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Diffstat (limited to 'Source/Core/AudioCommon/AudioStretcher.cpp')
| -rw-r--r-- | Source/Core/AudioCommon/AudioStretcher.cpp | 89 |
1 files changed, 89 insertions, 0 deletions
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 <algorithm> +#include <cmath> +#include <cstddef> + +#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<double>(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<double>(num_in) / static_cast<double>(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 |
