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audio: Wait for samples on the emulated DSP side to avoid desyncs
Waiting on the host side is inaccurate and leads to desyncs in the event of the sink missing a deadline that require stalls to fix. By waiting for the sink to have space before even starting rendering such desyncs can be avoided.
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@ -205,6 +205,10 @@ void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::siz
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// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
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// paused and we'll desync, so just play silence.
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if (system.IsPaused() || system.IsShuttingDown()) {
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if (system.IsShuttingDown()) {
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release_cv.notify_one();
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}
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static constexpr std::array<s16, 6> silence{};
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for (size_t i = frames_written; i < num_frames; i++) {
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std::memcpy(&output_buffer[i * frame_size], &silence[0], frame_size_bytes);
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@ -240,6 +244,12 @@ void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::siz
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}
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// Successfully dequeued a new buffer.
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queued_buffers--;
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{
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std::unique_lock lk{release_mutex};
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}
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release_cv.notify_one();
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}
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// Get the minimum frames available between the currently playing buffer, and the
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@ -303,4 +313,9 @@ u64 SinkStream::GetExpectedPlayedSampleCount() {
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return std::min<u64>(exp_played_sample_count, max_played_sample_count);
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}
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void SinkStream::WaitFreeSpace() {
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std::unique_lock lk{release_mutex};
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release_cv.wait(lk, [this]() { return queued_buffers < max_queue_size || system.IsShuttingDown(); });
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}
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} // namespace AudioCore::Sink
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