diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj b/Source/Core/AudioCommon/AudioCommon.vcxproj index 6ab07e55f4..dad6e891ed 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj @@ -38,6 +38,7 @@ + @@ -54,6 +55,7 @@ + diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters index 35ba0a6c7d..661c6ce0ea 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters @@ -8,6 +8,7 @@ + @@ -31,6 +32,7 @@ + diff --git a/Source/Core/AudioCommon/AudioStretcher.cpp b/Source/Core/AudioCommon/AudioStretcher.cpp new file mode 100644 index 0000000000..88721b0e35 --- /dev/null +++ b/Source/Core/AudioCommon/AudioStretcher.cpp @@ -0,0 +1,89 @@ +// Copyright 2017 Dolphin Emulator Project +// Licensed under GPLv2+ +// Refer to the license.txt file included. + +#include +#include +#include + +#include "AudioCommon/AudioStretcher.h" +#include "Common/Logging/Log.h" +#include "Core/ConfigManager.h" + +namespace AudioCommon +{ +AudioStretcher::AudioStretcher(unsigned int sample_rate) : m_sample_rate(sample_rate) +{ + m_sound_touch.setChannels(2); + m_sound_touch.setSampleRate(sample_rate); + m_sound_touch.setPitch(1.0); + m_sound_touch.setTempo(1.0); + m_sound_touch.setSetting(SETTING_USE_QUICKSEEK, 0); + m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62); + m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28); + m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8); +} + +void AudioStretcher::Clear() +{ + m_sound_touch.clear(); +} + +void AudioStretcher::ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out) +{ + const double time_delta = static_cast(num_out) / m_sample_rate; // seconds + + // 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 = SConfig::GetInstance().m_audio_stretch_max_latency; + const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio; + 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 + num_in = 0; + } + + // We ideally want the backlog to be about 50% full. + // This gives some headroom both ways to prevent underflow and overflow. + // We tweak current_ratio to encourage this. + constexpr double tweak_time_scale = 0.5; // seconds + current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale); + + // This low-pass filter smoothes out variance in the calculated stretch ratio. + // The time-scale determines how responsive this filter is. + constexpr double lpf_time_scale = 1.0; // seconds + 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 + // many silence samples. These do not need to be timestretched. + m_stretch_ratio = std::max(m_stretch_ratio, 0.1); + m_sound_touch.setTempo(m_stretch_ratio); + + DEBUG_LOG(AUDIO, "Audio stretching: samples:%u/%u ratio:%f backlog:%f gain: %f", num_in, num_out, + m_stretch_ratio, backlog_fullness, lpf_gain); + + m_sound_touch.putSamples(in, num_in); +} + +void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out) +{ + const size_t samples_received = m_sound_touch.receiveSamples(out, num_out); + + if (samples_received != 0) + { + m_last_stretched_sample[0] = out[samples_received * 2 - 2]; + m_last_stretched_sample[1] = out[samples_received * 2 - 1]; + } + + // Perform padding if we've run out of samples. + for (size_t i = samples_received; i < num_out; i++) + { + out[i * 2 + 0] = m_last_stretched_sample[0]; + out[i * 2 + 1] = m_last_stretched_sample[1]; + } +} + +} // namespace AudioCommon diff --git a/Source/Core/AudioCommon/AudioStretcher.h b/Source/Core/AudioCommon/AudioStretcher.h new file mode 100644 index 0000000000..b914fb924b --- /dev/null +++ b/Source/Core/AudioCommon/AudioStretcher.h @@ -0,0 +1,28 @@ +// Copyright 2017 Dolphin Emulator Project +// Licensed under GPLv2+ +// Refer to the license.txt file included. + +#pragma once + +#include + +#include + +namespace AudioCommon +{ +class AudioStretcher +{ +public: + explicit AudioStretcher(unsigned int sample_rate); + void ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out); + void GetStretchedSamples(short* out, unsigned int num_out); + void Clear(); + +private: + unsigned int m_sample_rate; + std::array m_last_stretched_sample = {}; + soundtouch::SoundTouch m_sound_touch; + double m_stretch_ratio = 1.0; +}; + +} // AudioCommon diff --git a/Source/Core/AudioCommon/CMakeLists.txt b/Source/Core/AudioCommon/CMakeLists.txt index 94eb45c9ed..7ede29eb41 100644 --- a/Source/Core/AudioCommon/CMakeLists.txt +++ b/Source/Core/AudioCommon/CMakeLists.txt @@ -1,5 +1,6 @@ set(SRCS AudioCommon.cpp + AudioStretcher.cpp CubebStream.cpp CubebUtils.cpp DPL2Decoder.cpp diff --git a/Source/Core/AudioCommon/CubebStream.cpp b/Source/Core/AudioCommon/CubebStream.cpp index c9052e29c9..272060cd74 100644 --- a/Source/Core/AudioCommon/CubebStream.cpp +++ b/Source/Core/AudioCommon/CubebStream.cpp @@ -21,29 +21,9 @@ long CubebStream::DataCallback(cubeb_stream* stream, void* user_data, const void auto* self = static_cast(user_data); if (self->m_stereo) - { self->m_mixer->Mix(static_cast(output_buffer), num_frames); - } else - { - size_t required_capacity = num_frames * 2; - if (required_capacity > self->m_short_buffer.capacity() || - required_capacity > self->m_floatstereo_buffer.capacity()) - { - INFO_LOG(AUDIO, "Expanding conversion buffers size: %li frames", num_frames); - self->m_short_buffer.reserve(required_capacity); - self->m_floatstereo_buffer.reserve(required_capacity); - } - - self->m_mixer->Mix(self->m_short_buffer.data(), num_frames); - - // s16 to float - for (size_t i = 0; i < static_cast(num_frames) * 2; ++i) - self->m_floatstereo_buffer[i] = self->m_short_buffer[i] / static_cast(1 << 15); - - // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR - DPL2Decode(self->m_floatstereo_buffer.data(), num_frames, static_cast(output_buffer)); - } + self->m_mixer->MixSurround(static_cast(output_buffer), num_frames); return num_frames; } diff --git a/Source/Core/AudioCommon/Mixer.cpp b/Source/Core/AudioCommon/Mixer.cpp index 8149da088c..659deb819f 100644 --- a/Source/Core/AudioCommon/Mixer.cpp +++ b/Source/Core/AudioCommon/Mixer.cpp @@ -7,24 +7,18 @@ #include #include +#include "AudioCommon/DPL2Decoder.h" #include "Common/CommonTypes.h" #include "Common/Logging/Log.h" #include "Common/MathUtil.h" #include "Common/Swap.h" #include "Core/ConfigManager.h" -CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate) +CMixer::CMixer(unsigned int BackendSampleRate) + : m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate) { INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized"); - - m_sound_touch.setChannels(2); - m_sound_touch.setSampleRate(BackendSampleRate); - m_sound_touch.setPitch(1.0); - m_sound_touch.setTempo(1.0); - m_sound_touch.setSetting(SETTING_USE_QUICKSEEK, 0); - m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62); - m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28); - m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8); + DPL2Reset(); } CMixer::~CMixer() @@ -135,18 +129,19 @@ unsigned int CMixer::Mix(short* samples, unsigned int num_samples) unsigned int available_samples = std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples()); - m_stretch_buffer.fill(0); + m_scratch_buffer.fill(0); - m_dma_mixer.Mix(m_stretch_buffer.data(), available_samples, false); - m_streaming_mixer.Mix(m_stretch_buffer.data(), available_samples, false); - m_wiimote_speaker_mixer.Mix(m_stretch_buffer.data(), available_samples, false); + m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false); + m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false); + m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false); if (!m_is_stretching) { - m_sound_touch.clear(); + m_stretcher.Clear(); m_is_stretching = true; } - StretchAudio(m_stretch_buffer.data(), available_samples, samples, num_samples); + m_stretcher.ProcessSamples(m_scratch_buffer.data(), available_samples, num_samples); + m_stretcher.GetStretchedSamples(samples, num_samples); } else { @@ -159,58 +154,25 @@ unsigned int CMixer::Mix(short* samples, unsigned int num_samples) return num_samples; } -void CMixer::StretchAudio(const short* in, unsigned int num_in, short* out, unsigned int num_out) +unsigned int CMixer::MixSurround(float* samples, unsigned int num_samples) { - const double time_delta = static_cast(num_out) / m_sampleRate; // seconds + if (!num_samples) + return 0; - // 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); + memset(samples, 0, num_samples * 6 * sizeof(float)); - const double max_latency = SConfig::GetInstance().m_audio_stretch_max_latency; - const double max_backlog = m_sampleRate * max_latency / 1000.0 / m_stretch_ratio; - const double backlog_fullness = m_sound_touch.numSamples() / max_backlog; - if (backlog_fullness > 5.0) + // Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads and + // writes. + unsigned int available_samples = Mix(m_scratch_buffer.data(), num_samples); + for (size_t i = 0; i < static_cast(available_samples) * 2; ++i) { - // Too many samples in backlog: Don't push anymore on - num_in = 0; + m_float_conversion_buffer[i] = + m_scratch_buffer[i] / static_cast(std::numeric_limits::max()); } - // We ideally want the backlog to be about 50% full. - // This gives some headroom both ways to prevent underflow and overflow. - // We tweak current_ratio to encourage this. - constexpr double tweak_time_scale = 0.5; // seconds - current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale); + DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples); - // This low-pass filter smoothes out variance in the calculated stretch ratio. - // The time-scale determines how responsive this filter is. - constexpr double lpf_time_scale = 1.0; // seconds - const double m_lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale); - m_stretch_ratio += m_lpf_gain * (current_ratio - m_stretch_ratio); - - // Place a lower limit of 10% 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_sound_touch.setTempo(m_stretch_ratio); - - DEBUG_LOG(AUDIO, "Audio stretching: samples:%u/%u ratio:%f backlog:%f gain: %f", num_in, num_out, - m_stretch_ratio, backlog_fullness, m_lpf_gain); - - m_sound_touch.putSamples(in, num_in); - - const size_t samples_received = m_sound_touch.receiveSamples(out, num_out); - - if (samples_received != 0) - { - m_last_stretched_sample[0] = out[samples_received * 2 - 2]; - m_last_stretched_sample[1] = out[samples_received * 2 - 1]; - } - - // Preform padding if we've run out of samples. - for (size_t i = samples_received; i < num_out; i++) - { - out[i * 2 + 0] = m_last_stretched_sample[0]; - out[i * 2 + 1] = m_last_stretched_sample[1]; - } + return available_samples; } void CMixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples) diff --git a/Source/Core/AudioCommon/Mixer.h b/Source/Core/AudioCommon/Mixer.h index f2127cceba..2707df3383 100644 --- a/Source/Core/AudioCommon/Mixer.h +++ b/Source/Core/AudioCommon/Mixer.h @@ -7,12 +7,10 @@ #include #include +#include "AudioCommon/AudioStretcher.h" #include "AudioCommon/WaveFile.h" #include "Common/CommonTypes.h" -#include -#include - class CMixer final { public: @@ -21,6 +19,7 @@ public: // Called from audio threads unsigned int Mix(short* samples, unsigned int numSamples); + unsigned int MixSurround(float* samples, unsigned int num_samples); // Called from main thread void PushSamples(const short* samples, unsigned int num_samples); @@ -75,18 +74,15 @@ private: u32 m_frac = 0; }; - void StretchAudio(const short* in, unsigned int num_in, short* out, unsigned int num_out); - MixerFifo m_dma_mixer{this, 32000}; MixerFifo m_streaming_mixer{this, 48000}; MixerFifo m_wiimote_speaker_mixer{this, 3000}; unsigned int m_sampleRate; bool m_is_stretching = false; - soundtouch::SoundTouch m_sound_touch; - double m_stretch_ratio = 1.0; - std::array m_last_stretched_sample = {}; - std::array m_stretch_buffer; + AudioCommon::AudioStretcher m_stretcher; + std::array m_scratch_buffer; + std::array m_float_conversion_buffer; WaveFileWriter m_wave_writer_dtk; WaveFileWriter m_wave_writer_dsp; diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp index 3d4bc90e1d..ca343be005 100644 --- a/Source/Core/AudioCommon/OpenALStream.cpp +++ b/Source/Core/AudioCommon/OpenALStream.cpp @@ -6,7 +6,6 @@ #include #include -#include "AudioCommon/DPL2Decoder.h" #include "AudioCommon/OpenALStream.h" #include "AudioCommon/aldlist.h" #include "Common/Logging/Log.h" @@ -66,9 +65,6 @@ bool OpenALStream::Start() PanicAlertT("OpenAL: can't find sound devices"); } - // Initialize DPL2 parameters - DPL2Reset(); - return bReturn; } @@ -228,23 +224,18 @@ void OpenALStream::SoundLoop() numBuffersQueued -= numBuffersProcessed; } - // DPL2 accepts 240 samples minimum (FWRDURATION) - unsigned int minSamples = surround_capable ? 240 : 0; - unsigned int numSamples = OAL_MAX_SAMPLES; - numSamples = m_mixer->Mix(realtimeBuffer, numSamples); - - // Convert the samples from short to float - for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) - sampleBuffer[i] = static_cast(realtimeBuffer[i]) / (1 << 15); - - if (numSamples <= minSamples) - continue; if (surround_capable) { + // DPL2 accepts 240 samples minimum (FWRDURATION) + unsigned int minSamples = 240; + float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS]; - DPL2Decode(sampleBuffer, numSamples, dpl2); + numSamples = m_mixer->MixSurround(dpl2, numSamples); + + if (numSamples < minSamples) + continue; // zero-out the subwoofer channel - DPL2Decode generates a pretty // good 5.0 but not a good 5.1 output. Sadly there is not a 5.0 @@ -311,6 +302,15 @@ void OpenALStream::SoundLoop() } else { + numSamples = m_mixer->Mix(realtimeBuffer, numSamples); + + // Convert the samples from short to float + for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i) + sampleBuffer[i] = static_cast(realtimeBuffer[i]) / (1 << 15); + + if (!numSamples) + continue; + if (float32_capable) { alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, diff --git a/Source/Core/AudioCommon/PulseAudioStream.cpp b/Source/Core/AudioCommon/PulseAudioStream.cpp index e6e6a45ce2..827816d4e8 100644 --- a/Source/Core/AudioCommon/PulseAudioStream.cpp +++ b/Source/Core/AudioCommon/PulseAudioStream.cpp @@ -4,7 +4,6 @@ #include -#include "AudioCommon/DPL2Decoder.h" #include "AudioCommon/PulseAudioStream.h" #include "Common/CommonTypes.h" #include "Common/Logging/Log.h" @@ -30,9 +29,6 @@ bool PulseAudio::Start() m_run_thread.Set(); m_thread = std::thread(&PulseAudio::SoundLoop, this); - // Initialize DPL2 parameters - DPL2Reset(); - return true; } @@ -194,23 +190,12 @@ void PulseAudio::WriteCallback(pa_stream* s, size_t length) } else { - // get a floating point mix - s16 s16buffer_stereo[frames * 2]; - m_mixer->Mix(s16buffer_stereo, frames); // implicitly mixes to 16-bit stereo - - float floatbuffer_stereo[frames * 2]; - // s16 to float - for (int i = 0; i < frames * 2; ++i) - { - floatbuffer_stereo[i] = s16buffer_stereo[i] / float(1 << 15); - } - if (m_channels == 5) // Extract dpl2/5.0 Surround { float floatbuffer_6chan[frames * 6]; - // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR - DPL2Decode(floatbuffer_stereo, frames, floatbuffer_6chan); + m_mixer->MixSurround(floatbuffer_6chan, frames); + // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR // Discard the subwoofer channel - DPL2Decode generates a pretty // good 5.0 but not a good 5.1 output. const int dpl2_to_5chan[] = {0, 1, 2, 4, 5};