#include "sdl.h" #include #include #include #include #define MIN_QUEUED_FRAMES 2 #define MAX_QUEUED_FRAMES 4 #define STOP_THE_WORLD_LIMIT 20 #define DROP_RATIO_DENOM 32 // Detecting this with SDL is quite problematic, so we'll use Qt's // multimedia framework to do so. It appears to be actually // accurate on Linux and macOS, unlike using SDL and relying // on a channel change in the format received. int SdlAudioRenderer::detectAudioConfiguration() { int preferredChannelCount = QAudioDeviceInfo::defaultOutputDevice().preferredFormat().channelCount(); SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, "Audio output device prefers %d channel configuration", preferredChannelCount); // We can better downmix 5.1 to quad than we can upmix stereo if (preferredChannelCount > 2) { return AUDIO_CONFIGURATION_51_SURROUND; } else { return AUDIO_CONFIGURATION_STEREO; } } bool SdlAudioRenderer::testAudio(int audioConfiguration) { SDL_AudioSpec want, have; SDL_AudioDeviceID dev; SDL_zero(want); want.freq = 48000; want.format = AUDIO_S16; want.samples = SAMPLES_PER_FRAME; switch (audioConfiguration) { case AUDIO_CONFIGURATION_STEREO: want.channels = 2; break; case AUDIO_CONFIGURATION_51_SURROUND: want.channels = 6; break; default: SDL_assert(false); return false; } // Test audio device for functionality dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0); if (dev == 0) { SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Audio test - Failed to open audio device: %s", SDL_GetError()); return false; } SDL_CloseAudioDevice(dev); SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, "Audio test - Successful with %d channels", want.channels); return true; } SdlAudioRenderer::SdlAudioRenderer() : m_AudioDevice(0), m_ChannelCount(0), m_PendingDrops(0), m_PendingHardDrops(0), m_SampleIndex(0), m_BaselinePendingData(0) { SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO)); if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) { SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s", SDL_GetError()); SDL_assert(SDL_WasInit(SDL_INIT_AUDIO)); } } bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig) { SDL_AudioSpec want, have; SDL_zero(want); want.freq = opusConfig->sampleRate; want.format = AUDIO_S16; want.channels = opusConfig->channelCount; // This is supposed to be a power of 2, but our // frames contain a non-power of 2 number of samples, // so the slop would require buffering another full frame. // Specifying non-Po2 seems to work for our supported platforms. want.samples = SAMPLES_PER_FRAME; m_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0); if (m_AudioDevice == 0) { SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Failed to open audio device: %s", SDL_GetError()); SDL_QuitSubSystem(SDL_INIT_AUDIO); return false; } // SDL counts pending samples in the queued // audio size using the WASAPI backend. This // includes silence, which can throw off our // pending data count. Get a baseline so we // can exclude that data. m_BaselinePendingData = 0; #ifdef Q_OS_WIN32 for (int i = 0; i < 100; i++) { m_BaselinePendingData = qMax(m_BaselinePendingData, SDL_GetQueuedAudioSize(m_AudioDevice)); SDL_Delay(10); } #endif m_BaselinePendingData *= 2; SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, "Baseline pending audio data: %d bytes", m_BaselinePendingData); m_ChannelCount = opusConfig->channelCount; m_SampleIndex = 0; m_PendingDrops = m_PendingHardDrops = 0; // Start playback SDL_PauseAudioDevice(m_AudioDevice, 0); return true; } SdlAudioRenderer::~SdlAudioRenderer() { if (m_AudioDevice != 0) { // Stop playback SDL_PauseAudioDevice(m_AudioDevice, 1); SDL_CloseAudioDevice(m_AudioDevice); } SDL_QuitSubSystem(SDL_INIT_AUDIO); SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO)); } void SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize) { m_SampleIndex++; Uint32 queuedAudio = qMax((int)SDL_GetQueuedAudioSize(m_AudioDevice) - (int)m_BaselinePendingData, 0); Uint32 framesQueued = queuedAudio / (SAMPLES_PER_FRAME * m_ChannelCount * sizeof(short)); // We must check this prior to the below checks to ensure we don't // underflow if framesQueued - m_PendingHardDrops < 0. if (framesQueued <= MIN_QUEUED_FRAMES) { m_PendingDrops = m_PendingHardDrops = 0; } // Pend enough drops to get us back to MIN_QUEUED_FRAMES else if (framesQueued - m_PendingHardDrops > STOP_THE_WORLD_LIMIT) { m_PendingHardDrops = framesQueued - MIN_QUEUED_FRAMES; SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION, "Pending hard drop of %u audio frames", m_PendingHardDrops); } else if (framesQueued - m_PendingHardDrops - m_PendingDrops > MAX_QUEUED_FRAMES) { m_PendingDrops = framesQueued - MIN_QUEUED_FRAMES; } // Determine if this frame should be dropped if (m_PendingHardDrops != 0) { // Hard drops happen all at once to forcefully // resync with the source. m_PendingHardDrops--; return; } else if (m_PendingDrops != 0 && m_SampleIndex % DROP_RATIO_DENOM == 0) { // Normal drops are interspersed with the audio data // to hide the glitches. m_PendingDrops--; return; } if (SDL_QueueAudio(m_AudioDevice, audioBuffer, audioSize) < 0) { SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Failed to queue audio sample: %s", SDL_GetError()); } }