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Refactor audio rendering into a separate class

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This commit is contained in:
Cameron Gutman 2018-09-13 06:23:06 -07:00
commit 12496e4432
9 changed files with 244 additions and 131 deletions
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240
app/streaming/audio/renderers/sdlaud.cpp Normal file
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#include "sdl.h"
#include <Limelight.h>
#include <SDL.h>
#include <QtGlobal>
#define MIN_QUEUED_FRAMES 2
#define MAX_QUEUED_FRAMES 4
#define STOP_THE_WORLD_LIMIT 20
#define DROP_RATIO_DENOM 32
int SdlAudioRenderer::detectAudioConfiguration()
{
SDL_AudioSpec want, have;
SDL_AudioDeviceID dev;
int ret;
if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
SDL_GetError());
return AUDIO_CONFIGURATION_STEREO;
}
SDL_zero(want);
want.freq = 48000;
want.format = AUDIO_S16;
want.channels = 6;
want.samples = 1024;
// Try to open for 5.1 surround sound, but allow SDL to tell us that's
// not available.
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_CHANNELS_CHANGE);
if (dev == 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to open audio device");
// We'll probably have issues during audio stream init, but we'll
// try anyway
ret = AUDIO_CONFIGURATION_STEREO;
goto Exit;
}
SDL_CloseAudioDevice(dev);
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Audio device has %d channels", have.channels);
if (have.channels > 2) {
// We don't support quadraphonic or 7.1 surround, but SDL
// should be able to downmix or upmix better from 5.1 than
// from stereo, so use 5.1 in non-stereo cases.
ret = AUDIO_CONFIGURATION_51_SURROUND;
}
else {
ret = AUDIO_CONFIGURATION_STEREO;
}
Exit:
SDL_QuitSubSystem(SDL_INIT_AUDIO);
return ret;
}
bool SdlAudioRenderer::testAudio(int audioConfiguration)
{
SDL_AudioSpec want, have;
SDL_AudioDeviceID dev;
bool ret;
if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Audio test - SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
SDL_GetError());
return false;
}
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);
ret = false;
goto Exit;
}
// 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());
ret = false;
goto Exit;
}
SDL_CloseAudioDevice(dev);
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Audio test - Successful with %d channels",
want.channels);
ret = true;
Exit:
SDL_QuitSubSystem(SDL_INIT_AUDIO);
return ret;
}
SdlAudioRenderer::SdlAudioRenderer()
: m_AudioDevice(0),
m_ChannelCount(0),
m_PendingDrops(0),
m_PendingHardDrops(0),
m_SampleIndex(0),
m_BaselinePendingData(0)
{
}
bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
{
SDL_AudioSpec want, have;
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());
return -1;
}
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 -1;
}
// 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 0;
}
SdlAudioRenderer::~SdlAudioRenderer()
{
// 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());
}
}