mirror of
https://github.com/moonlight-stream/moonlight-qt
synced 2024-12-24 10:03:09 +00:00
482 lines
17 KiB
C++
482 lines
17 KiB
C++
#include "soundioaudiorenderer.h"
|
|
|
|
#include <SDL.h>
|
|
|
|
#include <QtGlobal>
|
|
|
|
SoundIoAudioRenderer::SoundIoAudioRenderer()
|
|
: m_OpusChannelCount(0),
|
|
m_SoundIo(nullptr),
|
|
m_Device(nullptr),
|
|
m_OutputStream(nullptr),
|
|
m_RingBuffer(nullptr),
|
|
m_AudioPacketDuration(0),
|
|
m_Latency(0),
|
|
m_Errored(false)
|
|
{
|
|
|
|
}
|
|
|
|
SoundIoAudioRenderer::~SoundIoAudioRenderer()
|
|
{
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Audio latency: %f",
|
|
m_Latency);
|
|
|
|
if (m_OutputStream != nullptr) {
|
|
soundio_outstream_destroy(m_OutputStream);
|
|
}
|
|
|
|
// Must be destroyed after the stream is stopped
|
|
// or we could still get sioWriteCallback() calls.
|
|
if (m_RingBuffer != nullptr) {
|
|
soundio_ring_buffer_destroy(m_RingBuffer);
|
|
}
|
|
|
|
if (m_Device != nullptr) {
|
|
soundio_device_unref(m_Device);
|
|
}
|
|
|
|
if (m_SoundIo != nullptr) {
|
|
soundio_destroy(m_SoundIo);
|
|
}
|
|
}
|
|
|
|
int SoundIoAudioRenderer::scoreChannelLayout(const struct SoundIoChannelLayout* layout, const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
|
|
{
|
|
int score = 50;
|
|
|
|
// Compute a score for this layout based on how many matching channels
|
|
// we find (or acceptable alternatives).
|
|
for (int i = 0; i < layout->channel_count; i++) {
|
|
if (opusConfig->channelCount >= 2) {
|
|
switch (layout->channels[i]) {
|
|
case SoundIoChannelIdFrontLeft:
|
|
case SoundIoChannelIdFrontRight:
|
|
score += 2;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (opusConfig->channelCount >= 6) {
|
|
switch (layout->channels[i]) {
|
|
case SoundIoChannelIdFrontCenter:
|
|
case SoundIoChannelIdLfe:
|
|
score += 2;
|
|
break;
|
|
|
|
case SoundIoChannelIdSideLeft:
|
|
case SoundIoChannelIdSideRight:
|
|
score++;
|
|
break;
|
|
|
|
case SoundIoChannelIdBackLeft:
|
|
case SoundIoChannelIdBackRight:
|
|
if (opusConfig->channelCount == 6) {
|
|
// Score back channels higher in 5.1 mode to
|
|
// discourage selection of side channel layouts.
|
|
score += 2;
|
|
}
|
|
else {
|
|
// Score back channels normally for 7.1 mode
|
|
score++;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Now subtract the difference between the desired and actual channel count
|
|
// to punish layouts that have extra unused speakers.
|
|
if (opusConfig->channelCount < layout->channel_count) {
|
|
score -= layout->channel_count - opusConfig->channelCount;
|
|
}
|
|
|
|
return score;
|
|
}
|
|
|
|
bool SoundIoAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
|
|
{
|
|
m_SoundIo = soundio_create();
|
|
if (m_SoundIo == nullptr) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_create() failed");
|
|
return false;
|
|
}
|
|
|
|
m_SoundIo->app_name = "Moonlight";
|
|
m_SoundIo->userdata = this;
|
|
m_SoundIo->on_backend_disconnect = sioBackendDisconnect;
|
|
m_SoundIo->on_devices_change = sioDevicesChanged;
|
|
|
|
int err = soundio_connect(m_SoundIo);
|
|
if (err != SoundIoErrorNone) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_connect() failed: %s",
|
|
soundio_strerror(err));
|
|
return false;
|
|
}
|
|
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Audio backend: %s",
|
|
soundio_backend_name(m_SoundIo->current_backend));
|
|
|
|
// Don't continue if we could only open the dummy backend
|
|
if (m_SoundIo->current_backend == SoundIoBackendDummy) {
|
|
return false;
|
|
}
|
|
|
|
// Flush events to update with new device arrivals
|
|
soundio_flush_events(m_SoundIo);
|
|
|
|
// Remember the actual channel count for later
|
|
m_OpusChannelCount = opusConfig->channelCount;
|
|
|
|
int outputDeviceIndex = soundio_default_output_device_index(m_SoundIo);
|
|
if (outputDeviceIndex < 0) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"No output device found");
|
|
return false;
|
|
}
|
|
|
|
m_Device = soundio_get_output_device(m_SoundIo, outputDeviceIndex);
|
|
if (m_Device == nullptr) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_get_output_device() failed");
|
|
return false;
|
|
}
|
|
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Selected audio device: %s",
|
|
m_Device->name);
|
|
|
|
m_OutputStream = soundio_outstream_create(m_Device);
|
|
if (m_OutputStream == nullptr) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_outstream_create() failed");
|
|
return false;
|
|
}
|
|
|
|
m_AudioPacketDuration = (opusConfig->samplesPerFrame / (opusConfig->sampleRate / 1000)) / 1000.0;
|
|
|
|
m_OutputStream->format = SoundIoFormatS16NE;
|
|
m_OutputStream->sample_rate = opusConfig->sampleRate;
|
|
m_OutputStream->software_latency = m_AudioPacketDuration;
|
|
m_OutputStream->name = "Moonlight";
|
|
m_OutputStream->userdata = this;
|
|
m_OutputStream->error_callback = sioErrorCallback;
|
|
m_OutputStream->write_callback = sioWriteCallback;
|
|
|
|
SoundIoChannelLayout bestLayout = m_Device->current_layout;
|
|
for (int i = 0; i < m_Device->layout_count; i++) {
|
|
if (scoreChannelLayout(&bestLayout, opusConfig) <
|
|
scoreChannelLayout(&m_Device->layouts[i], opusConfig)) {
|
|
bestLayout = m_Device->layouts[i];
|
|
}
|
|
}
|
|
|
|
if (bestLayout.channel_count < opusConfig->channelCount) {
|
|
SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
|
|
"No compatible channel layouts found. Some channels may not be played!");
|
|
}
|
|
|
|
m_OutputStream->layout = bestLayout;
|
|
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Native layout: %s (%d channels)",
|
|
m_OutputStream->layout.name ?
|
|
m_OutputStream->layout.name : "<UNKNOWN>",
|
|
m_OutputStream->layout.channel_count);
|
|
|
|
err = soundio_outstream_open(m_OutputStream);
|
|
if (err != SoundIoErrorNone) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_outstream_open() failed: %s",
|
|
soundio_strerror(err));
|
|
return false;
|
|
}
|
|
|
|
if (m_OutputStream->layout_error) {
|
|
SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Channel layout failed: %s",
|
|
soundio_strerror(m_OutputStream->layout_error));
|
|
|
|
// ALSA through PulseAudio appears to fail snd_pcm_set_chmap()
|
|
// even after claiming the layout is supported (and even on totally
|
|
// standard layouts like Stereo). We'll just ignore this for ALSA
|
|
// and only bail if we get an actual failure out of one of these APIs.
|
|
if (m_SoundIo->current_backend != SoundIoBackendAlsa) {
|
|
return false;
|
|
}
|
|
}
|
|
|
|
m_EffectiveLayout = m_OutputStream->layout;
|
|
for (int i = 0; i < m_EffectiveLayout.channel_count; i++) {
|
|
if (opusConfig->channelCount == 6) {
|
|
// For 5.1, replace side L/R with back L/R so our channel position
|
|
// logic in sioWriteCallback() works.
|
|
if (m_EffectiveLayout.channels[i] == SoundIoChannelIdSideLeft) {
|
|
m_EffectiveLayout.channels[i] = SoundIoChannelIdBackLeft;
|
|
}
|
|
if (m_EffectiveLayout.channels[i] == SoundIoChannelIdSideRight) {
|
|
m_EffectiveLayout.channels[i] = SoundIoChannelIdBackRight;
|
|
}
|
|
}
|
|
else if (opusConfig->channelCount == 8) {
|
|
// For 5.1, replace side L/R with LOC/ROC so our channel position
|
|
// logic in sioWriteCallback() works.
|
|
if (m_EffectiveLayout.channels[i] == SoundIoChannelIdSideLeft) {
|
|
m_EffectiveLayout.channels[i] = SoundIoChannelIdFrontLeftCenter;
|
|
}
|
|
if (m_EffectiveLayout.channels[i] == SoundIoChannelIdSideRight) {
|
|
m_EffectiveLayout.channels[i] = SoundIoChannelIdFrontRightCenter;
|
|
}
|
|
}
|
|
}
|
|
|
|
int packetsToBuffer;
|
|
|
|
if (m_SoundIo->current_backend == SoundIoBackendWasapi) {
|
|
// 15 ms buffer seems to be fine for WASAPI
|
|
packetsToBuffer = (int)ceil(0.015 / m_AudioPacketDuration);
|
|
}
|
|
else {
|
|
// 30 ms buffer on CoreAudio to avoid glitching on macOS
|
|
packetsToBuffer = (int)ceil(0.030 / m_AudioPacketDuration);
|
|
}
|
|
|
|
// Always buffer at least 2 packets
|
|
packetsToBuffer = qMax(2, packetsToBuffer);
|
|
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Audio buffer size: %f seconds",
|
|
packetsToBuffer * m_AudioPacketDuration);
|
|
|
|
m_RingBuffer = soundio_ring_buffer_create(m_SoundIo,
|
|
m_OutputStream->bytes_per_sample *
|
|
m_OpusChannelCount *
|
|
opusConfig->samplesPerFrame *
|
|
packetsToBuffer);
|
|
if (m_RingBuffer == nullptr) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_ring_buffer_create() failed");
|
|
return false;
|
|
}
|
|
|
|
err = soundio_outstream_start(m_OutputStream);
|
|
if (err != SoundIoErrorNone) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_outstream_start() failed: %s",
|
|
soundio_strerror(err));
|
|
return false;
|
|
}
|
|
|
|
// HACK: For some reason, a constant latency hangs around in the audio pipeline
|
|
// unless we wait for the audio stream to drain before actually submitting any samples.
|
|
// This is a gross hack, but it works remarkably well.
|
|
SDL_Delay(500);
|
|
|
|
return true;
|
|
}
|
|
|
|
void* SoundIoAudioRenderer::getAudioBuffer(int* size)
|
|
{
|
|
// We must always write a full frame of audio. If we don't,
|
|
// the reader will get out of sync with the writer and our
|
|
// channels will get all mixed up. To ensure this is always
|
|
// the case, round our bytes free down to the next multiple
|
|
// of our frame size.
|
|
int bytesFree = soundio_ring_buffer_free_count(m_RingBuffer);
|
|
int bytesPerFrame = m_OpusChannelCount * m_OutputStream->bytes_per_sample;
|
|
*size = qMin(*size, (bytesFree / bytesPerFrame) * bytesPerFrame);
|
|
return soundio_ring_buffer_write_ptr(m_RingBuffer);
|
|
}
|
|
|
|
bool SoundIoAudioRenderer::submitAudio(int bytesWritten)
|
|
{
|
|
if (m_Errored) {
|
|
return false;
|
|
}
|
|
|
|
if (bytesWritten == 0) {
|
|
// Nothing to do
|
|
return true;
|
|
}
|
|
|
|
// Flush events to update with new device arrivals
|
|
soundio_flush_events(m_SoundIo);
|
|
|
|
// Advance the write pointer
|
|
soundio_ring_buffer_advance_write_ptr(m_RingBuffer, bytesWritten);
|
|
|
|
return true;
|
|
}
|
|
|
|
int SoundIoAudioRenderer::getCapabilities()
|
|
{
|
|
// TODO: Tweak buffer sizes then re-enable arbitrary audio duration
|
|
return CAPABILITY_DIRECT_SUBMIT /* | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION */;
|
|
}
|
|
|
|
void SoundIoAudioRenderer::sioErrorCallback(SoundIoOutStream* stream, int err)
|
|
{
|
|
auto me = reinterpret_cast<SoundIoAudioRenderer*>(stream->userdata);
|
|
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Audio rendering error: %s",
|
|
soundio_strerror(err));
|
|
|
|
// Trigger reinitialization
|
|
me->m_Errored = true;
|
|
}
|
|
|
|
void SoundIoAudioRenderer::sioBackendDisconnect(SoundIo* soundio, int err)
|
|
{
|
|
auto me = reinterpret_cast<SoundIoAudioRenderer*>(soundio->userdata);
|
|
|
|
SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Audio backend disconnected: %s",
|
|
soundio_strerror(err));
|
|
|
|
// Trigger reinitialization
|
|
me->m_Errored = true;
|
|
}
|
|
|
|
void SoundIoAudioRenderer::sioDevicesChanged(SoundIo* soundio)
|
|
{
|
|
auto me = reinterpret_cast<SoundIoAudioRenderer*>(soundio->userdata);
|
|
|
|
if (me->m_Device == nullptr) {
|
|
// Ignore calls that take place during initialization
|
|
return;
|
|
}
|
|
|
|
int outputDeviceIndex = soundio_default_output_device_index(soundio);
|
|
if (outputDeviceIndex >= 0) {
|
|
struct SoundIoDevice* outputDevice = soundio_get_output_device(soundio, outputDeviceIndex);
|
|
if (outputDevice == nullptr) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_get_output_device() failed");
|
|
return;
|
|
}
|
|
|
|
if (!soundio_device_equal(outputDevice, me->m_Device)) {
|
|
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
|
|
"Default audio output device changed");
|
|
|
|
// Trigger reinitialization
|
|
me->m_Errored = true;
|
|
}
|
|
|
|
soundio_device_unref(outputDevice);
|
|
}
|
|
}
|
|
|
|
// bytes_per_frame should never be used on the ring buffer! It's not always
|
|
// the same number of bytes per frames as the output stream!
|
|
void SoundIoAudioRenderer::sioWriteCallback(SoundIoOutStream* stream, int frameCountMin, int frameCountMax)
|
|
{
|
|
auto me = reinterpret_cast<SoundIoAudioRenderer*>(stream->userdata);
|
|
char* readPtr = soundio_ring_buffer_read_ptr(me->m_RingBuffer);
|
|
int framesLeft = soundio_ring_buffer_fill_count(me->m_RingBuffer) /
|
|
(me->m_OpusChannelCount * stream->bytes_per_sample);
|
|
int bytesRead = 0;
|
|
|
|
// Ensure we always write at least a buffer, even if it's silence, to avoid
|
|
// busy looping when no audio data is available while libsoundio tries to keep
|
|
// us from starving the output device.
|
|
frameCountMin = qMax(frameCountMin, (int)(stream->sample_rate * me->m_AudioPacketDuration));
|
|
|
|
// Clamp frameCountMax to at least 2 packets or 20 ms to stop our latency from growing if audio packets lag.
|
|
// This makes sure that we never increase our latency far beyond what the sink is consuming.
|
|
frameCountMax = qMin(frameCountMax, (int)(stream->sample_rate * qMax(me->m_AudioPacketDuration * 2, 0.020)));
|
|
frameCountMin = qMin(frameCountMin, frameCountMax);
|
|
|
|
// Clamp framesLeft to frameCountMax
|
|
framesLeft = qMin(framesLeft, frameCountMax);
|
|
|
|
// Track latency on queueing-based backends
|
|
if (me->m_SoundIo->current_backend != SoundIoBackendCoreAudio && me->m_SoundIo->current_backend != SoundIoBackendJack) {
|
|
soundio_outstream_get_latency(stream, &me->m_Latency);
|
|
}
|
|
|
|
for (;;) {
|
|
int frameCount;
|
|
int err;
|
|
struct SoundIoChannelArea* areas;
|
|
|
|
// Always meet the minimum but don't write more than that
|
|
// if we'll have to insert silence
|
|
frameCount = qMax(framesLeft, frameCountMin);
|
|
|
|
if (frameCount == 0) {
|
|
// Nothing more to write
|
|
break;
|
|
}
|
|
|
|
err = soundio_outstream_begin_write(stream, &areas, &frameCount);
|
|
if (err != SoundIoErrorNone) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_outstream_begin_write() failed: %s",
|
|
soundio_strerror(err));
|
|
break;
|
|
}
|
|
|
|
for (int frame = 0; frame < frameCount; frame++) {
|
|
for (int ch = 0; ch < me->m_EffectiveLayout.channel_count; ch++) {
|
|
// SoundIoChannelId - 1 happens to match Moonlight's channel layout
|
|
// after we've applied our fixups to m_EffectiveLayout for 5.1 and 7.1.
|
|
int readPtrChannel = me->m_EffectiveLayout.channels[ch] - 1;
|
|
|
|
if (frame >= framesLeft || readPtrChannel >= me->m_OpusChannelCount) {
|
|
// Write silence if we have no buffered frames left or
|
|
// nothing in the audio stream for this channel
|
|
memset(areas[ch].ptr, 0, stream->bytes_per_sample);
|
|
}
|
|
else {
|
|
// Write audio data from our ring buffer
|
|
memcpy(areas[ch].ptr,
|
|
&readPtr[readPtrChannel * stream->bytes_per_sample],
|
|
stream->bytes_per_sample);
|
|
}
|
|
|
|
areas[ch].ptr += areas[ch].step;
|
|
}
|
|
|
|
// Move on to the next frame if we aren't inserting silence
|
|
if (frame < framesLeft) {
|
|
readPtr += stream->bytes_per_sample * me->m_OpusChannelCount;
|
|
bytesRead += stream->bytes_per_sample * me->m_OpusChannelCount;
|
|
}
|
|
}
|
|
|
|
err = soundio_outstream_end_write(stream);
|
|
if (err != SoundIoErrorNone && err != SoundIoErrorUnderflow) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
|
|
"soundio_outstream_end_write() failed: %s",
|
|
soundio_strerror(err));
|
|
break;
|
|
}
|
|
|
|
if (framesLeft >= frameCount) {
|
|
framesLeft -= frameCount;
|
|
}
|
|
else {
|
|
framesLeft = 0;
|
|
}
|
|
|
|
if (frameCountMin >= frameCount) {
|
|
frameCountMin -= frameCount;
|
|
}
|
|
else {
|
|
frameCountMin = 0;
|
|
}
|
|
}
|
|
|
|
soundio_ring_buffer_advance_read_ptr(me->m_RingBuffer, bytesRead);
|
|
}
|