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author | MerryMage <MerryMage@users.noreply.github.com> | 2018-09-08 17:48:13 +0200 |
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committer | MerryMage <MerryMage@users.noreply.github.com> | 2018-09-08 19:56:38 +0200 |
commit | e51bd49f87052d0706a016fab88d18ffef05b8b1 (patch) | |
tree | 76eccf3f2b428ada51c8025bc3ab7c512263854d /src/audio_core/time_stretch.cpp | |
parent | cubeb_sink: Hold last available value instead of writing zeros (diff) | |
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Diffstat (limited to '')
-rw-r--r-- | src/audio_core/time_stretch.cpp | 64 |
1 files changed, 64 insertions, 0 deletions
diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp new file mode 100644 index 000000000..17e128323 --- /dev/null +++ b/src/audio_core/time_stretch.cpp @@ -0,0 +1,64 @@ +// Copyright 2018 yuzu Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <algorithm> +#include <cmath> +#include <cstddef> +#include "audio_core/time_stretch.h" +#include "common/logging/log.h" + +namespace AudioCore { + +TimeStretcher::TimeStretcher(u32 sample_rate, u32 channel_count) + : m_sample_rate(sample_rate), m_channel_count(channel_count) { + m_sound_touch.setChannels(channel_count); + m_sound_touch.setSampleRate(sample_rate); + m_sound_touch.setPitch(1.0); + m_sound_touch.setTempo(1.0); +} + +void TimeStretcher::Clear() { + m_sound_touch.clear(); +} + +size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t 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 = 0.3; // seconds + 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 = 0.05; // seconds + const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale); + current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0); + + // 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 = 2.0; // 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); + + LOG_DEBUG(Audio, "Audio Stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in, num_out, + m_stretch_ratio, backlog_fullness, lpf_gain); + + m_sound_touch.putSamples(in, num_in); + return m_sound_touch.receiveSamples(out, num_out); +} + +} // namespace AudioCore |