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Remove excess latency control logic from SDL renderer, since it doesn't appear to work very well anyway

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This commit is contained in:
Cameron Gutman 2019-01-06 13:28:43 -08:00
parent c21ba5c808
commit 52ef84873e
2 changed files with 1 additions and 84 deletions
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5
app/streaming/audio/renderers/sdl.h
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@ -16,9 +16,4 @@ public:
private:
SDL_AudioDeviceID m_AudioDevice;
Uint32 m_ChannelCount;
Uint32 m_PendingDrops;
Uint32 m_PendingHardDrops;
Uint32 m_SampleIndex;
Uint32 m_BaselinePendingData;
};

80
app/streaming/audio/renderers/sdlaud.cpp
View File

@ -5,18 +5,8 @@
#include <QtGlobal>
#define MIN_QUEUED_FRAMES 2
#define MAX_QUEUED_FRAMES 4
#define STOP_THE_WORLD_LIMIT 20
#define DROP_RATIO_DENOM 32
SdlAudioRenderer::SdlAudioRenderer()
: m_AudioDevice(0),
m_ChannelCount(0),
m_PendingDrops(0),
m_PendingHardDrops(0),
m_SampleIndex(0),
m_BaselinePendingData(0)
: m_AudioDevice(0)
{
SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
@ -61,27 +51,6 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
have.samples,
have.size);
// 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);
@ -102,53 +71,6 @@ SdlAudioRenderer::~SdlAudioRenderer()
bool SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
{
m_SampleIndex++;
Uint32 queuedAudio = SDL_GetQueuedAudioSize(m_AudioDevice);
if (queuedAudio > m_BaselinePendingData) {
queuedAudio -= m_BaselinePendingData;
}
else {
queuedAudio = 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, checking first
// to ensure we don't underflow.
else if (framesQueued > m_PendingHardDrops &&
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);
}
// If we're under the stop the world limit, we can drop samples
// gracefully over the next little while.
else if (framesQueued > m_PendingHardDrops + m_PendingDrops &&
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 true;
}
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 true;
}
if (SDL_QueueAudio(m_AudioDevice, audioBuffer, audioSize) < 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to queue audio sample: %s",