec = ac; -} - void NuppelVideoPlayer::SetEffDsp(int dsprate) { if (audioOutput) @@ -5294,6 +5291,13 @@ void NuppelVideoPlayer::ToggleAdjustFill(AdjustFillMode adjustfillMode) } } +bool NuppelVideoPlayer::ToggleUpmix() +{ + if (audioOutput) + return audioOutput->ToggleUpmix(); + return false; +} + void NuppelVideoPlayer::Zoom(ZoomDirection direction) { if (videoOutput) diff --git a/mythtv/libs/libmythtv/NuppelVideoPlayer.h b/mythtv/libs/libmythtv/NuppelVideoPlayer.h index 2ceb683..06743d4 100644 --- a/mythtv/libs/libmythtv/NuppelVideoPlayer.h +++ b/mythtv/libs/libmythtv/NuppelVideoPlayer.h @@ -125,7 +125,8 @@ class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader void SetAudioStretchFactor(float factor) { audio_stretchfactor = factor; } void SetAudioOutput(AudioOutput *ao) { audioOutput = ao; } void SetAudioInfo(const QString &main, const QString &passthru, uint rate); - void SetAudioParams(int bits, int channels, int samplerate, bool passthru); + void SetAudioParams(int bits, int channels, int codec, int samplerate, + bool passthru); void SetEffDsp(int dsprate); uint AdjustVolume(int change); bool SetMuted(bool mute); @@ -179,6 +180,7 @@ class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader // Toggle Sets void ToggleAspectOverride(AspectOverrideMode aspectMode = kAspect_Toggle); void ToggleAdjustFill(AdjustFillMode adjustfillMode = kAdjustFill_Toggle); + bool ToggleUpmix(void); // Gets QSize GetVideoBufferSize(void) const { return video_dim; } @@ -711,10 +713,10 @@ class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader QString audio_main_device; QString audio_passthru_device; int audio_channels; + int audio_codec; int audio_bits; int audio_samplerate; float audio_stretchfactor; - void *audio_codec; bool audio_passthru; // Picture-in-Picture diff --git a/mythtv/libs/libmythtv/avformatdecoder.cpp b/mythtv/libs/libmythtv/avformatdecoder.cpp index bbc68e3..4530bc4 100644 --- a/mythtv/libs/libmythtv/avformatdecoder.cpp +++ b/mythtv/libs/libmythtv/avformatdecoder.cpp @@ -444,7 +444,7 @@ AvFormatDecoder::AvFormatDecoder(NuppelVideoPlayer *parent, audioSamples(NULL), allow_ac3_passthru(false), allow_dts_passthru(false), disable_passthru(false), max_channels(2), - dummy_frame(NULL), + last_ac3_channels(0), dummy_frame(NULL), // DVD lastdvdtitle(-1), decodeStillFrame(false), @@ -2019,7 +2019,7 @@ int AvFormatDecoder::ScanStreams(bool novideo) // waiting on audio. if (GetNVP()->HasAudioIn() && tracks[kTrackTypeAudio].empty()) { - GetNVP()->SetAudioParams(-1, -1, -1, false /* AC3/DTS pass-through */); + GetNVP()->SetAudioParams(-1, -1, CODEC_ID_NONE, -1, false /* AC3/DTS pass-through */); GetNVP()->ReinitAudio(); if (ringBuffer && ringBuffer->isDVD()) audioIn = AudioInfo(); @@ -3054,15 +3054,9 @@ int AvFormatDecoder::AutoSelectAudioTrack(void) { int idx = atracks[i].av_stream_index; AVCodecContext *codec_ctx = ic->streams[idx]->codec; - bool do_ac3_passthru = (allow_ac3_passthru && !transcoding && - !disable_passthru && - (codec_ctx->codec_id == CODEC_ID_AC3)); - bool do_dts_passthru = (allow_dts_passthru && !transcoding && - !disable_passthru && - (codec_ctx->codec_id == CODEC_ID_DTS)); AudioInfo item(codec_ctx->codec_id, codec_ctx->sample_rate, codec_ctx->channels, - do_ac3_passthru || do_dts_passthru); + DoPassThrough(codec_ctx)); VERBOSE(VB_AUDIO, LOC + " * " + item.toString()); } #endif @@ -3196,6 +3190,7 @@ static void extract_mono_channel(uint channel, AudioInfo *audioInfo, bool AvFormatDecoder::GetFrame(int onlyvideo) { AVPacket *pkt = NULL; + AC3HeaderInfo hdr; int len; unsigned char *ptr; int data_size = 0; @@ -3392,12 +3387,13 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) pts = 0; AVStream *curstream = ic->streams[pkt->stream_index]; + AVCodecContext *ctx = curstream->codec; if (pkt->dts != (int64_t)AV_NOPTS_VALUE) pts = (long long)(av_q2d(curstream->time_base) * pkt->dts * 1000); - if (ringBuffer->isDVD() && - curstream->codec->codec_type == CODEC_TYPE_VIDEO) + if (ringBuffer->isDVD() && + ctx->codec_type == CODEC_TYPE_VIDEO) { MpegPreProcessPkt(curstream, pkt); @@ -3419,7 +3415,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) if (!d->HasMPEG2Dec()) { - int current_width = curstream->codec->width; + int current_width = ctx->width; int video_width = GetNVP()->GetVideoSize().width(); if (dvd_xvmc_enabled && GetNVP() && GetNVP()->getVideoOutput()) { @@ -3460,7 +3456,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) } if (storevideoframes && - curstream->codec->codec_type == CODEC_TYPE_VIDEO) + ctx->codec_type == CODEC_TYPE_VIDEO) { av_dup_packet(pkt); storedPackets.append(pkt); @@ -3468,20 +3464,19 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) continue; } - if (len > 0 && curstream->codec->codec_type == CODEC_TYPE_VIDEO && + if (len > 0 && ctx->codec_type == CODEC_TYPE_VIDEO && pkt->stream_index == selectedVideoIndex) { - AVCodecContext *context = curstream->codec; - if (context->codec_id == CODEC_ID_MPEG1VIDEO || - context->codec_id == CODEC_ID_MPEG2VIDEO || - context->codec_id == CODEC_ID_MPEG2VIDEO_XVMC || - context->codec_id == CODEC_ID_MPEG2VIDEO_XVMC_VLD) + if (ctx->codec_id == CODEC_ID_MPEG1VIDEO || + ctx->codec_id == CODEC_ID_MPEG2VIDEO || + ctx->codec_id == CODEC_ID_MPEG2VIDEO_XVMC || + ctx->codec_id == CODEC_ID_MPEG2VIDEO_XVMC_VLD) { if (!ringBuffer->isDVD()) MpegPreProcessPkt(curstream, pkt); } - else if (context->codec_id == CODEC_ID_H264) + else if (ctx->codec_id == CODEC_ID_H264) { H264PreProcessPkt(curstream, pkt); } @@ -3526,8 +3521,8 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) } if (len > 0 && - curstream->codec->codec_type == CODEC_TYPE_DATA && - curstream->codec->codec_id == CODEC_ID_MPEG2VBI) + ctx->codec_type == CODEC_TYPE_DATA && + ctx->codec_id == CODEC_ID_MPEG2VBI) { ProcessVBIDataPacket(curstream, pkt); @@ -3536,8 +3531,8 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) } if (len > 0 && - curstream->codec->codec_type == CODEC_TYPE_DATA && - curstream->codec->codec_id == CODEC_ID_DVB_VBI) + ctx->codec_type == CODEC_TYPE_DATA && + ctx->codec_id == CODEC_ID_DVB_VBI) { ProcessDVBDataPacket(curstream, pkt); @@ -3547,8 +3542,8 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) #ifdef USING_MHEG if (len > 0 && - curstream->codec->codec_type == CODEC_TYPE_DATA && - curstream->codec->codec_id == CODEC_ID_DSMCC_B) + ctx->codec_type == CODEC_TYPE_DATA && + ctx->codec_id == CODEC_ID_DSMCC_B) { ProcessDSMCCPacket(curstream, pkt); @@ -3569,20 +3564,20 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) #endif // USING_MHEG // we don't care about other data streams - if (curstream->codec->codec_type == CODEC_TYPE_DATA) + if (ctx->codec_type == CODEC_TYPE_DATA) { av_free_packet(pkt); continue; } - if (!curstream->codec->codec) + if (!ctx->codec) { VERBOSE(VB_PLAYBACK, LOC + QString("No codec for stream index %1, type(%2) id(%3:%4)") .arg(pkt->stream_index) - .arg(codec_type_string(curstream->codec->codec_type)) - .arg(codec_id_string(curstream->codec->codec_id)) - .arg(curstream->codec->codec_id)); + .arg(codec_type_string(ctx->codec_type)) + .arg(codec_id_string(ctx->codec_id)) + .arg(ctx->codec_id)); av_free_packet(pkt); continue; } @@ -3591,7 +3586,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) have_err = false; avcodeclock.lock(); - int ctype = curstream->codec->codec_type; + int ctype = ctx->codec_type; int audIdx = selectedTrack[kTrackTypeAudio].av_stream_index; int audSubIdx = selectedTrack[kTrackTypeAudio].av_substream_index; int subIdx = selectedTrack[kTrackTypeSubtitle].av_stream_index; @@ -3616,51 +3611,57 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) // detect switches between stereo and dual languages bool wasDual = audSubIdx != -1; - bool isDual = curstream->codec->avcodec_dual_language; + bool isDual = ctx->avcodec_dual_language; if ((wasDual && !isDual) || (!wasDual && isDual)) { SetupAudioStreamSubIndexes(audIdx); reselectAudioTrack = true; } - bool do_ac3_passthru = - (allow_ac3_passthru && !transcoding && - (curstream->codec->codec_id == CODEC_ID_AC3)); - bool do_dts_passthru = - (allow_dts_passthru && !transcoding && - (curstream->codec->codec_id == CODEC_ID_DTS)); - bool using_passthru = do_ac3_passthru || do_dts_passthru; - // detect channels on streams that need // to be decoded before we can know this bool already_decoded = false; - if (!curstream->codec->channels) + if (!ctx->channels) { QMutexLocker locker(&avcodeclock); VERBOSE(VB_IMPORTANT, LOC + QString("Setting channels to %1") .arg(audioOut.channels)); - if (using_passthru) + if (DoPassThrough(ctx)) { // for passthru let it select the max number // of channels - curstream->codec->channels = 0; - curstream->codec->request_channels = 0; + ctx->channels = 0; + ctx->request_channels = 0; } else { - curstream->codec->channels = audioOut.channels; - curstream->codec->request_channels = + ctx->channels = audioOut.channels; + ctx->request_channels = audioOut.channels; } + data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; - ret = avcodec_decode_audio2(curstream->codec, - audioSamples, &data_size, - ptr, len); + ret = avcodec_decode_audio2(ctx, audioSamples, + &data_size, ptr, len); already_decoded = true; - reselectAudioTrack |= curstream->codec->channels; + reselectAudioTrack |= ctx->channels; + } + + if (ctx->codec_id == CODEC_ID_AC3) + { + GetBitContext gbc; + init_get_bits(&gbc, ptr, len * 8); + if (!ff_ac3_parse_header(&gbc, &hdr)) + { + if (hdr.channels != last_ac3_channels) + { + last_ac3_channels = ctx->channels = hdr.channels; + SetupAudioStream(); + } + } } if (reselectAudioTrack) @@ -3676,6 +3677,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) .av_stream_index; audSubIdx = selectedTrack[kTrackTypeAudio] .av_substream_index; + ctx = curstream->codec; } if ((onlyvideo > 0) || (pkt->stream_index != audIdx)) @@ -3707,14 +3709,12 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) if (audioOut.do_passthru) { data_size = pkt->size; - bool dts = CODEC_ID_DTS == curstream->codec->codec_id; + bool dts = CODEC_ID_DTS == ctx->codec_id; ret = encode_frame(dts, ptr, len, audioSamples, data_size); } else { - AVCodecContext *ctx = curstream->codec; - if ((ctx->channels == 0) || (ctx->channels > audioOut.channels)) { @@ -3723,10 +3723,9 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) if (!already_decoded) { - curstream->codec->request_channels = - audioOut.channels; data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; - ret = avcodec_decode_audio2(ctx, audioSamples, + ctx->request_channels = audioOut.channels; + ret = avcodec_decode_audio2(ctx, audioSamples, &data_size, ptr, len); } @@ -3742,6 +3741,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) audIdx = -1; AutoSelectAudioTrack(); data_size = 0; + ctx = curstream->codec; } } avcodeclock.unlock(); @@ -3759,8 +3759,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) // calc for next frame lastapts += (long long)((double)(data_size * 1000) / - (curstream->codec->channels * 2) / - curstream->codec->sample_rate); + (ctx->channels * 2) / ctx->sample_rate); VERBOSE(VB_PLAYBACK+VB_TIMESTAMP, LOC + QString("audio timecode %1 %2 %3 %4") @@ -3828,7 +3827,6 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) continue; } - AVCodecContext *context = curstream->codec; AVFrame mpa_pic; bzero(&mpa_pic, sizeof(AVFrame)); @@ -3843,24 +3841,24 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) // HACK while (!gotpicture && count < 5) { - ret = d->DecodeMPEG2Video(context, &mpa_pic, + ret = d->DecodeMPEG2Video(ctx, &mpa_pic, &gotpicture, ptr, len); count++; } } else { - ret = d->DecodeMPEG2Video(context, &mpa_pic, + ret = d->DecodeMPEG2Video(ctx, &mpa_pic, &gotpicture, ptr, len); } } else { - ret = avcodec_decode_video(context, &mpa_pic, + ret = avcodec_decode_video(ctx, &mpa_pic, &gotpicture, ptr, len); // Reparse it to not drop the DVD still frame if (decodeStillFrame) - ret = avcodec_decode_video(context, &mpa_pic, + ret = avcodec_decode_video(ctx, &mpa_pic, &gotpicture, ptr, len); } avcodeclock.unlock(); @@ -3928,9 +3926,9 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) myth_sws_img_convert( &tmppicture, PIX_FMT_YUV420P, (AVPicture *)&mpa_pic, - context->pix_fmt, - context->width, - context->height); + ctx->pix_fmt, + ctx->width, + ctx->height); if (xf) { @@ -3953,10 +3951,10 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) (temppts + 10000 > lastvpts || temppts < 0)) { temppts = lastvpts; - temppts += (long long)(1000 * av_q2d(context->time_base)); + temppts += (long long)(1000 * av_q2d(ctx->time_base)); // MPEG2 frames can be repeated, update pts accordingly temppts += (long long)(mpa_pic.repeat_pict * 500 - * av_q2d(curstream->codec->time_base)); + * av_q2d(ctx->time_base)); } VERBOSE(VB_PLAYBACK+VB_TIMESTAMP, LOC + @@ -3992,7 +3990,7 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) picframe->frameNumber = framesPlayed; GetNVP()->ReleaseNextVideoFrame(picframe, temppts); if (d->HasMPEG2Dec() && mpa_pic.data[3]) - context->release_buffer(context, &mpa_pic); + ctx->release_buffer(ctx, &mpa_pic); decoded_video_frame = picframe; gotvideo = 1; @@ -4048,11 +4046,10 @@ bool AvFormatDecoder::GetFrame(int onlyvideo) } default: { - AVCodecContext *enc = curstream->codec; VERBOSE(VB_IMPORTANT, LOC_ERR + QString("Decoding - id(%1) type(%2)") - .arg(codec_id_string(enc->codec_id)) - .arg(codec_type_string(enc->codec_type))); + .arg(codec_id_string(ctx->codec_id)) + .arg(codec_type_string(ctx->codec_type))); have_err = true; break; } @@ -4201,6 +4198,24 @@ void AvFormatDecoder::SetDisablePassThrough(bool disable) } } +bool AvFormatDecoder::DoPassThrough(const AVCodecContext *ctx) +{ + bool passthru = false; + + if (ctx->codec_id == CODEC_ID_AC3) + passthru = allow_ac3_passthru && + ctx->channels >= (int)max_channels; + else if (ctx->codec_id == CODEC_ID_DTS) + passthru = allow_dts_passthru; + + passthru &= !transcoding && !disable_passthru; + // Don't know any cards that support spdif clocked at < 44100 + // Some US cable transmissions have 2ch 32k AC-3 streams + passthru &= ctx->sample_rate >= 44100; + + return passthru; +} + /** \fn AvFormatDecoder::SetupAudioStream(void) * \brief Reinitializes audio if it needs to be reinitialized. * @@ -4214,7 +4229,6 @@ bool AvFormatDecoder::SetupAudioStream(void) AVStream *curstream = NULL; AVCodecContext *codec_ctx = NULL; AudioInfo old_in = audioIn; - AudioInfo old_out = audioOut; bool using_passthru = false; if ((currentTrack[kTrackTypeAudio] >= 0) && @@ -4228,14 +4242,9 @@ bool AvFormatDecoder::SetupAudioStream(void) codec_ctx = curstream->codec; if (codec_ctx) { - bool do_ac3_passthru = (allow_ac3_passthru && !transcoding && - (codec_ctx->codec_id == CODEC_ID_AC3)); - bool do_dts_passthru = (allow_dts_passthru && !transcoding && - (codec_ctx->codec_id == CODEC_ID_DTS)); - using_passthru = do_ac3_passthru || do_dts_passthru; - info = AudioInfo(codec_ctx->codec_id, - codec_ctx->sample_rate, codec_ctx->channels, - using_passthru && !disable_passthru); + using_passthru = DoPassThrough(codec_ctx); + info = AudioInfo(codec_ctx->codec_id, codec_ctx->sample_rate, + codec_ctx->channels, using_passthru); } } @@ -4252,41 +4261,25 @@ bool AvFormatDecoder::SetupAudioStream(void) QString("audio track #%1").arg(currentTrack[kTrackTypeAudio]+1)); audioOut = audioIn = info; - AudioInfo tmpAudioOut = audioOut; - // A passthru stream looks like a 48KHz 2ch (@ 16bit) to the sound card - if (using_passthru && !disable_passthru) + if (!using_passthru && audioOut.channels > (int)max_channels) { - tmpAudioOut.channels = 2; - tmpAudioOut.sample_rate = 48000; - tmpAudioOut.sample_size = 4; - } - - if (audioOut.channels > (int) max_channels) - { - audioOut.channels = (int) max_channels; + audioOut.channels = (int)max_channels; audioOut.sample_size = audioOut.channels * 2; - codec_ctx->channels = audioOut.channels; + codec_ctx->channels = audioOut.channels; } - if (!using_passthru) - tmpAudioOut = audioOut; - VERBOSE(VB_AUDIO, LOC + "Audio format changed " + - QString("%1%2\n\t\t\tfrom %3 ; %4\n\t\t\tto %5 ; %6") - .arg((using_passthru) ? "digital passthrough " : "") - .arg((using_passthru) ? tmpAudioOut.toString() : QString("")) - .arg(old_in.toString()).arg(old_out.toString()) - .arg(audioIn.toString()).arg(audioOut.toString())); + QString("\n\t\t\tfrom %1 to %2") + .arg(old_in.toString()).arg(audioOut.toString())); - if (tmpAudioOut.sample_rate > 0) - GetNVP()->SetEffDsp(tmpAudioOut.sample_rate * 100); + if (audioOut.sample_rate > 0) + GetNVP()->SetEffDsp(audioOut.sample_rate * 100); - GetNVP()->SetAudioParams(tmpAudioOut.bps(), tmpAudioOut.channels, - tmpAudioOut.sample_rate, audioIn.do_passthru); + GetNVP()->SetAudioParams(audioOut.bps(), audioOut.channels, + audioOut.codec_id, audioOut.sample_rate, + audioOut.do_passthru); - // allow the audio stuff to reencode - GetNVP()->SetAudioCodec((using_passthru) ? codec_ctx : NULL); GetNVP()->ReinitAudio(); return true; diff --git a/mythtv/libs/libmythtv/avformatdecoder.h b/mythtv/libs/libmythtv/avformatdecoder.h index 9ead273..ea1fb23 100644 --- a/mythtv/libs/libmythtv/avformatdecoder.h +++ b/mythtv/libs/libmythtv/avformatdecoder.h @@ -198,6 +198,7 @@ class AvFormatDecoder : public DecoderBase void SeekReset(long long, uint skipFrames, bool doFlush, bool discardFrames); + bool DoPassThrough(const AVCodecContext *ctx); bool SetupAudioStream(void); void SetupAudioStreamSubIndexes(int streamIndex); void RemoveAudioStreams(); @@ -270,6 +271,7 @@ class AvFormatDecoder : public DecoderBase bool allow_dts_passthru; bool disable_passthru; uint max_channels; + uint last_ac3_channels; VideoFrame *dummy_frame; diff --git a/mythtv/libs/libmythtv/nuppeldecoder.cpp b/mythtv/libs/libmythtv/nuppeldecoder.cpp index 88d2fbb..010a458 100644 --- a/mythtv/libs/libmythtv/nuppeldecoder.cpp +++ b/mythtv/libs/libmythtv/nuppeldecoder.cpp @@ -495,6 +495,7 @@ int NuppelDecoder::OpenFile(RingBuffer *rbuffer, bool novideo, #endif GetNVP()->SetAudioParams(extradata.audio_bits_per_sample, extradata.audio_channels, + CODEC_ID_NONE, extradata.audio_sample_rate, false /* AC3/DTS pass through */); GetNVP()->ReinitAudio(); diff --git a/mythtv/libs/libmythtv/tv_play.cpp b/mythtv/libs/libmythtv/tv_play.cpp index babe363..99e62a0 100644 --- a/mythtv/libs/libmythtv/tv_play.cpp +++ b/mythtv/libs/libmythtv/tv_play.cpp @@ -491,6 +491,7 @@ void TV::InitKeys(void) REG_KEY("TV Playback", "VOLUMEDOWN", "Volume down", "[,{,F10,Volume Down"); REG_KEY("TV Playback", "VOLUMEUP", "Volume up", "],},F11,Volume Up"); REG_KEY("TV Playback", "MUTE", "Mute", "|,\\,F9,Volume Mute"); + REG_KEY("TV Playback", "TOGGLEUPMIX", "Toggle upmixer", "Ctrl+U"); REG_KEY("TV Playback", "TOGGLEPIPMODE", "Toggle Picture-in-Picture view", "V"); REG_KEY("TV Playback", "TOGGLEPBPMODE", "Toggle Picture-by-Picture view", @@ -630,7 +631,7 @@ void TV::InitKeys(void) Teletext F2,F3,F4,F5,F6,F7,F8 ITV F2,F3,F4,F5,F6,F7,F12 - Playback: Ctrl-B,Ctrl-G,Ctrl-Y + Playback: Ctrl-B,Ctrl-G,Ctrl-Y,Ctrl-U */ } @@ -4345,6 +4346,8 @@ bool TV::ToggleHandleAction(PlayerContext *ctx, DoTogglePictureAttribute(ctx, kAdjustingPicture_Playback); else if (has_action("TOGGLESTRETCH", actions)) ToggleTimeStretch(ctx); + else if (has_action("TOGGLEUPMIX", actions)) + ToggleUpmix(ctx); else if (has_action("TOGGLESLEEP", actions)) ToggleSleepTimer(ctx); else if (has_action("TOGGLERECORD", actions) && islivetv) @@ -8010,6 +8013,20 @@ void TV::ChangeTimeStretch(PlayerContext *ctx, int dir, bool allowEdit) SetSpeedChangeTimer(0, __LINE__); } +void TV::ToggleUpmix(PlayerContext *ctx) +{ + if (!ctx->nvp || !ctx->nvp->HasAudioOut()) + return; + QString text; + if (ctx->nvp->ToggleUpmix()) + text = tr("Upmixer On"); + else + text = tr("Upmixer Off"); + + if (ctx->nvp->GetOSD() && !browsemode) + ctx->nvp->GetOSD()->SetSettingsText(text, 5); +} + // dir in 10ms jumps void TV::ChangeAudioSync(PlayerContext *ctx, int dir, bool allowEdit) { @@ -9642,6 +9659,8 @@ void TV::TreeMenuSelected(OSDListTreeItemSelectedEvent *e) SetManualZoom(actx, true, tr("Zoom Mode ON")); else if (action == "TOGGLESTRETCH") ToggleTimeStretch(actx); + else if (action == "TOGGLEUPMIX") + ToggleUpmix(actx); else if (action.left(13) == "ADJUSTSTRETCH") { bool floatRead; @@ -9995,6 +10014,8 @@ void TV::FillOSDTreeMenu( if (category == "AUDIOSYNC") new OSDGenericTree(treeMenu, tr("Adjust Audio Sync"), "TOGGLEAUDIOSYNC"); + else if (category == "TOGGLEUPMIX") + new OSDGenericTree(treeMenu, tr("Toggle Upmixer"), "TOGGLEUPMIX"); else if (category == "TIMESTRETCH") FillMenuTimeStretch(ctx, treeMenu); else if (category == "VIDEOSCAN") diff --git a/mythtv/libs/libmythtv/tv_play.h b/mythtv/libs/libmythtv/tv_play.h index b361a7e..f9b93bf 100644 --- a/mythtv/libs/libmythtv/tv_play.h +++ b/mythtv/libs/libmythtv/tv_play.h @@ -419,6 +419,7 @@ class MPUBLIC TV : public QThread ARBSEEK_FORWARD, ARBSEEK_END }; + void DoArbSeek(PlayerContext*, ArbSeekWhence whence); void NormalSpeed(PlayerContext*); void ChangeSpeed(PlayerContext*, int direction); @@ -427,6 +428,7 @@ class MPUBLIC TV : public QThread bool TimeStretchHandleAction(PlayerContext*, const QStringList &actions); + void ToggleUpmix(PlayerContext*); void ChangeAudioSync(PlayerContext*, int dir, bool allowEdit = true); bool AudioSyncHandleAction(PlayerContext*, const QStringList &actions); diff --git a/mythtv/libs/libmythtv/tvosdmenuentry.cpp b/mythtv/libs/libmythtv/tvosdmenuentry.cpp index 994dcb2..ef714ea 100644 --- a/mythtv/libs/libmythtv/tvosdmenuentry.cpp +++ b/mythtv/libs/libmythtv/tvosdmenuentry.cpp @@ -232,6 +232,8 @@ void TVOSDMenuEntryList::InitDefaultEntries(void) curMenuEntries.append( new TVOSDMenuEntry("AUDIOSYNC", 1, 1, 1, 1 , "Audio Sync")); curMenuEntries.append( + new TVOSDMenuEntry("TOGGLEUPMIX", 1, 1, 1, 1, "Toggle Upmixer")); + curMenuEntries.append( new TVOSDMenuEntry("TIMESTRETCH", 1, 1, 1, 1, "Time Stretch")); curMenuEntries.append( new TVOSDMenuEntry("VIDEOSCAN", 1, 1, 1, 1, "Video Scan")); diff --git a/mythtv/programs/mythfrontend/globalsettings.cpp b/mythtv/programs/mythfrontend/globalsettings.cpp index d61cf26..c5d0406 100644 --- a/mythtv/programs/mythfrontend/globalsettings.cpp +++ b/mythtv/programs/mythfrontend/globalsettings.cpp @@ -119,6 +119,27 @@ static HostComboBox *AudioUpmixType() return gc; } +static HostComboBox *SRCQuality() +{ + HostComboBox *gc = new HostComboBox("SRCQuality", false); + gc->setLabel(QObject::tr("Sample Rate Conversion")); + gc->addSelection(QObject::tr("Best"), "3", true); // default + gc->addSelection(QObject::tr("Medium"), "2"); + gc->addSelection(QObject::tr("Fastest"), "1"); + gc->setHelpText( + QObject::tr( + "Set the quality of audio sample rate conversion. " + "This only affects non 48000Hz PCM audio. " + "All three options offer a worst-case SNR of 97dB. " + "'Best' at a bandwidth of 97%. " + "'Medium' at a bandwidth of 90%. " + "'Fastest' at a bandwidth of 80%. " + ) + ); + return gc; +} + + static HostComboBox *PassThroughOutputDevice() { HostComboBox *gc = new HostComboBox("PassThruOutputDevice", true); @@ -3446,6 +3467,7 @@ class AudioSystemSettingsGroup : public VerticalConfigurationGroup addChild(MaxAudioChannels()); addChild(AudioUpmixType()); + addChild(SRCQuality()); // General boolean settings addChild(AC3PassThrough()); diff --git a/mythtv/programs/mythtranscode/transcode.cpp b/mythtv/programs/mythtranscode/transcode.cpp index a69ceb1..9958473 100644 --- a/mythtv/programs/mythtranscode/transcode.cpp +++ b/mythtv/programs/mythtranscode/transcode.cpp @@ -49,7 +49,7 @@ class AudioReencodeBuffer : public AudioOutput AudioReencodeBuffer(int audio_bits, int audio_channels) { Reset(); - const AudioSettings settings(audio_bits, audio_channels, 0, false); + const AudioSettings settings(audio_bits, audio_channels, 0, 0, false); Reconfigure(settings); bufsize = 512000; audiobuffer = new unsigned char[bufsize]; @@ -222,6 +222,11 @@ class AudioReencodeBuffer : public AudioOutput // Do nothing return kMuteOff; } + virtual bool ToggleUpmix(void) + { + // Do nothing + return false; + } // These are pure virtual in AudioOutput, but we don't need them here virtual void bufferOutputData(bool){ return; }