From 0d29dda4c1e2ea5f4562df282c4ddfa8d6251fb2 Mon Sep 17 00:00:00 2001 From: snt Date: Fri, 24 May 2024 01:49:21 +0200 Subject: [PATCH] =?UTF-8?q?quita=20el=20glitch=20al=20resproducir=20despu?= =?UTF-8?q?=C3=A9s=20de=20cargar=20un=20fichero.=20varias=20optimizaciones?= =?UTF-8?q?=20y=20comprobaciones?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- docs/changelog.txt | 7 ++--- docs/roadmap.txt | 38 ++++++++++---------------- src/audiolayerwidget.cpp | 11 ++++---- src/libremediaserver-audio.cpp | 11 ++++---- src/ma_writer_node.c | 10 +++---- src/ma_writer_node.h | 4 ++- src/miniaudioengine.cpp | 50 ++++++++++++++++++---------------- src/miniaudioengine.h | 6 ++-- 8 files changed, 66 insertions(+), 71 deletions(-) diff --git a/docs/changelog.txt b/docs/changelog.txt index 9a2f61b..5cd2ca3 100644 --- a/docs/changelog.txt +++ b/docs/changelog.txt @@ -1,10 +1,10 @@ ******************************************************************************* -Libre Media Server Audio - An Open source Media Server for arts and performing. +LibreMediaServer Audio - An open source media server for arts and performing. (c) Criptomart - Santiago Noreña 2012-2024 https://git.criptomart.net/libremediaserver ******************************************************************************* -Libre Media Server ChangeLog +LibreMediaServer Changelog v 0.2.0 Antígona (26/05/2024) + Change audio engine to miniaudio because is imposible pan in SFML and it has not access to low API and audio processing. @@ -20,9 +20,7 @@ v 0.2.0 Antígona (26/05/2024) + OlaThread send double channels (volume, entry point, load media) only once for each dmx frame buffer. + Terminal mode without graphical interface. All audio methods has been refactorized out of QWidget world. + Compilation without GUI (-DNOGUI). -+ New Status "Iddle" in playbacks if is not loaded. + New DMX personality version, better sort for audio needs (first load media, set vol, pan, etc, last playback order); -+ Refresh layer values when it loads a new sound file. + No QtSignals for sending data, better performance about 20% in my machine. Now, libremediaserver only updates values in AudioWidget, ui refresh is doing with a timer in audiowidget, so there is not problems between graphical and ola thread (the callback). Signals are used only from Ui to libreMediaServer to notify user interactions and Qtimers. + Load media files from ui clicking in the media labels. + New Play Modes: @@ -31,6 +29,7 @@ v 0.2.0 Antígona (26/05/2024) - Play all medias in one folder randomly. + Multi audio devices output. + Vumeter for each layer ++ Show device name on Ui and ouput bus slider. v 0.1.3 Leúcade (19/04/2024) + Ubuntu 22.04 jammy. diff --git a/docs/roadmap.txt b/docs/roadmap.txt index bc06163..bc786a9 100644 --- a/docs/roadmap.txt +++ b/docs/roadmap.txt @@ -6,38 +6,32 @@ https://git.criptomart.net/libremediaserver Libre Media Server Roadmap -v 0.2.x -- skin, UI/UX +v 0.3.0 +- Ui/Ux: skin, style. +- Ui/Ux; Keyboards strokes. +- Ui/Ux: Dar la opción clickeando en el widget de tiempo de poner una cuenta atrás en vez de hacia delante. +- Ui/Ux: seek cursor playback - live input. -- insertar/bypass/eliminar audio procesadores sin reiniciar por capa y master. (compresor, equs). -- FX en capas master para que se puedan usar como envíos de auxiliar. -- Enroutado de masters en otros masters (retorno de efectos). - -v 0.2.2 -- Use sACN directly. +- remove ola and use sACN directly. + la instalación de OLA es mediante compilación, el repo de paquetes no está actualizado, nada user-friendly. + hay que empaquetar OLA, incluirlo en el binario, o implementar sACN y linkarlo estáticamente. + https://github.com/ETCLabs/sACN - Qt6. -- Audio processing (eq, rev, compresor, ...) by master and layer. -- CIPT/MSex, send icons play/pause/stop. +- CIPT/MSex. - Rasp build. - Octopus Sound Card support (6 outputs - 8 inputs). - -v 0.2.1 -- mute/panic on layer. - Master Bus Layer: - - each layer will have one "Gain" prefader that acts in source, "Vol" in v 1.3. - - each layer will have one volume dmx channel for each bus layer. One aux "Send" prefader. - mute/panic. - fader + value - - pan. - - magicq .hed + - pan + - magicq personality .hed - audio device linked, outputs will be redirected there. - dmx address + universe settings. -- Rose noise and sine generator in menu to test system. -- Ui/Ux; Keyboards strokes. -- Ui/Ux: Dar la opción clickeando en el widget de tiempo de poner una cuenta atrás en vez de hacia delante. + - compresor/limiter. +- Layer: + - audio procesadores (compresor, reveb, delay). + - mute/panic. +- Rose noise and sine generator. - Logs, verbosity, timestamp. - New play mode without pitch control, it saves resources. MA_SOUND_FLAG_NO_PITCH - SettingsDialog. @@ -45,8 +39,4 @@ v 0.2.1 - ¿Exit Point? is it needed? - Hardening: check return errors, try/catch exceptions, i'm too happy.... - Tests: errors on wrong conf file. -- Ui/Ux: seek cursor playback - ampliar writer para recibir un número n de entradas y escribirlas cada una en un buffer - -v0.2.0: -- mostrad información de envíos y dispositivos en ui diff --git a/src/audiolayerwidget.cpp b/src/audiolayerwidget.cpp index 7fc467b..b1eb8b7 100644 --- a/src/audiolayerwidget.cpp +++ b/src/audiolayerwidget.cpp @@ -198,6 +198,7 @@ void AudioLayerWidget::openMediaDialog() fileNames = dialog.selectedFiles(); emit uiLoadMedia(m_layer, fileNames.at(0)); this->setMediaFile(fileNames.at(0)); + this->setPlaybackStatus(Status::Stopped); } // from DMX signals @@ -267,18 +268,18 @@ void AudioLayerWidget::setCurrentTime(float progress) void AudioLayerWidget::setFilterParam(int channel, int value) { - if (channel <= FILTER_BANK_GAIN - FILTER_CHANNELS){ + if (channel <= FILTER_BANK_GAIN - HP_FREQ){ m_filterBank->blockSignals(true); m_filterBank->setValue(channel, value); m_filterBank->blockSignals(false); } else if (channel == SEND1 - HP_FREQ) { - m_bus1->blockSignals(false); - m_bus1->setValue((value * 256) + 255); m_bus1->blockSignals(true); + m_bus1->setValue((value * 256) + 255); + m_bus1->blockSignals(false); } else if (channel == SEND2 - HP_FREQ) { - m_bus2->blockSignals(false); - m_bus2->setValue(value * 256 + 255); m_bus2->blockSignals(true); + m_bus2->setValue(value * 256 + 255); + m_bus2->blockSignals(false); } } diff --git a/src/libremediaserver-audio.cpp b/src/libremediaserver-audio.cpp index 177eec2..dd6cedd 100644 --- a/src/libremediaserver-audio.cpp +++ b/src/libremediaserver-audio.cpp @@ -70,8 +70,7 @@ void libreMediaServerAudio::loadMedia(int layer, int folder, int file) if (strcmp(mediaFile.toLatin1().constData(), m_currentMedia[layer].toLatin1().constData()) == 0) return; if (QFile::exists(mediaFile)){ - m_mae.loadMedia(layer, mediaFile.toLatin1().data(),\ - m_dmxSettings.at(layer).audioDevice); + m_mae.loadMedia(layer, mediaFile.toLatin1().data()); m_currentMedia[layer] = mediaFile; #ifndef NOGUI if (m_ui) @@ -135,7 +134,7 @@ void libreMediaServerAudio::dmxInput(int layer, int channel, int value) } #endif } else if (channel >= HP_FREQ) { - m_mae.filterParamChanged(layer, m_dmxSettings.at(layer).audioDevice, channel, value); + m_mae.filterParamChanged(layer, channel, value); #ifndef NOGUI if (m_ui) { m_lmsUi->m_aw->filterParamChanged(layer, channel, value); @@ -252,10 +251,10 @@ void libreMediaServerAudio::uiSliderChanged(int layer, Slider s, int value) m_mae.setBypass(m_dmxSettings.at(layer).audioDevice, layer, value); break; case Slider::Bus1: - m_mae.filterParamChanged(layer, m_dmxSettings.at(layer).audioDevice, SEND1, value / 255); + m_mae.filterParamChanged(layer, SEND1, value / 255.0f); break; case Slider::Bus2: - m_mae.filterParamChanged(layer, m_dmxSettings.at(layer).audioDevice, SEND2, value / 255); + m_mae.filterParamChanged(layer, SEND2, value / 255.0f); break; } } @@ -278,7 +277,7 @@ void libreMediaServerAudio::uiLoadMedia(int layer, QString mediaFile) if (strcmp(mediaFile.toLatin1().constData(), m_currentMedia[layer].toLatin1().constData()) == 0) return; - result = m_mae.loadMedia(layer, mediaFile.toLatin1().data(), m_dmxSettings[layer].audioDevice); + result = m_mae.loadMedia(layer, mediaFile.toLatin1().data()); if (result == MA_SUCCESS) { m_currentMedia[layer] = mediaFile; m_lmsUi->m_aw->mediaLoaded(layer, mediaFile, m_mae.getDuration(layer)); diff --git a/src/ma_writer_node.c b/src/ma_writer_node.c index 2845649..5b20f66 100644 --- a/src/ma_writer_node.c +++ b/src/ma_writer_node.c @@ -38,8 +38,6 @@ static ma_node_vtable g_ma_writer_node_vtable = 2, 1, 0 -// MA_NODE_FLAG_CONTINUOUS_PROCESSING -// MA_NODE_FLAG_SILENT_OUTPUT }; MA_API ma_result ma_writer_node_init(ma_node_graph* pNodeGraph, const ma_writer_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_writer_node* pWriteNode) @@ -178,7 +176,7 @@ void ma_data_source_rb_uninit(ma_data_source_rb* pMyDataSource) * vumeter */ -MA_API ma_vumeter_node_config ma_vumeter_node_config_init(ma_uint32 channels, ma_uint32 sampleRate) +MA_API ma_vumeter_node_config ma_vumeter_node_config_init(ma_uint32 channels, ma_uint32 format, ma_uint32 sampleRate) { ma_vumeter_node_config config; @@ -186,6 +184,7 @@ MA_API ma_vumeter_node_config ma_vumeter_node_config_init(ma_uint32 channels, ma config.nodeConfig = ma_node_config_init(); config.channels = channels; config.sampleRate = sampleRate; + config.format = format; return config; } @@ -201,7 +200,7 @@ static void ma_vumeter_node_process_pcm_frames(ma_node* pNode, const float** ppF float input = fabsf(ppFramesIn[0][i]); pVumeterNode->level += pVumeterNode->alpha * (input - pVumeterNode->level); } - ma_copy_pcm_frames(ppFramesOut[0], ppFramesIn[0], *pFrameCountOut, ma_format_f32, pVumeterNode->channels); + ma_copy_pcm_frames(ppFramesOut[0], ppFramesIn[0], *pFrameCountOut, pVumeterNode->format, pVumeterNode->channels); } static ma_node_vtable g_ma_vumeter_node_vtable = @@ -238,8 +237,9 @@ MA_API ma_result ma_vumeter_node_init(ma_node_graph* pNodeGraph, const ma_vumete } pVumeterNode->sampleRate = pConfig->sampleRate; pVumeterNode->channels = pConfig->channels; + pVumeterNode->format = pConfig->format; pVumeterNode->level = 0; - pVumeterNode->TC = 10000.0f; + pVumeterNode->TC = 0.250f; pVumeterNode->alpha = 1.0 - expf( (-2.0 * M_PI) / (pVumeterNode->TC * pConfig->sampleRate)); return MA_SUCCESS; } diff --git a/src/ma_writer_node.h b/src/ma_writer_node.h index ac592c7..3e03df5 100644 --- a/src/ma_writer_node.h +++ b/src/ma_writer_node.h @@ -59,15 +59,17 @@ typedef struct ma_node_config nodeConfig; ma_uint32 channels; ma_uint32 sampleRate; + ma_uint32 format; } ma_vumeter_node_config; -MA_API ma_vumeter_node_config ma_vumeter_node_config_init(ma_uint32 channels, ma_uint32 sampleRate); +MA_API ma_vumeter_node_config ma_vumeter_node_config_init(ma_uint32 channels, ma_uint32 format, ma_uint32 sampleRate); typedef struct { ma_node_base baseNode; ma_uint32 channels; ma_uint32 sampleRate; + ma_uint32 format; float level; float TC; float alpha; diff --git a/src/miniaudioengine.cpp b/src/miniaudioengine.cpp index 292251e..1d0d5b3 100644 --- a/src/miniaudioengine.cpp +++ b/src/miniaudioengine.cpp @@ -132,7 +132,7 @@ ma_result MiniAudioEngine::createFilterBank(uint layer) cout << "ERROR " << result << ": Failed to init hi shelf filter node." << endl; return result; } - ma_vumeter_node_config vuc = ma_vumeter_node_config_init(CHANNELS, FORMAT); + ma_vumeter_node_config vuc = ma_vumeter_node_config_init(CHANNELS, FORMAT, SAMPLE_RATE); ma_vumeter_node_init(ng, &vuc, NULL, &fb->vumeter); if (result != MA_SUCCESS) { cout << "ERROR " << result << ": Failed to init vumeter node." << endl; @@ -278,7 +278,7 @@ ma_result MiniAudioEngine::startDevices() deviceConfig.dataCallback = audioDataCallback; engineConfig = ma_engine_config_init(); engineConfig.pResourceManager = &m_mae.resourceManager; - engineConfig.gainSmoothTimeInMilliseconds = SAMPLE_RATE / 25; + engineConfig.defaultVolumeSmoothTimeInPCMFrames = SAMPLE_RATE / 20; engineConfig.noAutoStart = MA_TRUE; for (uint internalId = 0; internalId < m_mae.audioDevicesQty; internalId++) { @@ -346,25 +346,28 @@ char* MiniAudioEngine::getDeviceName(uint id) } -ma_result MiniAudioEngine::loadMedia(int layer, char *file, uint audioDevice) +ma_result MiniAudioEngine::loadMedia(int layer, char *file) { ma_result result; if (m_mae.mediaLoaded[layer] == true) { + ma_sound_set_volume(&m_mae.sounds[layer], 0.0f); + ma_sound_stop(&m_mae.sounds[layer]); ma_sound_uninit(&m_mae.sounds[layer]); m_mae.mediaLoaded[layer] = false; } - result = ma_sound_init_from_file(&m_mae.engines[0], file, \ - MA_SOUND_FLAG_NO_SPATIALIZATION \ - , NULL, NULL, &m_mae.sounds[layer]); + ma_sound_config soundConfig = ma_sound_config_init(); + soundConfig = ma_sound_config_init(); + soundConfig.pFilePath = file; + soundConfig.pInitialAttachment = &m_mae.filters[layer].input; + soundConfig.initialAttachmentInputBusIndex = 0; + soundConfig.channelsIn = 0; + soundConfig.channelsOut = CHANNELS; + soundConfig.flags = MA_SOUND_FLAG_NO_SPATIALIZATION | MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT | MA_SOUND_FLAG_STREAM; //| MA_SOUND_FLAG_NO_PITCH + result = ma_sound_init_ex(&m_mae.engines[0], &soundConfig, &m_mae.sounds[layer]); if (result != MA_SUCCESS) { - cout << "Error " << result << ": Failed to load file " << file << endl; - return result; - } - result = ma_node_attach_output_bus(&m_mae.sounds[layer], 0, &m_mae.filters[layer].input, 0); - if (result != MA_SUCCESS) { - cout << "Error " << result << ": Failed to attach sound output bus " << audioDevice << endl; + cout << "Error" << result << ": Failed to load file " << file << endl; return result; } m_mae.mediaLoaded[layer] = true; @@ -435,7 +438,7 @@ void MiniAudioEngine::volChanged(int layer, int vol) db = 0; } else db = ma_volume_db_to_linear(db); - ma_sound_group_set_fade_in_milliseconds(&m_mae.sounds[layer], -1, db, FADE_TIME); + ma_sound_set_fade_in_milliseconds(&m_mae.sounds[layer], -1, db, FADE_TIME); m_mae.currentStatus[layer].vol = vol; } @@ -485,12 +488,13 @@ ma_result MiniAudioEngine::playbackChanged(int layer, Status status) ma_sound_set_stop_time_in_milliseconds(&m_mae.sounds[layer], ~(ma_uint64)0); ma_sound_set_looping(&m_mae.sounds[layer], loop); result = ma_sound_start(&m_mae.sounds[layer]); - if (m_mae.currentStatus[layer].cursor != 0) { - usleep(1000 * 50); // Avoid small glitch at start, how to flush the cached buffers in audio pipe line? - } - this->volChanged(layer, m_mae.currentStatus[layer].vol); - default: - break; + //this->volChanged(layer, m_mae.currentStatus[layer].vol); + float db = (m_mae.currentStatus[layer].vol / 771.0f) - 85.0f; + if (db <= -85.0f) { + db = 0; + } else + db = ma_volume_db_to_linear(db); + ma_sound_set_fade_in_milliseconds(&m_mae.sounds[layer], -1, db, 0); } if (result == MA_SUCCESS) m_mae.currentStatus[layer].status = status; @@ -506,7 +510,7 @@ ma_result MiniAudioEngine::seekToCursor(int layer, int cursor) return MA_DOES_NOT_EXIST; result = ma_sound_get_length_in_pcm_frames(&m_mae.sounds[layer], &end); if (result != MA_SUCCESS) { return result; } - start = (cursor * end) / 65025; + start = (cursor * end) / 65535; Status oldStatus = m_mae.currentStatus[layer].status; result = ma_sound_seek_to_pcm_frame(&m_mae.sounds[layer], start); //result = ma_data_source_set_loop_point_in_pcm_frames(&m_mae.sounds[layer], start, end); // this do nothing here, it must be done after set_looping or start? @@ -531,14 +535,14 @@ void MiniAudioEngine::refreshValues(int layer) { this->seekToCursor(layer, m_mae.currentStatus[layer].cursor); this->panChanged(layer, m_mae.currentStatus[layer].pan); - this->volChanged(layer, m_mae.currentStatus[layer].vol); this->pitchChanged(layer, m_mae.currentStatus[layer].pitch); + ma_sound_group_set_fade_in_milliseconds(&m_mae.sounds[layer], -1, 0, 0.0f); this->playbackChanged(layer, m_mae.currentStatus[layer].status); + this->volChanged(layer, m_mae.currentStatus[layer].vol); } -ma_result MiniAudioEngine::filterParamChanged(int layer, int audioDevice, int channel, int value) +ma_result MiniAudioEngine::filterParamChanged(int layer, int channel, int value) { - (void)audioDevice; ma_result result = MA_SUCCESS; filterBank *fb = &m_mae.filters[layer]; diff --git a/src/miniaudioengine.h b/src/miniaudioengine.h index b97a9e9..38107d8 100644 --- a/src/miniaudioengine.h +++ b/src/miniaudioengine.h @@ -65,7 +65,7 @@ protected: MiniAudioEngine(); void stopEngine(); bool startEngine(uint layersQty, uint* audioDevicesID, uint audioDevicesQty); - ma_result loadMedia(int layer, char *media, uint audioDevice); + ma_result loadMedia(int layer, char *media); void volChanged(int layer, int vol); void panChanged(int layer, float pan); void pitchChanged(int layer, float pitch); @@ -79,11 +79,11 @@ protected: return ma_sound_get_volume(&m_mae.sounds[layer]); }; inline bool getAtEnd(int layer) { return m_mae.sounds[layer].atEnd; } - ma_result filterParamChanged(int layer, int audioDevice, int channel, int value); + ma_result filterParamChanged(int layer, int channel, int value); bool setBypass(int audioDevice, int layer, bool bypass); inline float getLevel(int layer) { float level = ma_vumeter_node_get_level(&m_mae.filters[layer].vumeter); - return ma_volume_linear_to_db(level); + return ma_volume_linear_to_db(level) - 4.0f; }; char* getDeviceName(uint id);