mirror of
https://github.com/yuzu-emu/yuzu-mainline.git
synced 2024-12-12 20:54:22 +01:00
cubeb_sink: Perform audio stretching
This commit is contained in:
parent
e51bd49f87
commit
1aa195a9c0
@ -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));
|
||||
}
|
||||
|
||||
|
@ -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);
|
||||
|
@ -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;
|
||||
};
|
||||
|
Loading…
Reference in New Issue
Block a user