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Diffstat (limited to 'Source/Core/AudioCommon/AudioStretcher.cpp')
| -rw-r--r-- | Source/Core/AudioCommon/AudioStretcher.cpp | 85 |
1 files changed, 0 insertions, 85 deletions
diff --git a/Source/Core/AudioCommon/AudioStretcher.cpp b/Source/Core/AudioCommon/AudioStretcher.cpp deleted file mode 100644 index a5e54b8163..0000000000 --- a/Source/Core/AudioCommon/AudioStretcher.cpp +++ /dev/null @@ -1,85 +0,0 @@ -// Copyright 2017 Dolphin Emulator Project -// SPDX-License-Identifier: GPL-2.0-or-later - -#include "AudioCommon/AudioStretcher.h" - -#include <algorithm> -#include <cmath> -#include <cstddef> - -#include "Common/Logging/Log.h" -#include "Core/Config/MainSettings.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); -} - -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 = Config::Get(Config::MAIN_AUDIO_STRETCH_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_FMT(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); -} - -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 |
