diff --git a/src/audio_core/cubeb_sink.cpp b/src/audio_core/cubeb_sink.cpp index 3c129122fe..7982306b3e 100644 --- a/src/audio_core/cubeb_sink.cpp +++ b/src/audio_core/cubeb_sink.cpp @@ -6,8 +6,10 @@ #include #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 device_list; @@ -103,6 +95,7 @@ private: Common::RingBuffer queue; std::array 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(user_data); u8* buffer = reinterpret_cast(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 in{impl->queue.Pop()}; + const size_t num_in{in.size() / num_channels}; + s16* const out{reinterpret_cast(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 17e1283236..d2e3391c1b 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(num_in) / static_cast(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 cdead34a24..0322b8b786 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 m_last_stretched_sample = {}; soundtouch::SoundTouch m_sound_touch; double m_stretch_ratio = 1.0; };