cubeb_sink: Perform audio stretching
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e51bd49f87
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1aa195a9c0
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@ -6,8 +6,10 @@
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#include <cstring>
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#include <cstring>
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#include "audio_core/cubeb_sink.h"
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#include "audio_core/cubeb_sink.h"
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#include "audio_core/stream.h"
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#include "audio_core/stream.h"
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#include "audio_core/time_stretch.h"
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#include "common/logging/log.h"
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#include "common/logging/log.h"
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#include "common/ring_buffer.h"
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#include "common/ring_buffer.h"
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#include "core/settings.h"
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namespace AudioCore {
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namespace AudioCore {
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@ -15,14 +17,8 @@ class CubebSinkStream final : public SinkStream {
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public:
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public:
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CubebSinkStream(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
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CubebSinkStream(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
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const std::string& name)
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const std::string& name)
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: ctx{ctx}, num_channels{num_channels_} {
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: ctx{ctx}, is_6_channel{num_channels_ == 6}, num_channels{std::min(num_channels_, 2u)},
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time_stretch{sample_rate, num_channels} {
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if (num_channels == 6) {
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// 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
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// channel for now
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is_6_channel = true;
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num_channels = 2;
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}
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cubeb_stream_params params{};
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cubeb_stream_params params{};
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params.rate = sample_rate;
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params.rate = sample_rate;
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@ -89,10 +85,6 @@ public:
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return num_channels;
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return num_channels;
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}
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}
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u32 GetNumChannelsInQueue() const {
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return num_channels == 1 ? 1 : 2;
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}
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private:
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private:
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std::vector<std::string> device_list;
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std::vector<std::string> device_list;
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@ -103,6 +95,7 @@ private:
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Common::RingBuffer<s16, 0x10000> queue;
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Common::RingBuffer<s16, 0x10000> queue;
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std::array<s16, 2> last_frame;
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std::array<s16, 2> last_frame;
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TimeStretcher time_stretch;
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static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames);
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void* output_buffer, long num_frames);
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@ -153,7 +146,7 @@ SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
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}
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}
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long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames) {
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void* output_buffer, long num_frames) {
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CubebSinkStream* impl = static_cast<CubebSinkStream*>(user_data);
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CubebSinkStream* impl = static_cast<CubebSinkStream*>(user_data);
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u8* buffer = reinterpret_cast<u8*>(output_buffer);
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u8* buffer = reinterpret_cast<u8*>(output_buffer);
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@ -161,9 +154,19 @@ long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const
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return {};
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return {};
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}
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}
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const size_t num_channels = impl->GetNumChannelsInQueue();
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const size_t num_channels = impl->GetNumChannels();
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const size_t max_samples_to_write = num_channels * num_frames;
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const size_t samples_to_write = num_channels * num_frames;
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const size_t samples_written = impl->queue.Pop(buffer, max_samples_to_write);
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size_t samples_written;
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if (Settings::values.enable_audio_stretching) {
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const std::vector<s16> in{impl->queue.Pop()};
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const size_t num_in{in.size() / num_channels};
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s16* const out{reinterpret_cast<s16*>(buffer)};
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const size_t out_frames = impl->time_stretch.Process(in.data(), num_in, out, num_frames);
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samples_written = out_frames * num_channels;
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} else {
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samples_written = impl->queue.Pop(buffer, samples_to_write);
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}
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if (samples_written >= num_channels) {
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if (samples_written >= num_channels) {
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std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
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std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
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@ -171,7 +174,7 @@ long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const
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}
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}
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// Fill the rest of the frames with last_frame
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// Fill the rest of the frames with last_frame
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for (size_t i = samples_written; i < max_samples_to_write; i += num_channels) {
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for (size_t i = samples_written; i < samples_to_write; i += num_channels) {
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std::memcpy(buffer + i * sizeof(s16), &impl->last_frame[0], num_channels * sizeof(s16));
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std::memcpy(buffer + i * sizeof(s16), &impl->last_frame[0], num_channels * sizeof(s16));
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}
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}
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@ -28,8 +28,8 @@ size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num
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// We were given actual_samples number of samples, and num_samples were requested from us.
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// We were given actual_samples number of samples, and num_samples were requested from us.
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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const double max_latency = 0.3; // seconds
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const double max_latency = 1.0; // seconds
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const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
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const double max_backlog = m_sample_rate * max_latency;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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if (backlog_fullness > 5.0) {
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if (backlog_fullness > 5.0) {
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// Too many samples in backlog: Don't push anymore on
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// Too many samples in backlog: Don't push anymore on
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@ -49,13 +49,13 @@ size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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// Place a lower limit of 10% speed. When a game boots up, there will be
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// Place a lower limit of 5% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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// many silence samples. These do not need to be timestretched.
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m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
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m_stretch_ratio = std::max(m_stretch_ratio, 0.05);
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m_sound_touch.setTempo(m_stretch_ratio);
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m_sound_touch.setTempo(m_stretch_ratio);
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LOG_DEBUG(Audio, "Audio Stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in, num_out,
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LOG_DEBUG(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, m_stretch_ratio,
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m_stretch_ratio, backlog_fullness, lpf_gain);
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backlog_fullness);
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m_sound_touch.putSamples(in, num_in);
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m_sound_touch.putSamples(in, num_in);
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return m_sound_touch.receiveSamples(out, num_out);
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return m_sound_touch.receiveSamples(out, num_out);
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@ -27,7 +27,6 @@ public:
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private:
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private:
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u32 m_sample_rate;
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u32 m_sample_rate;
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u32 m_channel_count;
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u32 m_channel_count;
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std::array<s16, 2> m_last_stretched_sample = {};
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soundtouch::SoundTouch m_sound_touch;
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soundtouch::SoundTouch m_sound_touch;
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double m_stretch_ratio = 1.0;
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double m_stretch_ratio = 1.0;
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};
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};
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