Ticket #5900: audioencoding-trunk-7.3-mt.patch
File audioencoding-trunk-7.3-mt.patch, 100.7 KB (added by , 15 years ago) |
---|
-
mythtv/libs/libmyth/audiooutput.cpp
diff --git a/mythtv/libs/libmyth/audiooutput.cpp b/mythtv/libs/libmyth/audiooutput.cpp index 17f7c64..f97fe44 100644
a b using namespace std; 33 33 AudioOutput *AudioOutput::OpenAudio( 34 34 const QString &main_device, 35 35 const QString &passthru_device, 36 int audio_bits, int audio_channels, int audio_samplerate, 36 int audio_bits, int audio_channels, 37 int audio_codec, int audio_samplerate, 37 38 AudioOutputSource source, 38 39 bool set_initial_vol, bool audio_passthru) 39 40 { 40 41 AudioSettings settings( 41 42 main_device, passthru_device, audio_bits, 42 audio_channels, audio_ samplerate, source,43 audio_channels, audio_codec, audio_samplerate, source, 43 44 set_initial_vol, audio_passthru); 44 45 45 46 settings.FixPassThrough(); -
mythtv/libs/libmyth/audiooutput.h
diff --git a/mythtv/libs/libmyth/audiooutput.h b/mythtv/libs/libmyth/audiooutput.h index 943bd77..7c55ac0 100644
a b class MPUBLIC AudioOutput : public VolumeBase, public OutputListeners 15 15 static AudioOutput *OpenAudio( 16 16 const QString &audiodevice, 17 17 const QString &passthrudevice, 18 int audio_bits, int audio_channels, int audio_samplerate, 18 int audio_bits, int audio_channels, 19 int audio_codec, int audio_samplerate, 19 20 AudioOutputSource source, 20 21 bool set_initial_vol, bool audio_passthru); 21 22 … … class MPUBLIC AudioOutput : public VolumeBase, public OutputListeners 68 69 69 70 virtual void bufferOutputData(bool y) = 0; 70 71 virtual int readOutputData(unsigned char *read_buffer, int max_length) = 0; 72 virtual bool ToggleUpmix(void) = 0; 71 73 72 74 protected: 73 75 void Error(const QString &msg); -
mythtv/libs/libmyth/audiooutputalsa.cpp
diff --git a/mythtv/libs/libmyth/audiooutputalsa.cpp b/mythtv/libs/libmyth/audiooutputalsa.cpp index 5bd6445..02f6668 100644
a b AudioOutputALSA::AudioOutputALSA(const AudioSettings &settings) : 32 32 AudioOutputALSA::~AudioOutputALSA() 33 33 { 34 34 KillAudio(); 35 SetIECStatus(true); 36 } 37 38 void AudioOutputALSA::SetIECStatus(bool audio) { 39 40 snd_ctl_t *ctl; 41 const char *spdif_str = SND_CTL_NAME_IEC958("", PLAYBACK, DEFAULT); 42 int spdif_index = -1; 43 snd_ctl_elem_list_t *clist; 44 snd_ctl_elem_id_t *cid; 45 snd_ctl_elem_value_t *cval; 46 snd_aes_iec958_t iec958; 47 int cidx, controls; 48 49 VERBOSE(VB_AUDIO, QString("Setting IEC958 status: %1") 50 .arg(audio ? "audio" : "non-audio")); 51 52 snd_ctl_open(&ctl, "default", 0); 53 snd_ctl_elem_list_alloca(&clist); 54 snd_ctl_elem_list(ctl, clist); 55 snd_ctl_elem_list_alloc_space(clist, snd_ctl_elem_list_get_count(clist)); 56 snd_ctl_elem_list(ctl, clist); 57 controls = snd_ctl_elem_list_get_used(clist); 58 for (cidx = 0; cidx < controls; cidx++) { 59 if (!strcmp(snd_ctl_elem_list_get_name(clist, cidx), spdif_str)) 60 if (spdif_index < 0 || 61 snd_ctl_elem_list_get_index(clist, cidx) == (uint)spdif_index) 62 break; 63 } 64 65 if (cidx >= controls) 66 return; 67 68 snd_ctl_elem_id_alloca(&cid); 69 snd_ctl_elem_list_get_id(clist, cidx, cid); 70 snd_ctl_elem_value_alloca(&cval); 71 snd_ctl_elem_value_set_id(cval, cid); 72 snd_ctl_elem_read(ctl,cval); 73 snd_ctl_elem_value_get_iec958(cval, &iec958); 74 75 if (!audio) 76 iec958.status[0] |= IEC958_AES0_NONAUDIO; 77 else 78 iec958.status[0] &= ~IEC958_AES0_NONAUDIO; 79 80 snd_ctl_elem_value_set_iec958(cval, &iec958); 81 snd_ctl_elem_write(ctl, cval); 82 83 } 84 85 vector<int> AudioOutputALSA::GetSupportedRates() 86 { 87 snd_pcm_hw_params_t *params; 88 int err; 89 const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; 90 vector<int> rates(srates, srates + sizeof(srates) / sizeof(int) ); 91 QString real_device; 92 93 if (audio_passthru || audio_enc) 94 real_device = audio_passthru_device; 95 else 96 real_device = audio_main_device; 97 98 if((err = snd_pcm_open(&pcm_handle, real_device.toAscii(), 99 SND_PCM_STREAM_PLAYBACK, 100 SND_PCM_NONBLOCK|SND_PCM_NO_AUTO_RESAMPLE)) < 0) 101 { 102 Error(QString("snd_pcm_open(%1): %2") 103 .arg(real_device).arg(snd_strerror(err))); 104 105 if (pcm_handle) 106 { 107 snd_pcm_close(pcm_handle); 108 pcm_handle = NULL; 109 } 110 rates.clear(); 111 return rates; 112 } 113 114 snd_pcm_hw_params_alloca(¶ms); 115 116 if ((err = snd_pcm_hw_params_any(pcm_handle, params)) < 0) 117 { 118 Error(QString("Broken configuration for playback; no configurations" 119 " available: %1").arg(snd_strerror(err))); 120 snd_pcm_close(pcm_handle); 121 pcm_handle = NULL; 122 rates.clear(); 123 return rates; 124 } 125 126 vector<int>::iterator it; 127 128 for (it = rates.begin(); it < rates.end(); it++) 129 if(snd_pcm_hw_params_test_rate(pcm_handle, params, *it, 0) < 0) 130 rates.erase(it--); 131 132 snd_pcm_close(pcm_handle); 133 pcm_handle = NULL; 134 135 return rates; 35 136 } 36 137 37 138 bool AudioOutputALSA::OpenDevice() … … bool AudioOutputALSA::OpenDevice() 39 140 snd_pcm_format_t format; 40 141 unsigned int buffer_time, period_time; 41 142 int err; 143 QString real_device; 42 144 43 145 if (pcm_handle != NULL) 44 146 CloseDevice(); 45 147 46 148 pcm_handle = NULL; 47 149 numbadioctls = 0; 48 49 QString real_device = (audio_passthru) ? 50 audio_passthru_device : audio_main_device; 150 151 if (audio_passthru || audio_enc) 152 { 153 real_device = audio_passthru_device; 154 SetIECStatus(false); 155 } 156 else 157 { 158 real_device = audio_main_device; 159 SetIECStatus(true); 160 } 51 161 52 162 VERBOSE(VB_GENERAL, QString("Opening ALSA audio device '%1'.") 53 163 .arg(real_device)); … … void AudioOutputALSA::CloseDevice() 146 256 } 147 257 } 148 258 259 void AudioOutputALSA::ReorderSmpteToAlsa6ch(unsigned char *buf, int size) 260 { 261 if (audio_bits == 8) 262 _ReorderSmpteToAlsa6ch(buf, size); 263 else if (audio_bits == 16) 264 _ReorderSmpteToAlsa6ch((short *)buf, size / sizeof(short)); 265 266 } 267 268 template <class AudioDataType> 269 void AudioOutputALSA::_ReorderSmpteToAlsa6ch(AudioDataType *buf, int size) 270 { 271 AudioDataType tmpC, tmpLFE; 272 273 for (int i = 0; i < size; i+= 6) { 274 tmpC = buf[i+2]; 275 tmpLFE = buf[i+3]; 276 buf[i+2] = buf[i+4]; 277 buf[i+3] = buf[i+5]; 278 buf[i+4] = tmpC; 279 buf[i+5] = tmpLFE; 280 } 149 281 282 } 150 283 void AudioOutputALSA::WriteAudio(unsigned char *aubuf, int size) 151 284 { 152 285 unsigned char *tmpbuf; … … void AudioOutputALSA::WriteAudio(unsigned char *aubuf, int size) 158 291 VERBOSE(VB_IMPORTANT, QString("WriteAudio() called with pcm_handle == NULL!")); 159 292 return; 160 293 } 294 295 if (!(audio_passthru || audio_enc) && audio_channels == 6) 296 ReorderSmpteToAlsa6ch(aubuf, size); 297 161 298 162 299 tmpbuf = aubuf; 163 300 -
mythtv/libs/libmyth/audiooutputalsa.h
diff --git a/mythtv/libs/libmyth/audiooutputalsa.h b/mythtv/libs/libmyth/audiooutputalsa.h index a156edd..f8cc67a 100644
a b class AudioOutputALSA : public AudioOutputBase 65 65 virtual void WriteAudio(unsigned char *aubuf, int size); 66 66 virtual int GetSpaceOnSoundcard(void) const; 67 67 virtual int GetBufferedOnSoundcard(void) const; 68 virtual vector<int> GetSupportedRates(void); 68 69 69 70 private: 71 void SetIECStatus(bool audio); 70 72 inline int SetParameters(snd_pcm_t *handle, 71 73 snd_pcm_format_t format, unsigned int channels, 72 74 unsigned int rate, unsigned int buffer_time, 73 75 unsigned int period_time); 74 76 75 77 void ReorderSmpteToAlsa6ch(unsigned char *buf, int size); 78 template <class AudioDataType> 79 void _ReorderSmpteToAlsa6ch(AudioDataType *buf, int size); 76 80 // Volume related 77 81 void SetCurrentVolume(QString control, int channel, int volume); 78 82 void OpenMixer(bool setstartingvolume); -
mythtv/libs/libmyth/audiooutputarts.h
diff --git a/mythtv/libs/libmyth/audiooutputarts.h b/mythtv/libs/libmyth/audiooutputarts.h index f985ef8..35ec39f 100644
a b class AudioOutputARTS : public AudioOutputBase 27 27 virtual void WriteAudio(unsigned char *aubuf, int size); 28 28 virtual int GetSpaceOnSoundcard(void) const; 29 29 virtual int GetBufferedOnSoundcard(void) const; 30 31 30 virtual vector<int> GetSupportedRates(void) 31 { vector<int> rates; return rates; } 32 32 33 33 private: 34 34 arts_stream_t pcm_handle; -
mythtv/libs/libmyth/audiooutputbase.cpp
diff --git a/mythtv/libs/libmyth/audiooutputbase.cpp b/mythtv/libs/libmyth/audiooutputbase.cpp index ef2f3d5..7be9832 100644
a b 21 21 AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : 22 22 // protected 23 23 effdsp(0), effdspstretched(0), 24 audio_channels(-1), audio_ bytes_per_sample(0),25 audio_b its(-1), audio_samplerate(-1),26 audio_ buffer_unused(0),24 audio_channels(-1), audio_codec(CODEC_ID_NONE), 25 audio_bytes_per_sample(0), audio_bits(-1), 26 audio_samplerate(-1), audio_buffer_unused(0), 27 27 fragment_size(0), soundcard_buffer_size(0), 28 28 29 29 audio_main_device(settings.GetMainDevice()), 30 30 audio_passthru_device(settings.GetPassthruDevice()), 31 audio_passthru(false), audio_stretchfactor(1.0f), 31 audio_passthru(false), audio_enc(false), 32 audio_reenc(false), audio_stretchfactor(1.0f), 32 33 33 audio_codec(NULL),34 34 source(settings.source), killaudio(false), 35 35 36 36 pauseaudio(false), audio_actually_paused(false), … … AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : 48 48 encoder(NULL), 49 49 upmixer(NULL), 50 50 source_audio_channels(-1), 51 source_audio_samplerate(0), 51 52 source_audio_bytes_per_sample(0), 52 53 needs_upmix(false), 53 54 surround_mode(FreeSurround::SurroundModePassive), 55 old_audio_stretchfactor(1.0), 54 56 55 57 blocking(false), 56 58 … … AudioOutputBase::AudioOutputBase(const AudioSettings &settings) : 79 81 memset(&audiotime_updated, 0, sizeof(audiotime_updated)); 80 82 memset(audiobuffer, 0, sizeof(char) * kAudioRingBufferSize); 81 83 configured_audio_channels = gContext->GetNumSetting("MaxChannels", 2); 84 orig_config_channels = configured_audio_channels; 85 allow_ac3_passthru = gContext->GetNumSetting("AC3PassThru", false); 86 src_quality = gContext->GetNumSetting("SRCQuality", 3); 82 87 83 88 // You need to call Reconfigure from your concrete class. 84 89 // Reconfigure(laudio_bits, laudio_channels, … … void AudioOutputBase::SetStretchFactorLocked(float laudio_stretchfactor) 124 129 VERBOSE(VB_GENERAL, LOC + QString("Using time stretch %1") 125 130 .arg(audio_stretchfactor)); 126 131 pSoundStretch = new soundtouch::SoundTouch(); 127 if (audio_codec) 128 { 129 if (!encoder) 130 { 131 VERBOSE(VB_AUDIO, LOC + 132 QString("Creating Encoder for codec %1 origfs %2") 133 .arg(audio_codec->codec_id) 134 .arg(audio_codec->frame_size)); 135 136 encoder = new AudioOutputDigitalEncoder(); 137 if (!encoder->Init(audio_codec->codec_id, 138 audio_codec->bit_rate, 139 audio_codec->sample_rate, 140 audio_codec->channels 141 )) 142 { 143 // eeks 144 delete encoder; 145 encoder = NULL; 146 VERBOSE(VB_AUDIO, LOC + 147 QString("Failed to Create Encoder")); 148 } 149 } 150 } 151 if (audio_codec && encoder) 152 { 153 pSoundStretch->setSampleRate(audio_codec->sample_rate); 154 pSoundStretch->setChannels(audio_codec->channels); 155 } 156 else 157 { 158 pSoundStretch->setSampleRate(audio_samplerate); 159 pSoundStretch->setChannels(audio_channels); 160 } 132 pSoundStretch->setSampleRate(audio_samplerate); 133 pSoundStretch->setChannels(upmixer ? 134 configured_audio_channels : source_audio_channels); 161 135 162 136 pSoundStretch->setTempo(audio_stretchfactor); 163 137 pSoundStretch->setSetting(SETTING_SEQUENCE_MS, 35); … … void AudioOutputBase::SetStretchFactorLocked(float laudio_stretchfactor) 165 139 // dont need these with only tempo change 166 140 //pSoundStretch->setPitch(1.0); 167 141 //pSoundStretch->setRate(1.0); 168 169 142 //pSoundStretch->setSetting(SETTING_USE_QUICKSEEK, true); 170 143 //pSoundStretch->setSetting(SETTING_USE_AA_FILTER, false); 171 144 } … … float AudioOutputBase::GetStretchFactor(void) const 183 156 return audio_stretchfactor; 184 157 } 185 158 159 bool AudioOutputBase::ToggleUpmix(void) 160 { 161 if (orig_config_channels == 2 || source_audio_channels > 2 || 162 audio_passthru) 163 return false; 164 if (configured_audio_channels == 6) 165 configured_audio_channels = 2; 166 else 167 configured_audio_channels = 6; 168 169 const AudioSettings settings(audio_bits, source_audio_channels, 170 audio_codec, source_audio_samplerate, 171 audio_passthru); 172 Reconfigure(settings); 173 return (configured_audio_channels == 6); 174 } 175 176 186 177 void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 187 178 { 188 179 AudioSettings settings = orig_settings; 189 180 190 int codec_id = CODEC_ID_NONE;191 int lcodec_id = CODEC_ID_NONE;192 int lcchannels = 0;193 int cchannels = 0;194 181 int lsource_audio_channels = settings.channels; 195 182 bool lneeds_upmix = false; 183 bool laudio_reenc = false; 196 184 197 if (settings.codec) 185 // Are we reencoding a (previously) timestretched bitstream? 186 if ((settings.codec == CODEC_ID_AC3 || settings.codec == CODEC_ID_DTS) && 187 !settings.use_passthru && allow_ac3_passthru) 198 188 { 199 lcodec_id = ((AVCodecContext*)settings.codec)->codec_id; 200 settings.bits = 16; 201 settings.channels = 2; 202 lsource_audio_channels = settings.channels; 203 settings.samplerate = 48000; 204 lcchannels = ((AVCodecContext*)settings.codec)->channels; 189 laudio_reenc = true; 190 VERBOSE(VB_AUDIO, LOC + "Reencoding decoded AC3/DTS to AC3"); 205 191 } 206 192 207 if (audio_codec) 208 { 209 codec_id = audio_codec->codec_id; 210 cchannels = ((AVCodecContext*)audio_codec)->channels; 211 } 212 213 if ((configured_audio_channels == 6) && 214 !(settings.codec || audio_codec)) 193 // Enough channels? Upmix if not 194 if (settings.channels < configured_audio_channels && 195 !settings.use_passthru) 215 196 { 216 197 settings.channels = configured_audio_channels; 217 198 lneeds_upmix = true; … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 224 205 settings.samplerate == audio_samplerate && !need_resampler && 225 206 settings.use_passthru == audio_passthru && 226 207 lneeds_upmix == needs_upmix && 227 l codec_id == codec_id && lcchannels == cchannels);208 laudio_reenc == audio_reenc); 228 209 bool upmix_deps = 229 210 (lsource_audio_channels == source_audio_channels); 230 211 if (general_deps && upmix_deps) … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 251 232 waud = raud = 0; 252 233 audio_actually_paused = false; 253 234 254 bool redo_stretch = (pSoundStretch && audio_channels != settings.channels);255 235 audio_channels = settings.channels; 256 236 source_audio_channels = lsource_audio_channels; 257 237 audio_bits = settings.bits; 258 audio_samplerate = settings.samplerate; 259 audio_codec = (AVCodecContext*)settings.codec; 238 source_audio_samplerate = audio_samplerate = settings.samplerate; 239 audio_reenc = laudio_reenc; 240 audio_codec = settings.codec; 260 241 audio_passthru = settings.use_passthru; 261 242 needs_upmix = lneeds_upmix; 262 243 … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 265 246 Error("AudioOutput only supports 8 or 16bit audio."); 266 247 return; 267 248 } 268 audio_bytes_per_sample = audio_channels * audio_bits / 8; 269 source_audio_bytes_per_sample = source_audio_channels * audio_bits / 8; 249 250 VERBOSE(VB_AUDIO, LOC + QString("Original audio codec was %1") 251 .arg(codec_id_string((CodecID)audio_codec))); 270 252 271 253 need_resampler = false; 272 254 killaudio = false; … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 275 257 internal_vol = gContext->GetNumSetting("MythControlsVolume", 0); 276 258 277 259 numlowbuffer = 0; 260 261 // Find out what sample rates we can output (if output layer supports it) 262 vector<int> rates = GetSupportedRates(); 263 vector<int>::iterator it; 264 bool resample = true; 265 266 for (it = rates.begin(); it < rates.end(); it++) 267 { 268 VERBOSE(VB_AUDIO, LOC + QString("Sample rate %1 is supported") 269 .arg(*it)); 270 if (*it == audio_samplerate) 271 resample = false; 272 } 273 274 // Assume 48k if we can't get supported rates 275 if (rates.empty()) 276 rates.push_back(48000); 277 278 if (resample) 279 { 280 int error; 281 audio_samplerate = *(rates.end()); 282 VERBOSE(VB_GENERAL, LOC + QString("Using resampler. From: %1 to %2") 283 .arg(settings.samplerate).arg(audio_samplerate)); 284 src_ctx = src_new(3-src_quality, source_audio_channels, &error); 285 if (error) 286 { 287 Error(QString("Error creating resampler, the error was: %1") 288 .arg(src_strerror(error)) ); 289 src_ctx = NULL; 290 return; 291 } 292 src_data.src_ratio = (double) audio_samplerate / settings.samplerate; 293 src_data.data_in = src_in; 294 src_data.data_out = src_out; 295 src_data.output_frames = 16384*6; 296 need_resampler = true; 297 } 298 299 // Encode to AC-3 if not passing thru , there's > 2 channels 300 // and a passthru device is defined 301 if ( 302 !audio_passthru && allow_ac3_passthru && 303 (audio_channels > 2 || audio_reenc) 304 ) 305 { 306 VERBOSE(VB_AUDIO, LOC + "Creating AC-3 Encoder"); 307 encoder = new AudioOutputDigitalEncoder(); 308 if (!encoder->Init(CODEC_ID_AC3, 448000, audio_samplerate, 309 audio_channels)) 310 { 311 VERBOSE(VB_AUDIO, LOC + "Can't create AC-3 encoder"); 312 delete encoder; 313 encoder = NULL; 314 } 315 316 audio_enc = true; 317 } 318 319 if(audio_passthru || audio_enc) 320 // AC-3 output - soundcard expects a 2ch 48k stream 321 audio_channels = 2; 322 323 audio_bytes_per_sample = audio_channels * audio_bits / 8; 324 source_audio_bytes_per_sample = source_audio_channels * audio_bits / 8; 278 325 326 279 327 VERBOSE(VB_GENERAL, QString("Opening audio device '%1'. ch %2(%3) sr %4") 280 328 .arg(audio_main_device).arg(audio_channels) 281 329 .arg(source_audio_channels).arg(audio_samplerate)); … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 309 357 current_seconds = -1; 310 358 source_bitrate = -1; 311 359 312 // NOTE: this won't do anything as above samplerate vars are set equal313 // Check if we need the resampler314 if (audio_samplerate != settings.samplerate)315 {316 int error;317 VERBOSE(VB_GENERAL, LOC + QString("Using resampler. From: %1 to %2")318 .arg(settings.samplerate).arg(audio_samplerate));319 src_ctx = src_new (SRC_SINC_BEST_QUALITY, audio_channels, &error);320 if (error)321 {322 Error(QString("Error creating resampler, the error was: %1")323 .arg(src_strerror(error)) );324 return;325 }326 src_data.src_ratio = (double) audio_samplerate / settings.samplerate;327 src_data.data_in = src_in;328 src_data.data_out = src_out;329 src_data.output_frames = 16384*6;330 need_resampler = true;331 }332 333 360 if (needs_upmix) 334 361 { 335 362 VERBOSE(VB_AUDIO, LOC + QString("create upmixer")); … … void AudioOutputBase::Reconfigure(const AudioSettings &orig_settings) 344 371 (FreeSurround::SurroundMode)surround_mode); 345 372 346 373 VERBOSE(VB_AUDIO, LOC + 347 QString(" create upmixer done with surround mode %1")374 QString("Create upmixer done with surround mode %1") 348 375 .arg(surround_mode)); 349 376 } 350 377 351 378 VERBOSE(VB_AUDIO, LOC + QString("Audio Stretch Factor: %1") 352 379 .arg(audio_stretchfactor)); 353 VERBOSE(VB_AUDIO, QString("Audio Codec Used: %1")354 .arg((audio_codec) ?355 codec_id_string(audio_codec->codec_id) : "not set"));356 357 if (redo_stretch)358 {359 delete pSoundStretch;360 pSoundStretch = NULL;361 SetStretchFactorLocked(audio_stretchfactor);362 }363 else364 {365 SetStretchFactorLocked(audio_stretchfactor);366 if (pSoundStretch)367 {368 // if its passthru then we need to reencode369 if (audio_codec)370 {371 if (!encoder)372 {373 VERBOSE(VB_AUDIO, LOC +374 QString("Creating Encoder for codec %1")375 .arg(audio_codec->codec_id));376 377 encoder = new AudioOutputDigitalEncoder();378 if (!encoder->Init(audio_codec->codec_id,379 audio_codec->bit_rate,380 audio_codec->sample_rate,381 audio_codec->channels382 ))383 {384 // eeks385 delete encoder;386 encoder = NULL;387 VERBOSE(VB_AUDIO, LOC + "Failed to Create Encoder");388 }389 }390 }391 if (audio_codec && encoder)392 {393 pSoundStretch->setSampleRate(audio_codec->sample_rate);394 pSoundStretch->setChannels(audio_codec->channels);395 }396 else397 {398 pSoundStretch->setSampleRate(audio_samplerate);399 pSoundStretch->setChannels(audio_channels);400 }401 }402 }403 380 381 SetStretchFactorLocked(old_audio_stretchfactor); 382 404 383 // Setup visualisations, zero the visualisations buffers 405 384 prepareVisuals(); 406 385 … … void AudioOutputBase::KillAudio() 436 415 VERBOSE(VB_AUDIO, LOC + "Killing AudioOutputDSP"); 437 416 killaudio = true; 438 417 StopOutputThread(); 418 QMutexLocker lock1(&audio_buflock); 439 419 440 420 // Close resampler? 441 421 if (src_ctx) 422 { 442 423 src_delete(src_ctx); 424 src_ctx = NULL; 425 } 426 443 427 need_resampler = false; 444 428 445 429 // close sound stretcher … … void AudioOutputBase::KillAudio() 447 431 { 448 432 delete pSoundStretch; 449 433 pSoundStretch = NULL; 434 old_audio_stretchfactor = audio_stretchfactor; 435 audio_stretchfactor = 1.0; 450 436 } 451 437 452 438 if (encoder) … … void AudioOutputBase::KillAudio() 461 447 upmixer = NULL; 462 448 } 463 449 needs_upmix = false; 450 audio_enc = false; 464 451 465 452 CloseDevice(); 466 453 … … int AudioOutputBase::GetAudiotime(void) 562 549 ret += (now.tv_usec - audiotime_updated.tv_usec) / 1000; 563 550 ret = (long long)(ret * audio_stretchfactor); 564 551 565 #if 1566 552 VERBOSE(VB_AUDIO+VB_TIMESTAMP, 567 553 QString("GetAudiotime now=%1.%2, set=%3.%4, ret=%5, audt=%6 sf=%7") 568 554 .arg(now.tv_sec).arg(now.tv_usec) … … int AudioOutputBase::GetAudiotime(void) 571 557 .arg(audiotime) 572 558 .arg(audio_stretchfactor) 573 559 ); 574 #endif575 560 576 561 ret += audiotime; 577 562 … … void AudioOutputBase::SetAudiotime(void) 611 596 612 597 // include algorithmic latencies 613 598 if (pSoundStretch) 614 {615 // add the effect of any unused but processed samples,616 // AC3 reencode does this617 totalbuffer += (int)(pSoundStretch->numSamples() *618 audio_bytes_per_sample);619 // add the effect of unprocessed samples in time stretch algo620 599 totalbuffer += (int)((pSoundStretch->numUnprocessedSamples() * 621 600 audio_bytes_per_sample) / audio_stretchfactor); 622 }623 601 624 602 if (upmixer && needs_upmix) 625 {626 603 totalbuffer += upmixer->sampleLatency() * audio_bytes_per_sample; 627 } 604 605 if (encoder) 606 totalbuffer += encoder->Buffered(); 628 607 629 608 audiotime = audbuf_timecode - (int)(totalbuffer * 100000.0 / 630 609 (audio_bytes_per_sample * effdspstretched)); 631 610 632 611 gettimeofday(&audiotime_updated, NULL); 633 #if 1 612 634 613 VERBOSE(VB_AUDIO+VB_TIMESTAMP, 635 614 QString("SetAudiotime set=%1.%2, audt=%3 atc=%4 " 636 615 "tb=%5 sb=%6 eds=%7 abps=%8 sf=%9") … … void AudioOutputBase::SetAudiotime(void) 642 621 .arg(effdspstretched) 643 622 .arg(audio_bytes_per_sample) 644 623 .arg(audio_stretchfactor)); 645 #endif646 624 } 647 625 648 626 int AudioOutputBase::GetAudioBufferedTime(void) … … bool AudioOutputBase::AddSamples(char *buffers[], int samples, 681 659 return false; // would overflow 682 660 } 683 661 662 QMutexLocker lock1(&audio_buflock); 663 684 664 // resample input if necessary 685 665 if (need_resampler && src_ctx) 686 666 { … … bool AudioOutputBase::AddSamples(char *buffer, int samples, long long timecode) 725 705 int abps = (encoder) ? 726 706 encoder->audio_bytes_per_sample : audio_bytes_per_sample; 727 707 int len = samples * abps; 708 709 // Give original samples to mythmusic visualisation 710 dispatchVisual((unsigned char *)buffer, len, timecode, 711 source_audio_channels, audio_bits); 728 712 729 713 // Check we have enough space to write the data 730 714 if (need_resampler && src_ctx) … … bool AudioOutputBase::AddSamples(char *buffer, int samples, long long timecode) 749 733 return false; // would overflow 750 734 } 751 735 736 QMutexLocker lock1(&audio_buflock); 737 752 738 // resample input if necessary 753 739 if (need_resampler && src_ctx) 754 740 { … … int AudioOutputBase::WaitForFreeSpace(int samples) 808 794 if (src_ctx) 809 795 { 810 796 int error = src_reset(src_ctx); 811 if (error) 797 if (error) 798 { 812 799 VERBOSE(VB_IMPORTANT, LOC_ERR + QString( 813 800 "Error occured while resetting resampler: %1") 814 801 .arg(src_strerror(error))); 802 src_ctx = NULL; 803 } 815 804 } 816 805 } 817 806 } … … int AudioOutputBase::WaitForFreeSpace(int samples) 821 810 void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, 822 811 long long timecode) 823 812 { 824 audio_buflock.lock();825 826 813 int len; // = samples * audio_bytes_per_sample; 827 814 int audio_bytes = audio_bits / 8; 828 815 int org_waud = waud; … … void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, 839 826 .arg(samples * abps) 840 827 .arg(kAudioRingBufferSize-afree).arg(afree).arg(timecode) 841 828 .arg(needs_upmix)); 842 829 830 len = WaitForFreeSpace(samples); 831 843 832 if (upmixer && needs_upmix) 844 833 { 845 834 int out_samples = 0; 835 org_waud = waud; 846 836 int step = (interleaved)?source_audio_channels:1; 847 len = WaitForFreeSpace(samples); // test 837 848 838 for (int itemp = 0; itemp < samples; ) 849 839 { 850 // just in case it does a processing cycle, release the lock851 // to allow the output loop to do output852 audio_buflock.unlock();853 840 if (audio_bytes == 2) 854 841 { 855 842 itemp += upmixer->putSamples( … … void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, 866 853 source_audio_channels, 867 854 (interleaved) ? 0 : samples); 868 855 } 869 audio_buflock.lock();870 856 871 857 int copy_samples = upmixer->numSamples(); 872 858 if (copy_samples) … … void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, 900 886 } 901 887 else 902 888 { 903 len = WaitForFreeSpace(samples);904 905 889 if (interleaved) 906 890 { 907 891 char *mybuf = (char*)buffer; … … void AudioOutputBase::_AddSamples(void *buffer, bool interleaved, int samples, 936 920 } 937 921 } 938 922 939 if (samples > 0) 923 if (samples <= 0) 924 return; 925 926 if (pSoundStretch) 940 927 { 941 if (pSoundStretch) 928 // does not change the timecode, only the number of samples 929 // back to orig pos 930 org_waud = waud; 931 int bdiff = kAudioRingBufferSize - org_waud; 932 int nSamplesToEnd = bdiff/abps; 933 if (bdiff < len) 942 934 { 935 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) 936 (audiobuffer + 937 org_waud), nSamplesToEnd); 938 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer, 939 (len - bdiff) / abps); 940 } 941 else 942 { 943 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) 944 (audiobuffer + org_waud), 945 len / abps); 946 } 943 947 944 // does not change the timecode, only the number of samples 945 // back to orig pos 946 org_waud = waud; 947 int bdiff = kAudioRingBufferSize - org_waud; 948 int nSamplesToEnd = bdiff/abps; 949 if (bdiff < len) 950 { 951 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) 952 (audiobuffer + 953 org_waud), nSamplesToEnd); 954 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*)audiobuffer, 955 (len - bdiff) / abps); 948 int nSamples = pSoundStretch->numSamples(); 949 len = WaitForFreeSpace(nSamples); 950 951 while ((nSamples = pSoundStretch->numSamples())) 952 { 953 if (nSamples > nSamplesToEnd) 954 nSamples = nSamplesToEnd; 955 956 nSamples = pSoundStretch->receiveSamples( 957 (soundtouch::SAMPLETYPE*) 958 (audiobuffer + org_waud), nSamples 959 ); 960 961 if (nSamples == nSamplesToEnd) { 962 org_waud = 0; 963 nSamplesToEnd = kAudioRingBufferSize/abps; 956 964 } 957 else 958 { 959 pSoundStretch->putSamples((soundtouch::SAMPLETYPE*) 960 (audiobuffer + org_waud), 961 len / abps); 965 else { 966 org_waud += nSamples * abps; 967 nSamplesToEnd -= nSamples; 962 968 } 969 970 } 971 972 } 963 973 964 if (encoder)965 {966 // pull out a packet's worth and reencode it until we967 // don't have enough for any more packets968 soundtouch::SAMPLETYPE *temp_buff =969 (soundtouch::SAMPLETYPE*)encoder->GetFrameBuffer();970 size_t frameSize = encoder->FrameSize()/abps;974 // Encode to AC-3? 975 if (encoder) 976 { 977 978 org_waud = waud; 979 int bdiff = kAudioRingBufferSize - org_waud; 980 int to_get = 0; 971 981 972 VERBOSE(VB_AUDIO+VB_TIMESTAMP, 973 QString("_AddSamples Enc sfs=%1 bfs=%2 sss=%3") 974 .arg(frameSize) 975 .arg(encoder->FrameSize()) 976 .arg(pSoundStretch->numSamples())); 977 978 // process the same number of samples as it creates 979 // a full encoded buffer just like before 980 while (pSoundStretch->numSamples() >= frameSize) 981 { 982 int got = pSoundStretch->receiveSamples( 983 temp_buff, frameSize); 984 int amount = encoder->Encode(temp_buff); 985 986 VERBOSE(VB_AUDIO+VB_TIMESTAMP, 987 QString("_AddSamples Enc bytes=%1 got=%2 left=%3") 988 .arg(amount) 989 .arg(got) 990 .arg(pSoundStretch->numSamples())); 991 992 if (!amount) 993 continue; 994 995 //len = WaitForFreeSpace(amount); 996 char *ob = encoder->GetOutBuff(); 997 if (amount >= bdiff) 998 { 999 memcpy(audiobuffer + org_waud, ob, bdiff); 1000 ob += bdiff; 1001 amount -= bdiff; 1002 org_waud = 0; 1003 } 1004 if (amount > 0) 1005 memcpy(audiobuffer + org_waud, ob, amount); 1006 1007 bdiff = kAudioRingBufferSize - amount; 1008 org_waud = (org_waud + amount) % kAudioRingBufferSize; 1009 } 1010 } 1011 else 982 if (bdiff < len) 983 { 984 encoder->Encode(audiobuffer + org_waud, bdiff); 985 to_get = encoder->Encode(audiobuffer, len - bdiff); 986 } 987 else 988 to_get = encoder->Encode(audiobuffer + org_waud, len); 989 990 if (to_get > 0) 991 { 992 993 if (to_get >= bdiff) 1012 994 { 1013 int newLen = 0; 1014 int nSamples; 1015 len = WaitForFreeSpace(pSoundStretch->numSamples() * 1016 audio_bytes_per_sample); 1017 do 1018 { 1019 int samplesToGet = len/audio_bytes_per_sample; 1020 if (samplesToGet > nSamplesToEnd) 1021 { 1022 samplesToGet = nSamplesToEnd; 1023 } 1024 1025 nSamples = pSoundStretch->receiveSamples( 1026 (soundtouch::SAMPLETYPE*) 1027 (audiobuffer + org_waud), samplesToGet); 1028 if (nSamples == nSamplesToEnd) 1029 { 1030 org_waud = 0; 1031 nSamplesToEnd = kAudioRingBufferSize/audio_bytes_per_sample; 1032 } 1033 else 1034 { 1035 int bufsz = nSamples * audio_bytes_per_sample; 1036 org_waud = (org_waud + bufsz) % kAudioRingBufferSize; 1037 nSamplesToEnd -= nSamples; 1038 } 1039 1040 newLen += nSamples * audio_bytes_per_sample; 1041 len -= nSamples * audio_bytes_per_sample; 1042 } while (nSamples > 0); 995 encoder->GetFrames(audiobuffer + org_waud, bdiff); 996 to_get -= bdiff; 997 org_waud = 0; 1043 998 } 1044 } 999 if (to_get > 0) 1000 encoder->GetFrames(audiobuffer + org_waud, to_get); 1045 1001 1046 waud = org_waud; 1047 lastaudiolen = audiolen(false); 1002 org_waud += to_get; 1048 1003 1049 if (timecode < 0)1050 {1051 // mythmusic doesn't give timestamps..1052 timecode = (int)((samples_buffered * 100000.0) / effdsp);1053 1004 } 1054 1055 samples_buffered += samples;1056 1057 /* we want the time at the end -- but the file format stores1058 time at the start of the chunk. */1059 // even with timestretch, timecode is still calculated from original1060 // sample count1061 audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp);1062 1005 1063 if (interleaved)1064 {1065 dispatchVisual((unsigned char *)buffer, len, timecode,1066 source_audio_channels, audio_bits);1067 }1068 1006 } 1069 1007 1070 audio_buflock.unlock(); 1008 waud = org_waud; 1009 lastaudiolen = audiolen(false); 1010 1011 if (timecode < 0) 1012 // mythmusic doesn't give timestamps.. 1013 timecode = (int)((samples_buffered * 100000.0) / effdsp); 1014 1015 samples_buffered += samples; 1016 1017 /* we want the time at the end -- but the file format stores 1018 time at the start of the chunk. */ 1019 // even with timestretch, timecode is still calculated from original 1020 // sample count 1021 audbuf_timecode = timecode + (int)((samples * 100000.0) / effdsp); 1022 1071 1023 } 1072 1024 1073 1025 void AudioOutputBase::Status() -
mythtv/libs/libmyth/audiooutputbase.h
diff --git a/mythtv/libs/libmyth/audiooutputbase.h b/mythtv/libs/libmyth/audiooutputbase.h index 1f636e2..4bcaaa4 100644
a b class AudioOutputBase : public AudioOutput, public QThread 43 43 44 44 virtual void SetStretchFactor(float factor); 45 45 virtual float GetStretchFactor(void) const; 46 virtual bool ToggleUpmix(void); 46 47 47 48 virtual void Reset(void); 48 49 … … class AudioOutputBase : public AudioOutput, public QThread 85 86 virtual void WriteAudio(unsigned char *aubuf, int size) = 0; 86 87 virtual int GetSpaceOnSoundcard(void) const = 0; 87 88 virtual int GetBufferedOnSoundcard(void) const = 0; 89 virtual vector<int> GetSupportedRates(void) = 0; 90 88 91 /// You need to call this from any implementation in the dtor. 89 92 void KillAudio(void); 90 93 … … class AudioOutputBase : public AudioOutput, public QThread 122 125 123 126 // Basic details about the audio stream 124 127 int audio_channels; 128 int audio_codec; 125 129 int audio_bytes_per_sample; 126 130 int audio_bits; 127 131 int audio_samplerate; … … class AudioOutputBase : public AudioOutput, public QThread 132 136 QString audio_passthru_device; 133 137 134 138 bool audio_passthru; 139 bool audio_enc; 140 bool audio_reenc; 135 141 136 142 float audio_stretchfactor; 137 AVCodecContext *audio_codec;138 143 AudioOutputSource source; 139 144 140 145 bool killaudio; … … class AudioOutputBase : public AudioOutput, public QThread 144 149 bool buffer_output_data_for_use; // used by AudioOutputNULL 145 150 146 151 int configured_audio_channels; 152 int orig_config_channels; 153 int src_quality; 147 154 148 155 private: 149 156 // resampler … … class AudioOutputBase : public AudioOutput, public QThread 156 163 FreeSurround *upmixer; 157 164 158 165 int source_audio_channels; 166 int source_audio_samplerate; 159 167 int source_audio_bytes_per_sample; 160 168 bool needs_upmix; 161 169 int surround_mode; 170 bool allow_ac3_passthru; 171 float old_audio_stretchfactor; 162 172 163 173 bool blocking; // do AddSamples calls block? 164 174 -
mythtv/libs/libmyth/audiooutputca.h
diff --git a/mythtv/libs/libmyth/audiooutputca.h b/mythtv/libs/libmyth/audiooutputca.h index 5afe3e0..14192bb 100644
a b protected: 49 49 50 50 virtual bool StartOutputThread(void) { return true; } 51 51 virtual void StopOutputThread(void) {} 52 virtual vector<int> GetSupportedRates(void) 53 { vector<int> rates; return rates; } 52 54 53 55 private: 54 56 -
mythtv/libs/libmyth/audiooutputdigitalencoder.cpp
diff --git a/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp b/mythtv/libs/libmyth/audiooutputdigitalencoder.cpp index a9688c9..34f9e5e 100644
a b extern "C" { 32 32 AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) : 33 33 audio_bytes_per_sample(0), 34 34 av_context(NULL), 35 outbuf(NULL), 36 outbuf_size(0), 37 frame_buffer(NULL), 35 outbuflen(0), 36 inbuflen(0), 38 37 one_frame_bytes(0) 39 38 { 40 39 } … … void AudioOutputDigitalEncoder::Dispose() 52 51 av_free(av_context); 53 52 av_context = NULL; 54 53 } 55 56 if (outbuf)57 {58 delete [] outbuf;59 outbuf = NULL;60 outbuf_size = 0;61 }62 63 if (frame_buffer)64 {65 delete [] frame_buffer;66 frame_buffer = NULL;67 one_frame_bytes = 0;68 }69 54 } 70 55 71 56 //CODEC_ID_AC3 … … bool AudioOutputDigitalEncoder::Init( 80 65 .arg(bitrate) 81 66 .arg(samplerate) 82 67 .arg(channels)); 83 84 //codec = avcodec_find_encoder(codec_id); 68 69 // We need to do this when called from mythmusic 70 avcodec_init(); 71 avcodec_register_all(); 85 72 // always AC3 as there is no DTS encoder at the moment 2005/1/9 86 73 codec = avcodec_find_encoder(CODEC_ID_AC3); 87 74 if (!codec) … … bool AudioOutputDigitalEncoder::Init( 110 97 audio_bytes_per_sample = bytes_per_frame; 111 98 one_frame_bytes = bytes_per_frame * av_context->frame_size; 112 99 113 outbuf_size = 16384; // ok for AC3 but DTS?114 outbuf = new char [outbuf_size];115 100 VERBOSE(VB_AUDIO, QString("DigitalEncoder::Init fs=%1, bpf=%2 ofb=%3") 116 101 .arg(av_context->frame_size) 117 102 .arg(bytes_per_frame) … … typedef struct { 259 244 static int encode_frame( 260 245 bool dts, 261 246 unsigned char *data, 262 size_t &len)247 size_t enc_len) 263 248 { 264 249 unsigned char *payload = data + 8; // skip header, currently 52 or 54bits 265 size_t enc_len;266 250 int flags, sample_rate, bit_rate; 267 251 268 252 // we don't do any length/crc validation of the AC3 frame here; presumably … … static int encode_frame( 273 257 // ignore, and if so, may as well just assume that it will ignore 274 258 // anything with a bad CRC... 275 259 276 uint nr_samples = 0, block_len; 260 uint nr_samples = 0, block_len = 0; 261 277 262 if (dts) 278 263 { 279 264 enc_len = dts_syncinfo(payload, &flags, &sample_rate, &bit_rate); … … static int encode_frame( 302 287 } 303 288 } 304 289 305 if (enc_len == 0 || enc_len > len)306 {307 int l = len;308 len = 0;309 return l;310 }311 312 290 enc_len = std::min((uint)enc_len, block_len - 8); 313 291 314 292 //uint32_t x = *(uint32_t*)payload; … … static int encode_frame( 361 339 data[6] = (enc_len << 3) & 0xFF; 362 340 data[7] = (enc_len >> 5) & 0xFF; 363 341 memset(payload + enc_len, 0, block_len - 8 - enc_len); 364 len = block_len;365 342 366 343 return enc_len; 367 344 } 368 345 369 // must have exactly 1 frames worth of data 370 size_t AudioOutputDigitalEncoder::Encode(short *buff) 346 size_t AudioOutputDigitalEncoder::Encode(void *buf, int len) 371 347 { 372 int encsize = 0;373 348 size_t outsize = 0; 374 349 375 // put data in the correct spot for encode frame 376 outsize = avcodec_encode_audio( 377 av_context, ((uchar*)outbuf) + 8, outbuf_size - 8, buff); 378 379 size_t tmpsize = outsize; 380 381 outsize = MAX_AC3_FRAME_SIZE; 382 encsize = encode_frame( 383 /*av_context->codec_id==CODEC_ID_DTS*/ false, 384 (unsigned char*)outbuf, outsize); 350 int fs = FrameSize(); 351 memcpy(inbuf+inbuflen, buf, len); 352 inbuflen += len; 353 int frames = inbuflen / fs; 385 354 386 VERBOSE(VB_AUDIO+VB_TIMESTAMP, 387 QString("DigitalEncoder::Encode len1=%1 len2=%2 finallen=%3") 388 .arg(tmpsize).arg(encsize).arg(outsize)); 355 while (frames--) 356 { 357 // put data in the correct spot for encode frame 358 outsize = avcodec_encode_audio( 359 av_context, ((uchar*)outbuf) + outbuflen + 8, OUTBUFSIZE - 8, (short int *)inbuf); 360 361 encode_frame( 362 /*av_context->codec_id==CODEC_ID_DTS*/ false, 363 (unsigned char*)outbuf + outbuflen, outsize 364 ); 365 366 outbuflen += MAX_AC3_FRAME_SIZE; 367 inbuflen -= fs; 368 memmove(inbuf, inbuf+fs, inbuflen); 369 } 370 371 return outbuflen; 372 } 389 373 390 return outsize; 374 void AudioOutputDigitalEncoder::GetFrames(void *ptr, int maxlen) 375 { 376 int len = (maxlen < outbuflen ? maxlen : outbuflen); 377 memcpy(ptr, outbuf, len); 378 outbuflen -= len; 379 memmove(outbuf, outbuf+len, outbuflen); 391 380 } -
mythtv/libs/libmyth/audiooutputdigitalencoder.h
diff --git a/mythtv/libs/libmyth/audiooutputdigitalencoder.h b/mythtv/libs/libmyth/audiooutputdigitalencoder.h index 8a4689a..8d81298 100644
a b extern "C" { 5 5 #include "libavcodec/avcodec.h" 6 6 }; 7 7 8 #define INBUFSIZE 131072 9 #define OUTBUFSIZE 98304 10 8 11 class AudioOutputDigitalEncoder 9 12 { 10 13 public: … … class AudioOutputDigitalEncoder 13 16 14 17 bool Init(CodecID codec_id, int bitrate, int samplerate, int channels); 15 18 void Dispose(void); 16 size_t Encode(short * buff); 17 18 inline char *GetFrameBuffer(void); 19 size_t Encode(void *buf, int len); 20 void GetFrames(void *ptr, int maxlen); 19 21 size_t FrameSize(void) const { return one_frame_bytes; } 20 char *GetOutBuff(void) const { return outbuf;}22 int Buffered(void) const { return inbuflen; } 21 23 22 24 public: 23 25 size_t audio_bytes_per_sample; 24 26 25 27 private: 26 28 AVCodecContext *av_context; 27 char *outbuf; 28 int outbuf_size; 29 char *frame_buffer; 29 char outbuf[OUTBUFSIZE]; 30 char inbuf[INBUFSIZE]; 31 int outbuflen; 32 int inbuflen; 30 33 size_t one_frame_bytes; 31 34 }; 32 35 33 inline char *AudioOutputDigitalEncoder::GetFrameBuffer(void)34 {35 if (!frame_buffer && av_context)36 frame_buffer = new char [one_frame_bytes];37 38 return frame_buffer;39 }40 41 36 #endif -
mythtv/libs/libmyth/audiooutputdx.h
diff --git a/mythtv/libs/libmyth/audiooutputdx.h b/mythtv/libs/libmyth/audiooutputdx.h index e0d3e24..485adb1 100644
a b public: 15 15 16 16 virtual int GetVolumeChannel(int channel) const; 17 17 virtual void SetVolumeChannel(int channel, int volume); 18 virtual vector<int> GetSupportedRates(void) 19 { vector<int> rates; return rates; } 18 20 19 21 protected: 20 22 virtual bool OpenDevice(void); -
mythtv/libs/libmyth/audiooutputjack.cpp
diff --git a/mythtv/libs/libmyth/audiooutputjack.cpp b/mythtv/libs/libmyth/audiooutputjack.cpp index 7e0ad0d..f9a5848 100644
a b AudioOutputJACK::AudioOutputJACK(const AudioSettings &settings) : 30 30 Reconfigure(settings); 31 31 } 32 32 33 vector<int> AudioOutputJACK::GetSupportedRates() 34 { 35 const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; 36 vector<int> rates(srates, srates + sizeof(srates) / sizeof(int) ); 37 unsigned long jack_port_flags = 0; 38 unsigned int jack_port_name_count = 1; 39 const char *jack_port_name = audio_main_device.ascii(); 40 int err = -1; 41 audioid = -1; 42 vector<int>::iterator it; 43 44 for (it = rates.begin(); it < rates.end(); it++) 45 { 46 err = JACK_OpenEx(&audioid, 16, (unsigned long *)it, 47 2, 2, &jack_port_name, jack_port_name_count, 48 jack_port_flags); 49 50 if (err == 1) 51 { 52 Error(QString("Error connecting to jackd: %1. Is it running?") 53 .arg(audio_main_device)); 54 rates.clear(); 55 return rates; 56 } 57 else if (err == 2) 58 rates.erase(it--); 59 60 JACK_Close(audioid); 61 audioid = -1; 62 63 } 64 65 return rates; 66 } 67 68 33 69 AudioOutputJACK::~AudioOutputJACK() 34 70 { 35 71 // Close down all audio stuff -
mythtv/libs/libmyth/audiooutputjack.h
diff --git a/mythtv/libs/libmyth/audiooutputjack.h b/mythtv/libs/libmyth/audiooutputjack.h index f87538e..259c202 100644
a b class AudioOutputJACK : public AudioOutputBase 23 23 virtual void WriteAudio(unsigned char *aubuf, int size); 24 24 virtual int GetSpaceOnSoundcard(void) const; 25 25 virtual int GetBufferedOnSoundcard(void) const; 26 virtual vector<int> GetSupportedRates(void); 26 27 27 28 private: 28 29 -
mythtv/libs/libmyth/audiooutputnull.h
diff --git a/mythtv/libs/libmyth/audiooutputnull.h b/mythtv/libs/libmyth/audiooutputnull.h index 7eb247d..78a0f54 100644
a b class AudioOutputNULL : public AudioOutputBase 41 41 virtual void WriteAudio(unsigned char *aubuf, int size); 42 42 virtual int GetSpaceOnSoundcard(void) const; 43 43 virtual int GetBufferedOnSoundcard(void) const; 44 virtual vector<int> GetSupportedRates(void) 45 { vector<int> rates; return rates; } 44 46 45 47 private: 46 48 QMutex pcm_output_buffer_mutex; -
mythtv/libs/libmyth/audiooutputoss.cpp
diff --git a/mythtv/libs/libmyth/audiooutputoss.cpp b/mythtv/libs/libmyth/audiooutputoss.cpp index 127bf6e..8d3b135 100644
a b AudioOutputOSS::~AudioOutputOSS() 42 42 KillAudio(); 43 43 } 44 44 45 vector<int> AudioOutputOSS::GetSupportedRates() 46 { 47 const int srates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000 }; 48 vector<int> rates(srates, srates + sizeof(srates) / sizeof(int) ); 49 audiofd = open(audio_main_device.toAscii(), O_WRONLY | O_NONBLOCK); 50 51 if (audiofd < 0) 52 { 53 VERBOSE(VB_IMPORTANT, QString("Error opening audio device (%1), the" 54 " error was: %2").arg(audio_main_device).arg(strerror(errno))); 55 rates.clear(); 56 return rates; 57 } 58 59 vector<int>::iterator it; 60 61 for (it = rates.begin(); it < rates.end(); it++) 62 if(ioctl(audiofd, SNDCTL_DSP_SPEED, &audio_samplerate) < 0) 63 rates.erase(it--); 64 65 close(audiofd); 66 audiofd = -1; 67 68 return rates; 69 } 70 45 71 bool AudioOutputOSS::OpenDevice() 46 72 { 47 73 numbadioctls = 0; -
mythtv/libs/libmyth/audiooutputoss.h
diff --git a/mythtv/libs/libmyth/audiooutputoss.h b/mythtv/libs/libmyth/audiooutputoss.h index d5105a2..e218e31 100644
a b class AudioOutputOSS : public AudioOutputBase 23 23 virtual void WriteAudio(unsigned char *aubuf, int size); 24 24 virtual int GetSpaceOnSoundcard(void) const; 25 25 virtual int GetBufferedOnSoundcard(void) const; 26 virtual vector<int> GetSupportedRates(void); 26 27 27 28 private: 28 29 void VolumeInit(void); -
mythtv/libs/libmyth/audiooutputwin.h
diff --git a/mythtv/libs/libmyth/audiooutputwin.h b/mythtv/libs/libmyth/audiooutputwin.h index 8ba37e4..5fd5945 100644
a b class AudioOutputWin : public AudioOutputBase 23 23 virtual void WriteAudio(unsigned char *aubuf, int size); 24 24 virtual int GetSpaceOnSoundcard(void) const; 25 25 virtual int GetBufferedOnSoundcard(void) const; 26 virtual vector<int> GetSupportedRates(void) 27 { vector<int> rates; return rates; } 26 28 27 29 protected: 28 30 AudioOutputWinPrivate *m_priv; -
mythtv/libs/libmyth/audiosettings.cpp
diff --git a/mythtv/libs/libmyth/audiosettings.cpp b/mythtv/libs/libmyth/audiosettings.cpp index 68e515b..1f4ef18 100644
a b AudioSettings::AudioSettings() : 12 12 passthru_device(QString::null), 13 13 bits(-1), 14 14 channels(-1), 15 codec(0), 15 16 samplerate(-1), 16 17 set_initial_vol(false), 17 18 use_passthru(false), 18 codec(NULL),19 19 source(AUDIOOUTPUT_UNKNOWN) 20 20 { 21 21 } … … AudioSettings::AudioSettings(const AudioSettings &other) : 25 25 passthru_device(other.passthru_device), 26 26 bits(other.bits), 27 27 channels(other.channels), 28 codec(other.codec), 28 29 samplerate(other.samplerate), 29 30 set_initial_vol(other.set_initial_vol), 30 31 use_passthru(other.use_passthru), 31 codec(other.codec),32 32 source(other.source) 33 33 { 34 34 } … … AudioSettings::AudioSettings( 38 38 const QString &audio_passthru_device, 39 39 int audio_bits, 40 40 int audio_channels, 41 int audio_codec, 41 42 int audio_samplerate, 42 43 AudioOutputSource audio_source, 43 44 bool audio_set_initial_vol, 44 bool audio_use_passthru, 45 void *audio_codec) : 45 bool audio_use_passthru) : 46 46 main_device(audio_main_device), 47 47 passthru_device(audio_passthru_device), 48 48 bits(audio_bits), 49 49 channels(audio_channels), 50 codec(audio_codec), 50 51 samplerate(audio_samplerate), 51 52 set_initial_vol(audio_set_initial_vol), 52 53 use_passthru(audio_use_passthru), 53 codec(audio_codec),54 54 source(audio_source) 55 55 { 56 56 } … … AudioSettings::AudioSettings( 58 58 AudioSettings::AudioSettings( 59 59 int audio_bits, 60 60 int audio_channels, 61 int audio_codec, 61 62 int audio_samplerate, 62 bool audio_use_passthru, 63 void *audio_codec) : 63 bool audio_use_passthru) : 64 64 main_device(QString::null), 65 65 passthru_device(QString::null), 66 66 bits(audio_bits), 67 67 channels(audio_channels), 68 codec(audio_codec), 68 69 samplerate(audio_samplerate), 69 70 set_initial_vol(false), 70 71 use_passthru(audio_use_passthru), 71 codec(audio_codec),72 72 source(AUDIOOUTPUT_UNKNOWN) 73 73 { 74 74 } -
mythtv/libs/libmyth/audiosettings.h
diff --git a/mythtv/libs/libmyth/audiosettings.h b/mythtv/libs/libmyth/audiosettings.h index f9349f4..ba38f51 100644
a b class MPUBLIC AudioSettings 29 29 const QString &audio_passthru_device, 30 30 int audio_bits, 31 31 int audio_channels, 32 int audio_codec, 32 33 int audio_samplerate, 33 34 AudioOutputSource audio_source, 34 35 bool audio_set_initial_vol, 35 bool audio_use_passthru, 36 void *audio_codec = NULL); 36 bool audio_use_passthru); 37 37 38 38 AudioSettings(int audio_bits, 39 39 int audio_channels, 40 int audio_codec, 40 41 int audio_samplerate, 41 bool audio_use_passthru, 42 void *audio_codec = NULL); 42 bool audio_use_passthru); 43 43 44 44 void FixPassThrough(void); 45 45 void TrimDeviceType(void); … … class MPUBLIC AudioSettings 54 54 public: 55 55 int bits; 56 56 int channels; 57 int codec; 57 58 int samplerate; 58 59 bool set_initial_vol; 59 60 bool use_passthru; 60 void *codec;61 61 AudioOutputSource source; 62 62 }; 63 63 -
mythtv/libs/libmythfreesurround/el_processor.cpp
diff --git a/mythtv/libs/libmythfreesurround/el_processor.cpp b/mythtv/libs/libmythfreesurround/el_processor.cpp index 8f24737..1cf0dc8 100644
a b typedef std::complex<float> cfloat; 40 40 41 41 const float PI = 3.141592654; 42 42 const float epsilon = 0.000001; 43 //const float center_level = 0.5*sqrt(0.5); // gain of the center channel 44 //const float center_level = sqrt(0.5); // gain of the center channel 45 const float center_level = 1.0; // gain of the center channel 46 //const float center_level = 0.5; // gain of the center channel 47 48 // should be .6-.7 49 // but with centerlevel 2x what its supposed to be, we halve 0.68 50 // to keep center from clipping 51 //const float window_gain = 0.34; 52 //const float window_gain = 0.68; 53 const float window_gain = 0.95; // to prive a bit of margin 43 const float center_level = 0.5*sqrt(0.5); 54 44 55 45 // private implementation of the surround decoder 56 46 class decoder_impl { … … public: 98 88 outbuf[c].resize(N); 99 89 filter[c].resize(N); 100 90 } 101 // DC component of filters is always 0102 for (unsigned c=0;c<5;c++)103 {104 filter[c][0] = 0.0;105 filter[c][1] = 0.0;106 filter[c][halfN] = 0.0;107 }108 91 sample_rate(48000); 109 92 // generate the window function (square root of hann, b/c it is applied before and after the transform) 110 93 wnd.resize(N); 111 // dft normalization included in the window for zero cost scaling 112 // also add a gain factor of *2 due to processing gain in algo (see center_level) 113 surround_gain(1.0); 94 for (unsigned k=0;k<N;k++) 95 wnd[k] = sqrt(0.5*(1-cos(2*PI*k/N))/N); 114 96 current_buf = 0; 115 97 // set the default coefficients 116 98 surround_coefficients(0.8165,0.5774); … … public: 192 174 // set lfe filter params 193 175 void sample_rate(unsigned int srate) { 194 176 // lfe filter is just straight through band limited 195 unsigned int cutoff = ( 250*N)/srate;177 unsigned int cutoff = (30*N)/srate; 196 178 for (unsigned f=0;f<=halfN;f++) { 197 if ( (f>=2) && (f<cutoff))198 filter[5][f] = 1.0;179 if (f<cutoff) 180 filter[5][f] = 0.5*sqrt(0.5); 199 181 else 200 182 filter[5][f] = 0.0; 201 183 } … … public: 214 196 E = (o+v)*n; F = (o+u)*n; G = (o-v)*n; H = (o-u)*n; 215 197 } 216 198 217 void surround_gain(float gain) {218 master_gain = gain * window_gain * 0.5 * 0.25;219 for (unsigned k=0;k<N;k++)220 wnd[k] = sqrt(master_gain*(1-cos(2*PI*k/N))/N);221 }222 223 199 // set the phase shifting mode 224 200 void phase_mode(unsigned mode) { 225 201 const float modes[4][2] = {{0,0},{0,PI},{PI,0},{-PI/2,PI/2}}; … … private: 290 266 291 267 // 2. compare amplitude and phase of each DFT bin and produce the X/Y coordinates in the sound field 292 268 // but dont do DC or N/2 component 293 for (unsigned f= 2;f<halfN;f++) {269 for (unsigned f=0;f<halfN;f++) { 294 270 // get left/right amplitudes/phases 295 271 float ampL = amplitude(dftL[f]), ampR = amplitude(dftR[f]); 296 272 float phaseL = phase(dftL[f]), phaseR = phase(dftR[f]); … … private: 305 281 phaseDiff = abs(phaseDiff); 306 282 307 283 if (linear_steering) { 308 /* cfloat w = polar(sqrt(ampL*ampL+ampR*ampR), (phaseL+phaseR)/2);309 cfloat lt = cfloat(dftL[f][0],dftL[f][1])/w, rt = cfloat(dftR[f][0],dftR[f][1])/w; */310 // xfs[f] = -(C*(rt-H) - B*E + F*A + G*(D-lt)) / (G*A - C*E).real();311 // yfs[f] = (rt - (xfs[f]*E+H))/(F+xfs[f]*G);312 313 /*314 Problem:315 This assumes that the values are interpolated linearly between the cardinal points.316 But this way we have no chance of knowing the average volume...317 - Can we solve that computing everything under the assumption of normalized volume?318 No. Seemingly not.319 - Maybe we should add w explitcitly into the equation and see if we can solve it...320 */321 322 323 //cfloat lt(0.5,0),rt(0.5,0);324 //cfloat x(0,0), y(1,0);325 /*cfloat p = (C*(rt-H) - B*E + F*A + G*(D-lt)) / (G*A - C*E);326 cfloat q = B*(rt+H) + F*(D-lt) / (G*A - C*E);327 cfloat s = sqrt(p*p/4.0f - q);328 cfloat x = -p;329 cfloat x1 = -p/2.0f + s;330 cfloat x2 = -p/2.0f - s;331 float x = 0;332 if (x1.real() >= -1 && x1.real() <= 1)333 x = x1.real();334 else if (x2.real() >= -1 && x2.real() <= 1)335 x = x2.real();*/336 337 //cfloat yp = (rt - (x*E+H))/(F+x*G);338 //cfloat xp = (lt - (y*B+D))/(A+y*C);339 340 /*xfs[f] = x;341 yfs[f] = y.real();*/342 343 284 // --- this is the fancy new linear mode --- 344 285 345 286 // get sound field x/y position … … private: 597 538 float surround_high,surround_low; // high and low surround mixing coefficient (e.g. 0.8165/0.5774) 598 539 float surround_balance; // the xfs balance that follows from the coeffs 599 540 float surround_level; // gain for the surround channels (follows from the coeffs 600 float master_gain; // gain for all channels601 541 float phase_offsetL, phase_offsetR;// phase shifts to be applied to the rear channels 602 542 float front_separation; // front stereo separation 603 543 float rear_separation; // rear stereo separation … … void fsurround_decoder::flush() { impl->flush(); } 625 565 626 566 void fsurround_decoder::surround_coefficients(float a, float b) { impl->surround_coefficients(a,b); } 627 567 628 void fsurround_decoder::gain(float gain) { impl->surround_gain(gain); }629 630 568 void fsurround_decoder::phase_mode(unsigned mode) { impl->phase_mode(mode); } 631 569 632 570 void fsurround_decoder::steering_mode(bool mode) { impl->steering_mode(mode); } -
mythtv/libs/libmythfreesurround/el_processor.h
diff --git a/mythtv/libs/libmythfreesurround/el_processor.h b/mythtv/libs/libmythfreesurround/el_processor.h index 021786a..26452f6 100644
a b public: 47 47 // 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. 48 48 void surround_coefficients(float a, float b); 49 49 50 // override for master surround gain51 void gain(float gain);52 53 50 // set the phase shifting mode for decoding 54 51 // 0 = (+0°,+0°) - music mode 55 52 // 1 = (+0°,+180°) - PowerDVD compatibility -
mythtv/libs/libmythfreesurround/freesurround.cpp
diff --git a/mythtv/libs/libmythfreesurround/freesurround.cpp b/mythtv/libs/libmythfreesurround/freesurround.cpp index fc7ed8f..5be77c4 100644
a b using namespace std; 63 63 const unsigned default_block_size = 8192; 64 64 // there will be a slider for this in the future 65 65 //const float master_gain = 1.0; 66 //#define MASTER_GAIN * master_gain 66 //#define MASTER_GAIN * master_gain 67 67 #define MASTER_GAIN 68 //const float master_gain = 1.0/(1<<15); 69 //const float inv_master_gain = (1<<15); 68 //const float inv_master_gain = 1.0; 70 69 //#define INV_MASTER_GAIN * inv_master_gain 71 70 #define INV_MASTER_GAIN 72 71 … … FreeSurround::FreeSurround(uint srate, bool moviemode, SurroundMode smode) : 192 191 if (moviemode) 193 192 { 194 193 params.phasemode = 1; 195 params.center_width = 0;196 params. gain = 1.0;194 params.center_width = 25; 195 params.dimension = 0.5; 197 196 } 198 197 else 199 198 { 200 params.center_width = 70; 201 // for 50, gain should be about 1.9, c/lr about 2.7 202 // for 70, gain should be about 3.1, c/lr about 1.5 203 params.gain = 3.1; 199 params.center_width = 65; 200 params.dimension = 0.3; 204 201 } 205 202 switch (surround_mode) 206 203 { … … void FreeSurround::SetParams() 236 233 decoder->phase_mode(params.phasemode); 237 234 decoder->surround_coefficients(params.coeff_a, params.coeff_b); 238 235 decoder->separation(params.front_sep/100.0,params.rear_sep/100.0); 239 decoder->gain(params.gain);240 236 } 241 237 } 242 238 … … FreeSurround::fsurround_params::fsurround_params( 250 246 phasemode(0), 251 247 steering(1), 252 248 front_sep(100), 253 rear_sep(100), 254 gain(1.0) 249 rear_sep(100) 255 250 { 256 251 } 257 252 … … uint FreeSurround::putSamples(short* samples, uint numSamples, uint numChannels, 329 324 for (i=0;(i<numSamples) && (ic < bs);i++,ic++) 330 325 { 331 326 int16bufs->l[ic] = *samples++ >> 1; 332 int16bufs->c[ic] = *samples++ >> 1;333 327 int16bufs->r[ic] = *samples++ >> 1; 328 int16bufs->c[ic] = *samples++ >> 1; 329 int16bufs->lfe[ic] = *samples++ >> 1; 334 330 int16bufs->ls[ic] = *samples++ >> 1; 335 331 int16bufs->rs[ic] = *samples++ >> 1; 336 int16bufs->lfe[ic] = *samples++ >> 1;337 332 } 338 333 break; 339 334 } … … uint FreeSurround::putSamples(short* samples, uint numSamples, uint numChannels, 391 386 for (i=0;i<numSamples;i++) 392 387 { 393 388 *l++ = *samples++ >> 1; 394 *c++ = *samples++ >> 1;395 389 *r++ = *samples++ >> 1; 390 *c++ = *samples++ >> 1; 391 *lfe++ = *samples++ >> 1; 396 392 *ls++ = *samples++ >> 1; 397 393 *rs++ = *samples++ >> 1; 398 *lfe++ = *samples++ >> 1;399 394 } 400 395 } break; 401 396 } … … uint FreeSurround::putSamples(char* samples, uint numSamples, uint numChannels, 479 474 for (i=0;(i<numSamples) && (ic < bs);i++,ic++) 480 475 { 481 476 int16bufs->l[ic] = *samples++ << 7; 482 int16bufs->c[ic] = *samples++ << 7;483 477 int16bufs->r[ic] = *samples++ << 7; 478 int16bufs->c[ic] = *samples++ << 7; 479 int16bufs->lfe[ic] = *samples++ << 7; 484 480 int16bufs->ls[ic] = *samples++ << 7; 485 481 int16bufs->rs[ic] = *samples++ << 7; 486 int16bufs->lfe[ic] = *samples++ << 7;487 482 } 488 483 break; 489 484 } … … uint FreeSurround::putSamples(char* samples, uint numSamples, uint numChannels, 541 536 for (i=0;i<numSamples;i++) 542 537 { 543 538 *l++ = *samples++ << 7; 544 *c++ = *samples++ << 7;545 539 *r++ = *samples++ << 7; 540 *c++ = *samples++ << 7; 541 *lfe++ = *samples++ << 7; 546 542 *ls++ = *samples++ << 7; 547 543 *rs++ = *samples++ << 7; 548 *lfe++ = *samples++ << 7;549 544 } 550 545 } break; 551 546 } … … void FreeSurround::process_block() 655 650 { 656 651 if (decoder) 657 652 { 658 // actually these params need only be set when they change... but it doesn't hurt659 #if 0660 decoder->steering_mode(params.steering);661 decoder->phase_mode(params.phasemode);662 decoder->surround_coefficients(params.coeff_a, params.coeff_b);663 decoder->separation(params.front_sep/100.0,params.rear_sep/100.0);664 #endif665 // decode the bufs->block666 //decoder->decode(input,output,params.center_width/100.0,params.dimension/100.0);667 //decoder->decode(output,params.center_width/100.0,params.dimension/100.0);668 653 decoder->decode(params.center_width/100.0,params.dimension/100.0); 669 654 } 670 655 } -
mythtv/libs/libmythsamplerate/samplerate.c
diff --git a/mythtv/libs/libmythsamplerate/samplerate.c b/mythtv/libs/libmythsamplerate/samplerate.c index bb85f8b..a53a942 100644
a b src_float_to_short_array (const float *in, short *out, int len) 452 452 { len -- ; 453 453 454 454 scaled_value = in [len] * (8.0 * 0x10000000) ; 455 if ( !HAVE_CPU_CLIPS_POSITIVE &&scaled_value >= (1.0 * 0x7FFFFFFF))455 if (scaled_value >= (1.0 * 0x7FFFFFFF)) 456 456 { out [len] = 32767 ; 457 457 continue ; 458 458 } ; 459 if ( !HAVE_CPU_CLIPS_NEGATIVE &&scaled_value <= (-8.0 * 0x10000000))459 if (scaled_value <= (-8.0 * 0x10000000)) 460 460 { out [len] = -32768 ; 461 461 continue ; 462 462 } ; -
mythtv/libs/libmythtv/NuppelVideoPlayer.cpp
diff --git a/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp b/mythtv/libs/libmythtv/NuppelVideoPlayer.cpp index 955596a..61a0668 100644
a b NuppelVideoPlayer::NuppelVideoPlayer() 218 218 audioOutput(NULL), 219 219 audio_main_device(QString::null), 220 220 audio_passthru_device(QString::null), 221 audio_channels(2), audio_ bits(-1),222 audio_ samplerate(44100), audio_stretchfactor(1.0f),223 audio_ codec(NULL),audio_lock(QMutex::Recursive),221 audio_channels(2), audio_codec(0), 222 audio_bits(-1), audio_samplerate(44100), 223 audio_stretchfactor(1.0f), audio_lock(QMutex::Recursive), 224 224 // Picture-in-Picture stuff 225 225 pip_active(false), pip_visible(true), 226 226 // Preview window support … … QString NuppelVideoPlayer::ReinitAudio(void) 915 915 bool setVolume = gContext->GetNumSetting("MythControlsVolume", 1); 916 916 audioOutput = AudioOutput::OpenAudio(audio_main_device, 917 917 audio_passthru_device, 918 audio_bits, audio_channels, 919 audio_ samplerate,918 audio_bits, audio_channels, 919 audio_codec, audio_samplerate, 920 920 AUDIOOUTPUT_VIDEO, 921 921 setVolume, audio_passthru); 922 922 if (!audioOutput) … … QString NuppelVideoPlayer::ReinitAudio(void) 944 944 945 945 if (audioOutput) 946 946 { 947 const AudioSettings settings( 948 audio_bits, audio_channels, audio_samplerate, 949 audio_passthru, audio_codec); 947 const AudioSettings settings(audio_bits, audio_channels, audio_codec, 948 audio_samplerate, audio_passthru); 950 949 audioOutput->Reconfigure(settings); 950 if (audio_passthru) 951 audio_channels = 2; 951 952 errMsg = audioOutput->GetError(); 952 953 if (!errMsg.isEmpty()) 953 954 audioOutput->SetStretchFactor(audio_stretchfactor); … … bool NuppelVideoPlayer::StartPlaying(bool openfile) 3887 3888 return !IsErrored(); 3888 3889 } 3889 3890 3890 void NuppelVideoPlayer::SetAudioParams(int bps, int channels, 3891 void NuppelVideoPlayer::SetAudioParams(int bps, int channels, int codec, 3891 3892 int samplerate, bool passthru) 3892 3893 { 3893 3894 audio_bits = bps; 3894 3895 audio_channels = channels; 3896 audio_codec = codec; 3895 3897 audio_samplerate = samplerate; 3896 3898 audio_passthru = passthru; 3897 3899 } 3898 3900 3899 void NuppelVideoPlayer::SetAudioCodec(void *ac)3900 {3901 audio_codec = ac;3902 }3903 3904 3901 void NuppelVideoPlayer::SetEffDsp(int dsprate) 3905 3902 { 3906 3903 if (audioOutput) … … void NuppelVideoPlayer::ToggleAdjustFill(AdjustFillMode adjustfillMode) 5323 5320 } 5324 5321 } 5325 5322 5323 bool NuppelVideoPlayer::ToggleUpmix() 5324 { 5325 if (audioOutput) 5326 return audioOutput->ToggleUpmix(); 5327 return false; 5328 } 5329 5326 5330 void NuppelVideoPlayer::Zoom(ZoomDirection direction) 5327 5331 { 5328 5332 if (videoOutput) -
mythtv/libs/libmythtv/NuppelVideoPlayer.h
diff --git a/mythtv/libs/libmythtv/NuppelVideoPlayer.h b/mythtv/libs/libmythtv/NuppelVideoPlayer.h index fc44dcd..1bf953d 100644
a b class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader 125 125 void SetAudioStretchFactor(float factor) { audio_stretchfactor = factor; } 126 126 void SetAudioOutput(AudioOutput *ao) { audioOutput = ao; } 127 127 void SetAudioInfo(const QString &main, const QString &passthru, uint rate); 128 void SetAudioParams(int bits, int channels, int samplerate, bool passthru); 128 void SetAudioParams(int bits, int channels, int codec, int samplerate, 129 bool passthru); 129 130 void SetEffDsp(int dsprate); 130 131 uint AdjustVolume(int change); 131 132 bool SetMuted(bool mute); … … class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader 179 180 // Toggle Sets 180 181 void ToggleAspectOverride(AspectOverrideMode aspectMode = kAspect_Toggle); 181 182 void ToggleAdjustFill(AdjustFillMode adjustfillMode = kAdjustFill_Toggle); 183 bool ToggleUpmix(void); 182 184 183 185 // Gets 184 186 QSize GetVideoBufferSize(void) const { return video_dim; } … … class MPUBLIC NuppelVideoPlayer : public CC608Reader, public CC708Reader 711 713 QString audio_main_device; 712 714 QString audio_passthru_device; 713 715 int audio_channels; 716 int audio_codec; 714 717 int audio_bits; 715 718 int audio_samplerate; 716 719 float audio_stretchfactor; 717 void *audio_codec;718 720 bool audio_passthru; 719 721 QMutex audio_lock; 720 722 -
mythtv/libs/libmythtv/avformatdecoder.cpp
diff --git a/mythtv/libs/libmythtv/avformatdecoder.cpp b/mythtv/libs/libmythtv/avformatdecoder.cpp index 5f276f1..b33e814 100644
a b AvFormatDecoder::AvFormatDecoder(NuppelVideoPlayer *parent, 462 462 audioSamples(NULL), 463 463 allow_ac3_passthru(false), allow_dts_passthru(false), 464 464 disable_passthru(false), max_channels(2), 465 dummy_frame(NULL),465 last_ac3_channels(0), dummy_frame(NULL), 466 466 // DVD 467 467 lastdvdtitle(-1), 468 468 decodeStillFrame(false), … … int AvFormatDecoder::ScanStreams(bool novideo) 2020 2020 // waiting on audio. 2021 2021 if (GetNVP()->HasAudioIn() && tracks[kTrackTypeAudio].empty()) 2022 2022 { 2023 GetNVP()->SetAudioParams(-1, -1, -1, false /* AC3/DTS pass-through */);2023 GetNVP()->SetAudioParams(-1, -1, CODEC_ID_NONE, -1, false /* AC3/DTS pass-through */); 2024 2024 GetNVP()->ReinitAudio(); 2025 2025 if (ringBuffer && ringBuffer->isDVD()) 2026 2026 audioIn = AudioInfo(); … … int AvFormatDecoder::AutoSelectAudioTrack(void) 3055 3055 { 3056 3056 int idx = atracks[i].av_stream_index; 3057 3057 AVCodecContext *codec_ctx = ic->streams[idx]->codec; 3058 bool do_ac3_passthru = (allow_ac3_passthru && !transcoding &&3059 !disable_passthru &&3060 (codec_ctx->codec_id == CODEC_ID_AC3));3061 bool do_dts_passthru = (allow_dts_passthru && !transcoding &&3062 !disable_passthru &&3063 (codec_ctx->codec_id == CODEC_ID_DTS));3064 3058 AudioInfo item(codec_ctx->codec_id, 3065 3059 codec_ctx->sample_rate, codec_ctx->channels, 3066 do_ac3_passthru || do_dts_passthru);3060 DoPassThrough(codec_ctx)); 3067 3061 VERBOSE(VB_AUDIO, LOC + " * " + item.toString()); 3068 3062 } 3069 3063 #endif … … static void extract_mono_channel(uint channel, AudioInfo *audioInfo, 3197 3191 bool AvFormatDecoder::GetFrame(int onlyvideo) 3198 3192 { 3199 3193 AVPacket *pkt = NULL; 3194 AC3HeaderInfo hdr; 3200 3195 int len; 3201 3196 unsigned char *ptr; 3202 3197 int data_size = 0; … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3393 3388 pts = 0; 3394 3389 3395 3390 AVStream *curstream = ic->streams[pkt->stream_index]; 3391 AVCodecContext *ctx = curstream->codec; 3396 3392 3397 3393 if (pkt->dts != (int64_t)AV_NOPTS_VALUE) 3398 3394 pts = (long long)(av_q2d(curstream->time_base) * pkt->dts * 1000); 3399 3395 3400 if (ringBuffer->isDVD() && 3401 c urstream->codec->codec_type == CODEC_TYPE_VIDEO)3396 if (ringBuffer->isDVD() && 3397 ctx->codec_type == CODEC_TYPE_VIDEO) 3402 3398 { 3403 3399 MpegPreProcessPkt(curstream, pkt); 3404 3400 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3420 3416 3421 3417 if (!d->HasMPEG2Dec()) 3422 3418 { 3423 int current_width = c urstream->codec->width;3419 int current_width = ctx->width; 3424 3420 int video_width = GetNVP()->GetVideoSize().width(); 3425 3421 if (dvd_xvmc_enabled && GetNVP() && GetNVP()->getVideoOutput()) 3426 3422 { … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3460 3456 } 3461 3457 3462 3458 if (storevideoframes && 3463 c urstream->codec->codec_type == CODEC_TYPE_VIDEO)3459 ctx->codec_type == CODEC_TYPE_VIDEO) 3464 3460 { 3465 3461 av_dup_packet(pkt); 3466 3462 storedPackets.append(pkt); … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3468 3464 continue; 3469 3465 } 3470 3466 3471 if (len > 0 && c urstream->codec->codec_type == CODEC_TYPE_VIDEO &&3467 if (len > 0 && ctx->codec_type == CODEC_TYPE_VIDEO && 3472 3468 pkt->stream_index == selectedVideoIndex) 3473 3469 { 3474 AVCodecContext *context = curstream->codec;3475 3470 3476 if (CODEC_IS_FFMPEG_MPEG(c ontext->codec_id))3471 if (CODEC_IS_FFMPEG_MPEG(ctx->codec_id)) 3477 3472 { 3478 3473 if (!ringBuffer->isDVD()) 3479 3474 MpegPreProcessPkt(curstream, pkt); 3480 3475 } 3481 else if (CODEC_IS_H264(c ontext->codec_id))3476 else if (CODEC_IS_H264(ctx->codec_id)) 3482 3477 { 3483 3478 H264PreProcessPkt(curstream, pkt); 3484 3479 } … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3523 3518 } 3524 3519 3525 3520 if (len > 0 && 3526 c urstream->codec->codec_type == CODEC_TYPE_DATA &&3527 c urstream->codec->codec_id == CODEC_ID_MPEG2VBI)3521 ctx->codec_type == CODEC_TYPE_DATA && 3522 ctx->codec_id == CODEC_ID_MPEG2VBI) 3528 3523 { 3529 3524 ProcessVBIDataPacket(curstream, pkt); 3530 3525 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3533 3528 } 3534 3529 3535 3530 if (len > 0 && 3536 c urstream->codec->codec_type == CODEC_TYPE_DATA &&3537 c urstream->codec->codec_id == CODEC_ID_DVB_VBI)3531 ctx->codec_type == CODEC_TYPE_DATA && 3532 ctx->codec_id == CODEC_ID_DVB_VBI) 3538 3533 { 3539 3534 ProcessDVBDataPacket(curstream, pkt); 3540 3535 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3544 3539 3545 3540 #ifdef USING_MHEG 3546 3541 if (len > 0 && 3547 c urstream->codec->codec_type == CODEC_TYPE_DATA &&3548 c urstream->codec->codec_id == CODEC_ID_DSMCC_B)3542 ctx->codec_type == CODEC_TYPE_DATA && 3543 ctx->codec_id == CODEC_ID_DSMCC_B) 3549 3544 { 3550 3545 ProcessDSMCCPacket(curstream, pkt); 3551 3546 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3566 3561 #endif // USING_MHEG 3567 3562 3568 3563 // we don't care about other data streams 3569 if (c urstream->codec->codec_type == CODEC_TYPE_DATA)3564 if (ctx->codec_type == CODEC_TYPE_DATA) 3570 3565 { 3571 3566 av_free_packet(pkt); 3572 3567 continue; 3573 3568 } 3574 3569 3575 if (!c urstream->codec->codec)3570 if (!ctx->codec) 3576 3571 { 3577 3572 VERBOSE(VB_PLAYBACK, LOC + 3578 3573 QString("No codec for stream index %1, type(%2) id(%3:%4)") 3579 3574 .arg(pkt->stream_index) 3580 .arg(codec_type_string(c urstream->codec->codec_type))3581 .arg(codec_id_string(c urstream->codec->codec_id))3582 .arg(c urstream->codec->codec_id));3575 .arg(codec_type_string(ctx->codec_type)) 3576 .arg(codec_id_string(ctx->codec_id)) 3577 .arg(ctx->codec_id)); 3583 3578 av_free_packet(pkt); 3584 3579 continue; 3585 3580 } … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3588 3583 have_err = false; 3589 3584 3590 3585 avcodeclock.lock(); 3591 int ctype = c urstream->codec->codec_type;3586 int ctype = ctx->codec_type; 3592 3587 int audIdx = selectedTrack[kTrackTypeAudio].av_stream_index; 3593 3588 int audSubIdx = selectedTrack[kTrackTypeAudio].av_substream_index; 3594 3589 int subIdx = selectedTrack[kTrackTypeSubtitle].av_stream_index; … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3613 3608 3614 3609 // detect switches between stereo and dual languages 3615 3610 bool wasDual = audSubIdx != -1; 3616 bool isDual = c urstream->codec->avcodec_dual_language;3611 bool isDual = ctx->avcodec_dual_language; 3617 3612 if ((wasDual && !isDual) || (!wasDual && isDual)) 3618 3613 { 3619 3614 SetupAudioStreamSubIndexes(audIdx); 3620 3615 reselectAudioTrack = true; 3621 3616 } 3622 3617 3623 bool do_ac3_passthru =3624 (allow_ac3_passthru && !transcoding &&3625 (curstream->codec->codec_id == CODEC_ID_AC3));3626 bool do_dts_passthru =3627 (allow_dts_passthru && !transcoding &&3628 (curstream->codec->codec_id == CODEC_ID_DTS));3629 bool using_passthru = do_ac3_passthru || do_dts_passthru;3630 3631 3618 // detect channels on streams that need 3632 3619 // to be decoded before we can know this 3633 3620 bool already_decoded = false; 3634 if (!c urstream->codec->channels)3621 if (!ctx->channels) 3635 3622 { 3636 3623 QMutexLocker locker(&avcodeclock); 3637 3624 VERBOSE(VB_IMPORTANT, LOC + 3638 3625 QString("Setting channels to %1") 3639 3626 .arg(audioOut.channels)); 3640 3627 3641 if ( using_passthru)3628 if (DoPassThrough(ctx)) 3642 3629 { 3643 3630 // for passthru let it select the max number 3644 3631 // of channels 3645 c urstream->codec->channels = 0;3646 c urstream->codec->request_channels = 0;3632 ctx->channels = 0; 3633 ctx->request_channels = 0; 3647 3634 } 3648 3635 else 3649 3636 { 3650 c urstream->codec->channels = audioOut.channels;3651 c urstream->codec->request_channels =3637 ctx->channels = audioOut.channels; 3638 ctx->request_channels = 3652 3639 audioOut.channels; 3653 3640 } 3641 3654 3642 data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; 3655 ret = avcodec_decode_audio2(curstream->codec, 3656 audioSamples, &data_size, 3657 ptr, len); 3643 ret = avcodec_decode_audio2(ctx, audioSamples, 3644 &data_size, ptr, len); 3658 3645 already_decoded = true; 3659 3646 3660 reselectAudioTrack |= curstream->codec->channels; 3647 reselectAudioTrack |= ctx->channels; 3648 } 3649 3650 if (ctx->codec_id == CODEC_ID_AC3) 3651 { 3652 GetBitContext gbc; 3653 init_get_bits(&gbc, ptr, len * 8); 3654 if (!ff_ac3_parse_header(&gbc, &hdr)) 3655 { 3656 if (hdr.channels != last_ac3_channels) 3657 { 3658 last_ac3_channels = ctx->channels = hdr.channels; 3659 SetupAudioStream(); 3660 } 3661 } 3661 3662 } 3662 3663 3663 3664 if (reselectAudioTrack) … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3673 3674 .av_stream_index; 3674 3675 audSubIdx = selectedTrack[kTrackTypeAudio] 3675 3676 .av_substream_index; 3677 ctx = curstream->codec; 3676 3678 } 3677 3679 3678 3680 if ((onlyvideo > 0) || (pkt->stream_index != audIdx)) … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3704 3706 if (audioOut.do_passthru) 3705 3707 { 3706 3708 data_size = pkt->size; 3707 bool dts = CODEC_ID_DTS == c urstream->codec->codec_id;3709 bool dts = CODEC_ID_DTS == ctx->codec_id; 3708 3710 ret = encode_frame(dts, ptr, len, 3709 3711 audioSamples, data_size); 3710 3712 } 3711 3713 else 3712 3714 { 3713 AVCodecContext *ctx = curstream->codec;3714 3715 3715 if ((ctx->channels == 0) || 3716 3716 (ctx->channels > audioOut.channels)) 3717 3717 { … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3720 3720 3721 3721 if (!already_decoded) 3722 3722 { 3723 curstream->codec->request_channels =3724 audioOut.channels;3725 3723 data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; 3726 ret = avcodec_decode_audio2(ctx, audioSamples, 3724 ctx->request_channels = audioOut.channels; 3725 ret = avcodec_decode_audio2(ctx, audioSamples, 3727 3726 &data_size, ptr, len); 3728 3727 } 3729 3728 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3739 3738 audIdx = -1; 3740 3739 AutoSelectAudioTrack(); 3741 3740 data_size = 0; 3741 ctx = curstream->codec; 3742 3742 } 3743 3743 } 3744 3744 avcodeclock.unlock(); … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3762 3762 3763 3763 // calc for next frame 3764 3764 lastapts += (long long)((double)(data_size * 1000) / 3765 (curstream->codec->channels * 2) / 3766 curstream->codec->sample_rate); 3765 (ctx->channels * 2) / ctx->sample_rate); 3767 3766 3768 3767 VERBOSE(VB_PLAYBACK+VB_TIMESTAMP, 3769 3768 LOC + QString("audio timecode %1 %2 %3 %4") … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3831 3830 continue; 3832 3831 } 3833 3832 3834 AVCodecContext *context = curstream->codec;3835 3833 AVFrame mpa_pic; 3836 3834 bzero(&mpa_pic, sizeof(AVFrame)); 3837 3835 … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3846 3844 // HACK 3847 3845 while (!gotpicture && count < 5) 3848 3846 { 3849 ret = d->DecodeMPEG2Video(c ontext, &mpa_pic,3847 ret = d->DecodeMPEG2Video(ctx, &mpa_pic, 3850 3848 &gotpicture, ptr, len); 3851 3849 count++; 3852 3850 } 3853 3851 } 3854 3852 else 3855 3853 { 3856 ret = d->DecodeMPEG2Video(c ontext, &mpa_pic,3854 ret = d->DecodeMPEG2Video(ctx, &mpa_pic, 3857 3855 &gotpicture, ptr, len); 3858 3856 } 3859 3857 } 3860 3858 else 3861 3859 { 3862 ret = avcodec_decode_video(c ontext, &mpa_pic,3860 ret = avcodec_decode_video(ctx, &mpa_pic, 3863 3861 &gotpicture, ptr, len); 3864 3862 // Reparse it to not drop the DVD still frame 3865 3863 if (decodeStillFrame) 3866 ret = avcodec_decode_video(c ontext, &mpa_pic,3864 ret = avcodec_decode_video(ctx, &mpa_pic, 3867 3865 &gotpicture, ptr, len); 3868 3866 } 3869 3867 avcodeclock.unlock(); … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3931 3929 myth_sws_img_convert( 3932 3930 &tmppicture, PIX_FMT_YUV420P, 3933 3931 (AVPicture *)&mpa_pic, 3934 c ontext->pix_fmt,3935 c ontext->width,3936 c ontext->height);3932 ctx->pix_fmt, 3933 ctx->width, 3934 ctx->height); 3937 3935 3938 3936 if (xf) 3939 3937 { … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3956 3954 (temppts + 10000 > lastvpts || temppts < 0)) 3957 3955 { 3958 3956 temppts = lastvpts; 3959 temppts += (long long)(1000 * av_q2d(c ontext->time_base));3957 temppts += (long long)(1000 * av_q2d(ctx->time_base)); 3960 3958 // MPEG2 frames can be repeated, update pts accordingly 3961 3959 temppts += (long long)(mpa_pic.repeat_pict * 500 3962 * av_q2d(c urstream->codec->time_base));3960 * av_q2d(ctx->time_base)); 3963 3961 } 3964 3962 3965 3963 VERBOSE(VB_PLAYBACK+VB_TIMESTAMP, LOC + … … bool AvFormatDecoder::GetFrame(int onlyvideo) 3995 3993 picframe->frameNumber = framesPlayed; 3996 3994 GetNVP()->ReleaseNextVideoFrame(picframe, temppts); 3997 3995 if (d->HasMPEG2Dec() && mpa_pic.data[3]) 3998 c ontext->release_buffer(context, &mpa_pic);3996 ctx->release_buffer(ctx, &mpa_pic); 3999 3997 4000 3998 decoded_video_frame = picframe; 4001 3999 gotvideo = 1; … … bool AvFormatDecoder::GetFrame(int onlyvideo) 4051 4049 } 4052 4050 default: 4053 4051 { 4054 AVCodecContext *enc = curstream->codec;4055 4052 VERBOSE(VB_IMPORTANT, LOC_ERR + 4056 4053 QString("Decoding - id(%1) type(%2)") 4057 .arg(codec_id_string( enc->codec_id))4058 .arg(codec_type_string( enc->codec_type)));4054 .arg(codec_id_string(ctx->codec_id)) 4055 .arg(codec_type_string(ctx->codec_type))); 4059 4056 have_err = true; 4060 4057 break; 4061 4058 } … … void AvFormatDecoder::SetDisablePassThrough(bool disable) 4185 4182 } 4186 4183 } 4187 4184 4185 bool AvFormatDecoder::DoPassThrough(const AVCodecContext *ctx) 4186 { 4187 bool passthru = false; 4188 4189 if (ctx->codec_id == CODEC_ID_AC3) 4190 passthru = allow_ac3_passthru && 4191 ctx->channels >= (int)max_channels; 4192 else if (ctx->codec_id == CODEC_ID_DTS) 4193 passthru = allow_dts_passthru; 4194 4195 passthru &= !transcoding && !disable_passthru; 4196 // Don't know any cards that support spdif clocked at < 44100 4197 // Some US cable transmissions have 2ch 32k AC-3 streams 4198 passthru &= ctx->sample_rate >= 44100; 4199 4200 return passthru; 4201 } 4202 4188 4203 /** \fn AvFormatDecoder::SetupAudioStream(void) 4189 4204 * \brief Reinitializes audio if it needs to be reinitialized. 4190 4205 * … … bool AvFormatDecoder::SetupAudioStream(void) 4198 4213 AVStream *curstream = NULL; 4199 4214 AVCodecContext *codec_ctx = NULL; 4200 4215 AudioInfo old_in = audioIn; 4201 AudioInfo old_out = audioOut;4202 4216 bool using_passthru = false; 4203 4217 4204 4218 if ((currentTrack[kTrackTypeAudio] >= 0) && … … bool AvFormatDecoder::SetupAudioStream(void) 4212 4226 codec_ctx = curstream->codec; 4213 4227 if (codec_ctx) 4214 4228 { 4215 bool do_ac3_passthru = (allow_ac3_passthru && !transcoding && 4216 (codec_ctx->codec_id == CODEC_ID_AC3)); 4217 bool do_dts_passthru = (allow_dts_passthru && !transcoding && 4218 (codec_ctx->codec_id == CODEC_ID_DTS)); 4219 using_passthru = do_ac3_passthru || do_dts_passthru; 4220 info = AudioInfo(codec_ctx->codec_id, 4221 codec_ctx->sample_rate, codec_ctx->channels, 4222 using_passthru && !disable_passthru); 4229 using_passthru = DoPassThrough(codec_ctx); 4230 info = AudioInfo(codec_ctx->codec_id, codec_ctx->sample_rate, 4231 codec_ctx->channels, using_passthru); 4223 4232 } 4224 4233 } 4225 4234 … … bool AvFormatDecoder::SetupAudioStream(void) 4236 4245 QString("audio track #%1").arg(currentTrack[kTrackTypeAudio]+1)); 4237 4246 4238 4247 audioOut = audioIn = info; 4239 AudioInfo tmpAudioOut = audioOut;4240 4248 4241 // A passthru stream looks like a 48KHz 2ch (@ 16bit) to the sound card 4242 if (using_passthru && !disable_passthru) 4249 if (!using_passthru && audioOut.channels > (int)max_channels) 4243 4250 { 4244 tmpAudioOut.channels = 2; 4245 tmpAudioOut.sample_rate = 48000; 4246 tmpAudioOut.sample_size = 4; 4247 } 4248 4249 if (audioOut.channels > (int) max_channels) 4250 { 4251 audioOut.channels = (int) max_channels; 4251 audioOut.channels = (int)max_channels; 4252 4252 audioOut.sample_size = audioOut.channels * 2; 4253 codec_ctx->channels 4253 codec_ctx->channels = audioOut.channels; 4254 4254 } 4255 4255 4256 if (!using_passthru)4257 tmpAudioOut = audioOut;4258 4259 4256 VERBOSE(VB_AUDIO, LOC + "Audio format changed " + 4260 QString("%1%2\n\t\t\tfrom %3 ; %4\n\t\t\tto %5 ; %6") 4261 .arg((using_passthru) ? "digital passthrough " : "") 4262 .arg((using_passthru) ? tmpAudioOut.toString() : QString("")) 4263 .arg(old_in.toString()).arg(old_out.toString()) 4264 .arg(audioIn.toString()).arg(audioOut.toString())); 4257 QString("\n\t\t\tfrom %1 to %2") 4258 .arg(old_in.toString()).arg(audioOut.toString())); 4265 4259 4266 if ( tmpAudioOut.sample_rate > 0)4267 GetNVP()->SetEffDsp( tmpAudioOut.sample_rate * 100);4260 if (audioOut.sample_rate > 0) 4261 GetNVP()->SetEffDsp(audioOut.sample_rate * 100); 4268 4262 4269 GetNVP()->SetAudioParams(tmpAudioOut.bps(), tmpAudioOut.channels, 4270 tmpAudioOut.sample_rate, audioIn.do_passthru); 4263 GetNVP()->SetAudioParams(audioOut.bps(), audioOut.channels, 4264 audioOut.codec_id, audioOut.sample_rate, 4265 audioOut.do_passthru); 4271 4266 4272 // allow the audio stuff to reencode4273 GetNVP()->SetAudioCodec((using_passthru) ? codec_ctx : NULL);4274 4267 GetNVP()->ReinitAudio(); 4275 4268 4276 4269 return true; -
mythtv/libs/libmythtv/avformatdecoder.h
diff --git a/mythtv/libs/libmythtv/avformatdecoder.h b/mythtv/libs/libmythtv/avformatdecoder.h index 9c05180..4f799a3 100644
a b class AvFormatDecoder : public DecoderBase 188 188 189 189 void SeekReset(long long, uint skipFrames, bool doFlush, bool discardFrames); 190 190 191 bool DoPassThrough(const AVCodecContext *ctx); 191 192 bool SetupAudioStream(void); 192 193 void SetupAudioStreamSubIndexes(int streamIndex); 193 194 void RemoveAudioStreams(); … … class AvFormatDecoder : public DecoderBase 260 261 bool allow_dts_passthru; 261 262 bool disable_passthru; 262 263 uint max_channels; 264 uint last_ac3_channels; 263 265 264 266 VideoFrame *dummy_frame; 265 267 -
mythtv/libs/libmythtv/nuppeldecoder.cpp
diff --git a/mythtv/libs/libmythtv/nuppeldecoder.cpp b/mythtv/libs/libmythtv/nuppeldecoder.cpp index 88d2fbb..010a458 100644
a b int NuppelDecoder::OpenFile(RingBuffer *rbuffer, bool novideo, 495 495 #endif 496 496 GetNVP()->SetAudioParams(extradata.audio_bits_per_sample, 497 497 extradata.audio_channels, 498 CODEC_ID_NONE, 498 499 extradata.audio_sample_rate, 499 500 false /* AC3/DTS pass through */); 500 501 GetNVP()->ReinitAudio(); -
mythtv/libs/libmythtv/tv_play.cpp
diff --git a/mythtv/libs/libmythtv/tv_play.cpp b/mythtv/libs/libmythtv/tv_play.cpp index d469fed..89a2dc6 100644
a b void TV::InitKeys(void) 491 491 REG_KEY("TV Playback", "VOLUMEDOWN", "Volume down", "[,{,F10,Volume Down"); 492 492 REG_KEY("TV Playback", "VOLUMEUP", "Volume up", "],},F11,Volume Up"); 493 493 REG_KEY("TV Playback", "MUTE", "Mute", "|,\\,F9,Volume Mute"); 494 REG_KEY("TV Playback", "TOGGLEUPMIX", "Toggle upmixer", "Ctrl+U"); 494 495 REG_KEY("TV Playback", "TOGGLEPIPMODE", "Toggle Picture-in-Picture view", 495 496 "V"); 496 497 REG_KEY("TV Playback", "TOGGLEPBPMODE", "Toggle Picture-by-Picture view", … … void TV::InitKeys(void) 630 631 Teletext F2,F3,F4,F5,F6,F7,F8 631 632 ITV F2,F3,F4,F5,F6,F7,F12 632 633 633 Playback: Ctrl-B,Ctrl-G,Ctrl-Y 634 Playback: Ctrl-B,Ctrl-G,Ctrl-Y,Ctrl-U 634 635 */ 635 636 } 636 637 … … bool TV::ToggleHandleAction(PlayerContext *ctx, 4343 4344 DoTogglePictureAttribute(ctx, kAdjustingPicture_Playback); 4344 4345 else if (has_action("TOGGLESTRETCH", actions)) 4345 4346 ToggleTimeStretch(ctx); 4347 else if (has_action("TOGGLEUPMIX", actions)) 4348 ToggleUpmix(ctx); 4346 4349 else if (has_action("TOGGLESLEEP", actions)) 4347 4350 ToggleSleepTimer(ctx); 4348 4351 else if (has_action("TOGGLERECORD", actions) && islivetv) … … void TV::ChangeTimeStretch(PlayerContext *ctx, int dir, bool allowEdit) 8036 8039 SetSpeedChangeTimer(0, __LINE__); 8037 8040 } 8038 8041 8042 void TV::ToggleUpmix(PlayerContext *ctx) 8043 { 8044 if (!ctx->nvp || !ctx->nvp->HasAudioOut()) 8045 return; 8046 QString text; 8047 if (ctx->nvp->ToggleUpmix()) 8048 text = tr("Upmixer On"); 8049 else 8050 text = tr("Upmixer Off"); 8051 8052 if (ctx->nvp->GetOSD() && !browsemode) 8053 ctx->nvp->GetOSD()->SetSettingsText(text, 5); 8054 } 8055 8039 8056 // dir in 10ms jumps 8040 8057 void TV::ChangeAudioSync(PlayerContext *ctx, int dir, bool allowEdit) 8041 8058 { … … void TV::TreeMenuSelected(OSDListTreeItemSelectedEvent *e) 9669 9686 SetManualZoom(actx, true, tr("Zoom Mode ON")); 9670 9687 else if (action == "TOGGLESTRETCH") 9671 9688 ToggleTimeStretch(actx); 9689 else if (action == "TOGGLEUPMIX") 9690 ToggleUpmix(actx); 9672 9691 else if (action.left(13) == "ADJUSTSTRETCH") 9673 9692 { 9674 9693 bool floatRead; … … void TV::FillOSDTreeMenu( 10022 10041 10023 10042 if (category == "AUDIOSYNC") 10024 10043 new OSDGenericTree(treeMenu, tr("Adjust Audio Sync"), "TOGGLEAUDIOSYNC"); 10044 else if (category == "TOGGLEUPMIX") 10045 new OSDGenericTree(treeMenu, tr("Toggle Upmixer"), "TOGGLEUPMIX"); 10025 10046 else if (category == "TIMESTRETCH") 10026 10047 FillMenuTimeStretch(ctx, treeMenu); 10027 10048 else if (category == "VIDEOSCAN") -
mythtv/libs/libmythtv/tv_play.h
diff --git a/mythtv/libs/libmythtv/tv_play.h b/mythtv/libs/libmythtv/tv_play.h index d50e0d4..4895219 100644
a b class MPUBLIC TV : public QThread 419 419 ARBSEEK_FORWARD, 420 420 ARBSEEK_END 421 421 }; 422 422 423 void DoArbSeek(PlayerContext*, ArbSeekWhence whence); 423 424 void NormalSpeed(PlayerContext*); 424 425 void ChangeSpeed(PlayerContext*, int direction); … … class MPUBLIC TV : public QThread 427 428 bool TimeStretchHandleAction(PlayerContext*, 428 429 const QStringList &actions); 429 430 431 void ToggleUpmix(PlayerContext*); 430 432 void ChangeAudioSync(PlayerContext*, int dir, bool allowEdit = true); 431 433 bool AudioSyncHandleAction(PlayerContext*, const QStringList &actions); 432 434 -
mythtv/libs/libmythtv/tvosdmenuentry.cpp
diff --git a/mythtv/libs/libmythtv/tvosdmenuentry.cpp b/mythtv/libs/libmythtv/tvosdmenuentry.cpp index 994dcb2..ef714ea 100644
a b void TVOSDMenuEntryList::InitDefaultEntries(void) 232 232 curMenuEntries.append( 233 233 new TVOSDMenuEntry("AUDIOSYNC", 1, 1, 1, 1 , "Audio Sync")); 234 234 curMenuEntries.append( 235 new TVOSDMenuEntry("TOGGLEUPMIX", 1, 1, 1, 1, "Toggle Upmixer")); 236 curMenuEntries.append( 235 237 new TVOSDMenuEntry("TIMESTRETCH", 1, 1, 1, 1, "Time Stretch")); 236 238 curMenuEntries.append( 237 239 new TVOSDMenuEntry("VIDEOSCAN", 1, 1, 1, 1, "Video Scan")); -
mythtv/programs/mythfrontend/globalsettings.cpp
diff --git a/mythtv/programs/mythfrontend/globalsettings.cpp b/mythtv/programs/mythfrontend/globalsettings.cpp index 2f166ff..e6fcc3f 100644
a b static HostComboBox *AudioUpmixType() 119 119 return gc; 120 120 } 121 121 122 static HostComboBox *SRCQuality() 123 { 124 HostComboBox *gc = new HostComboBox("SRCQuality", false); 125 gc->setLabel(QObject::tr("Sample Rate Conversion")); 126 gc->addSelection(QObject::tr("Best"), "3", true); // default 127 gc->addSelection(QObject::tr("Medium"), "2"); 128 gc->addSelection(QObject::tr("Fastest"), "1"); 129 gc->setHelpText( 130 QObject::tr( 131 "Set the quality of audio sample rate conversion. " 132 "This only affects non 48000Hz PCM audio. " 133 "All three options offer a worst-case SNR of 97dB. " 134 "'Best' at a bandwidth of 97%. " 135 "'Medium' at a bandwidth of 90%. " 136 "'Fastest' at a bandwidth of 80%. " 137 ) 138 ); 139 return gc; 140 } 141 142 122 143 static HostComboBox *PassThroughOutputDevice() 123 144 { 124 145 HostComboBox *gc = new HostComboBox("PassThruOutputDevice", true); … … class AudioSystemSettingsGroup : public VerticalConfigurationGroup 3435 3456 3436 3457 addChild(MaxAudioChannels()); 3437 3458 addChild(AudioUpmixType()); 3459 addChild(SRCQuality()); 3438 3460 3439 3461 // General boolean settings 3440 3462 addChild(AC3PassThrough()); -
mythtv/programs/mythtranscode/transcode.cpp
diff --git a/mythtv/programs/mythtranscode/transcode.cpp b/mythtv/programs/mythtranscode/transcode.cpp index 1120976..9ec39a1 100644
a b class AudioReencodeBuffer : public AudioOutput 49 49 AudioReencodeBuffer(int audio_bits, int audio_channels) 50 50 { 51 51 Reset(); 52 const AudioSettings settings(audio_bits, audio_channels, 0, false);52 const AudioSettings settings(audio_bits, audio_channels, 0, 0, false); 53 53 Reconfigure(settings); 54 54 bufsize = 512000; 55 55 audiobuffer = new unsigned char[bufsize]; … … class AudioReencodeBuffer : public AudioOutput 222 222 // Do nothing 223 223 return kMuteOff; 224 224 } 225 virtual bool ToggleUpmix(void) 226 { 227 // Do nothing 228 return false; 229 } 225 230 226 231 // These are pure virtual in AudioOutput, but we don't need them here 227 232 virtual void bufferOutputData(bool){ return; }