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	cubeb_sink: Perform audio stretching
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				| @ -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; | ||||
| }; | ||||
|  | ||||
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