diff --git a/src/audio_core/cubeb_sink.cpp b/src/audio_core/cubeb_sink.cpp
index 3c129122fe3034da4028cc99570167c87f7b2baa..7982306b3ed9798ca05f29c8b13cc207db13fe1e 100644
--- a/src/audio_core/cubeb_sink.cpp
+++ b/src/audio_core/cubeb_sink.cpp
@@ -6,8 +6,10 @@
#include <cstring>
#include "audio_core/cubeb_sink.h"
#include "audio_core/stream.h"
+#include "audio_core/time_stretch.h"
#include "common/logging/log.h"
#include "common/ring_buffer.h"
+#include "core/settings.h"
namespace AudioCore {
@@ -15,14 +17,8 @@ class CubebSinkStream final : public SinkStream {
public:
CubebSinkStream(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
const std::string& name)
- : ctx{ctx}, num_channels{num_channels_} {
-
- if (num_channels == 6) {
- // 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
- // channel for now
- is_6_channel = true;
- num_channels = 2;
- }
+ : ctx{ctx}, is_6_channel{num_channels_ == 6}, num_channels{std::min(num_channels_, 2u)},
+ time_stretch{sample_rate, num_channels} {
cubeb_stream_params params{};
params.rate = sample_rate;
@@ -89,10 +85,6 @@ public:
return num_channels;
}
- u32 GetNumChannelsInQueue() const {
- return num_channels == 1 ? 1 : 2;
- }
-
private:
std::vector<std::string> device_list;
@@ -103,6 +95,7 @@ private:
Common::RingBuffer<s16, 0x10000> queue;
std::array<s16, 2> last_frame;
+ TimeStretcher time_stretch;
static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames);
@@ -153,7 +146,7 @@ SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
}
long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
- void* output_buffer, long num_frames) {
+ void* output_buffer, long num_frames) {
CubebSinkStream* impl = static_cast<CubebSinkStream*>(user_data);
u8* buffer = reinterpret_cast<u8*>(output_buffer);
@@ -161,9 +154,19 @@ long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const
return {};
}
- const size_t num_channels = impl->GetNumChannelsInQueue();
- const size_t max_samples_to_write = num_channels * num_frames;
- const size_t samples_written = impl->queue.Pop(buffer, max_samples_to_write);
+ const size_t num_channels = impl->GetNumChannels();
+ const size_t samples_to_write = num_channels * num_frames;
+ size_t samples_written;
+
+ if (Settings::values.enable_audio_stretching) {
+ const std::vector<s16> in{impl->queue.Pop()};
+ const size_t num_in{in.size() / num_channels};
+ s16* const out{reinterpret_cast<s16*>(buffer)};
+ const size_t out_frames = impl->time_stretch.Process(in.data(), num_in, out, num_frames);
+ samples_written = out_frames * num_channels;
+ } else {
+ samples_written = impl->queue.Pop(buffer, samples_to_write);
+ }
if (samples_written >= num_channels) {
std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
@@ -171,7 +174,7 @@ long CubebSinkStream::DataCallback(cubeb_stream* stream, void* user_data, const
}
// Fill the rest of the frames with last_frame
- for (size_t i = samples_written; i < max_samples_to_write; i += num_channels) {
+ for (size_t i = samples_written; i < samples_to_write; i += num_channels) {
std::memcpy(buffer + i * sizeof(s16), &impl->last_frame[0], num_channels * sizeof(s16));
}
diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp
index 17e1283236736618b0f50dd0da7b59aed760a772..d2e3391c1bcd89e1e004205d29834ba7677ce3fb 100644
--- a/src/audio_core/time_stretch.cpp
+++ b/src/audio_core/time_stretch.cpp
@@ -28,8 +28,8 @@ size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num
// We were given actual_samples number of samples, and num_samples were requested from us.
double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
- const double max_latency = 0.3; // seconds
- const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
+ const double max_latency = 1.0; // seconds
+ const double max_backlog = m_sample_rate * max_latency;
const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
if (backlog_fullness > 5.0) {
// Too many samples in backlog: Don't push anymore on
@@ -49,13 +49,13 @@ size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num
const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
- // Place a lower limit of 10% speed. When a game boots up, there will be
+ // Place a lower limit of 5% speed. When a game boots up, there will be
// many silence samples. These do not need to be timestretched.
- m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
+ m_stretch_ratio = std::max(m_stretch_ratio, 0.05);
m_sound_touch.setTempo(m_stretch_ratio);
- LOG_DEBUG(Audio, "Audio Stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in, num_out,
- m_stretch_ratio, backlog_fullness, lpf_gain);
+ LOG_DEBUG(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, m_stretch_ratio,
+ backlog_fullness);
m_sound_touch.putSamples(in, num_in);
return m_sound_touch.receiveSamples(out, num_out);
diff --git a/src/audio_core/time_stretch.h b/src/audio_core/time_stretch.h
index cdead34a247011694bcb7b8d21408e4734971edc..0322b8b78647ca6e297b442d68a901ac32ceff47 100644
--- a/src/audio_core/time_stretch.h
+++ b/src/audio_core/time_stretch.h
@@ -27,7 +27,6 @@ public:
private:
u32 m_sample_rate;
u32 m_channel_count;
- std::array<s16, 2> m_last_stretched_sample = {};
soundtouch::SoundTouch m_sound_touch;
double m_stretch_ratio = 1.0;
};