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471 lines
15 KiB
C++
471 lines
15 KiB
C++
// Copyright 2008 Dolphin Emulator Project
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// SPDX-License-Identifier: GPL-2.0-or-later
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#include "AudioCommon/Mixer.h"
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#include <algorithm>
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#include <cmath>
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#include <cstring>
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#include "AudioCommon/Enums.h"
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#include "Common/ChunkFile.h"
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#include "Common/CommonTypes.h"
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#include "Common/Logging/Log.h"
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#include "Common/Swap.h"
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#include "Core/Config/MainSettings.h"
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#include "Core/ConfigManager.h"
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#include "VideoCommon/PerformanceMetrics.h"
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static u32 DPL2QualityToFrameBlockSize(AudioCommon::DPL2Quality quality)
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{
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switch (quality)
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{
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case AudioCommon::DPL2Quality::Lowest:
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return 512;
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case AudioCommon::DPL2Quality::Low:
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return 1024;
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case AudioCommon::DPL2Quality::Highest:
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return 4096;
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default:
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return 2048;
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}
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}
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Mixer::Mixer(unsigned int BackendSampleRate)
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: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate),
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m_surround_decoder(BackendSampleRate,
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DPL2QualityToFrameBlockSize(Config::Get(Config::MAIN_DPL2_QUALITY)))
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{
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m_config_changed_callback_id = Config::AddConfigChangedCallback([this] { RefreshConfig(); });
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RefreshConfig();
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INFO_LOG_FMT(AUDIO_INTERFACE, "Mixer is initialized");
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}
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Mixer::~Mixer()
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{
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Config::RemoveConfigChangedCallback(m_config_changed_callback_id);
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}
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void Mixer::DoState(PointerWrap& p)
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{
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m_dma_mixer.DoState(p);
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m_streaming_mixer.DoState(p);
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m_wiimote_speaker_mixer.DoState(p);
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for (auto& mixer : m_gba_mixers)
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mixer.DoState(p);
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}
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// Executed from sound stream thread
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unsigned int Mixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
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bool consider_framelimit, float emulationspeed,
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int timing_variance)
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{
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unsigned int currentSample = 0;
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// Cache access in non-volatile variable
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// This is the only function changing the read value, so it's safe to
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// cache it locally although it's written here.
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// The writing pointer will be modified outside, but it will only increase,
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// so we will just ignore new written data while interpolating.
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// Without this cache, the compiler wouldn't be allowed to optimize the
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// interpolation loop.
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u32 indexR = m_indexR.load();
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u32 indexW = m_indexW.load();
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// render numleft sample pairs to samples[]
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// advance indexR with sample position
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// remember fractional offset
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float aid_sample_rate =
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FIXED_SAMPLE_RATE_DIVIDEND / static_cast<float>(m_input_sample_rate_divisor);
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if (consider_framelimit && emulationspeed > 0.0f)
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{
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float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
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u32 low_watermark = (FIXED_SAMPLE_RATE_DIVIDEND * timing_variance) /
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(static_cast<u64>(m_input_sample_rate_divisor) * 1000);
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low_watermark = std::min(low_watermark, MAX_SAMPLES / 2);
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m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
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float offset = (m_numLeftI - low_watermark) * CONTROL_FACTOR;
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if (offset > MAX_FREQ_SHIFT)
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offset = MAX_FREQ_SHIFT;
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if (offset < -MAX_FREQ_SHIFT)
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offset = -MAX_FREQ_SHIFT;
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aid_sample_rate = (aid_sample_rate + offset) * emulationspeed;
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}
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const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
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s32 lvolume = m_LVolume.load();
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s32 rvolume = m_RVolume.load();
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const auto read_buffer = [this](auto index) {
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return m_little_endian ? m_buffer[index] : Common::swap16(m_buffer[index]);
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};
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// TODO: consider a higher-quality resampling algorithm.
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for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
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{
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u32 indexR2 = indexR + 2; // next sample
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s16 l1 = read_buffer(indexR & INDEX_MASK); // current
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s16 l2 = read_buffer(indexR2 & INDEX_MASK); // next
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int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
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sampleL = (sampleL * lvolume) >> 8;
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sampleL += samples[currentSample + 1];
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samples[currentSample + 1] = std::clamp(sampleL, -32767, 32767);
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s16 r1 = read_buffer((indexR + 1) & INDEX_MASK); // current
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s16 r2 = read_buffer((indexR2 + 1) & INDEX_MASK); // next
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int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
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sampleR = (sampleR * rvolume) >> 8;
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sampleR += samples[currentSample];
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samples[currentSample] = std::clamp(sampleR, -32767, 32767);
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m_frac += ratio;
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indexR += 2 * (u16)(m_frac >> 16);
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m_frac &= 0xffff;
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}
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// Actual number of samples written to the buffer without padding.
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unsigned int actual_sample_count = currentSample / 2;
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// Padding
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short s[2];
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s[0] = read_buffer((indexR - 1) & INDEX_MASK);
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s[1] = read_buffer((indexR - 2) & INDEX_MASK);
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s[0] = (s[0] * rvolume) >> 8;
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s[1] = (s[1] * lvolume) >> 8;
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for (; currentSample < numSamples * 2; currentSample += 2)
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{
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int sampleR = std::clamp(s[0] + samples[currentSample + 0], -32767, 32767);
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int sampleL = std::clamp(s[1] + samples[currentSample + 1], -32767, 32767);
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samples[currentSample + 0] = sampleR;
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samples[currentSample + 1] = sampleL;
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}
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// Flush cached variable
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m_indexR.store(indexR);
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return actual_sample_count;
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}
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unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
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{
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if (!samples)
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return 0;
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memset(samples, 0, num_samples * 2 * sizeof(short));
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// TODO: Determine how emulation speed will be used in audio
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// const float emulation_speed = std::roundf(g_perf_metrics.GetSpeed()) / 100.f;
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const float emulation_speed = m_config_emulation_speed;
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const int timing_variance = m_config_timing_variance;
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if (m_config_audio_stretch)
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{
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unsigned int available_samples =
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std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
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ASSERT_MSG(AUDIO, available_samples <= MAX_SAMPLES,
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"Audio stretching would overflow m_scratch_buffer: min({}, {}) -> {} > {} ({})",
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m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples(),
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available_samples, MAX_SAMPLES, num_samples);
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m_scratch_buffer.fill(0);
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m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
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timing_variance);
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m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
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timing_variance);
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m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
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timing_variance);
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for (auto& mixer : m_gba_mixers)
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{
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mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
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timing_variance);
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}
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if (!m_is_stretching)
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{
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m_stretcher.Clear();
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m_is_stretching = true;
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}
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m_stretcher.ProcessSamples(m_scratch_buffer.data(), available_samples, num_samples);
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m_stretcher.GetStretchedSamples(samples, num_samples);
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}
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else
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{
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m_dma_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
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m_streaming_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
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m_wiimote_speaker_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
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for (auto& mixer : m_gba_mixers)
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mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
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m_is_stretching = false;
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}
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return num_samples;
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}
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unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
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{
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if (!num_samples)
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return 0;
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memset(samples, 0, num_samples * SURROUND_CHANNELS * sizeof(float));
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size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
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// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads
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// and writes.
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ASSERT_MSG(AUDIO, needed_frames <= MAX_SAMPLES,
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"needed_frames would overflow m_scratch_buffer: {} -> {} > {}", num_samples,
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needed_frames, MAX_SAMPLES);
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size_t available_frames = Mix(m_scratch_buffer.data(), static_cast<u32>(needed_frames));
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if (available_frames != needed_frames)
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{
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ERROR_LOG_FMT(AUDIO,
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"Error decoding surround frames: needed {} frames for {} samples but got {}",
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needed_frames, num_samples, available_frames);
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return 0;
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}
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m_surround_decoder.PutFrames(m_scratch_buffer.data(), needed_frames);
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m_surround_decoder.ReceiveFrames(samples, num_samples);
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return num_samples;
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}
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void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
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{
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// Cache access in non-volatile variable
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// indexR isn't allowed to cache in the audio throttling loop as it
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// needs to get updates to not deadlock.
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u32 indexW = m_indexW.load();
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// Check if we have enough free space
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// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
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if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
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return;
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// AyuanX: Actual re-sampling work has been moved to sound thread
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// to alleviate the workload on main thread
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// and we simply store raw data here to make fast mem copy
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int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
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if (over_bytes > 0)
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
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memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
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}
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else
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{
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memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
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}
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m_indexW.fetch_add(num_samples * 2);
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}
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void Mixer::PushSamples(const short* samples, unsigned int num_samples)
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{
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m_dma_mixer.PushSamples(samples, num_samples);
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if (m_log_dsp_audio)
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{
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int sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
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auto volume = m_dma_mixer.GetVolume();
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m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
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volume.second);
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}
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}
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void Mixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
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{
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m_streaming_mixer.PushSamples(samples, num_samples);
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if (m_log_dtk_audio)
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{
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int sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
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auto volume = m_streaming_mixer.GetVolume();
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m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
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volume.second);
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}
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}
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void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
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unsigned int sample_rate_divisor)
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{
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// Max 20 bytes/speaker report, may be 4-bit ADPCM so multiply by 2
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static constexpr u32 MAX_SPEAKER_SAMPLES = 20 * 2;
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std::array<short, MAX_SPEAKER_SAMPLES * 2> samples_stereo;
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ASSERT_MSG(AUDIO, num_samples <= MAX_SPEAKER_SAMPLES,
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"num_samples would overflow samples_stereo: {} > {}", num_samples,
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MAX_SPEAKER_SAMPLES);
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if (num_samples <= MAX_SPEAKER_SAMPLES)
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{
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m_wiimote_speaker_mixer.SetInputSampleRateDivisor(sample_rate_divisor);
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for (unsigned int i = 0; i < num_samples; ++i)
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{
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samples_stereo[i * 2] = samples[i];
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samples_stereo[i * 2 + 1] = samples[i];
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}
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m_wiimote_speaker_mixer.PushSamples(samples_stereo.data(), num_samples);
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}
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}
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void Mixer::PushGBASamples(int device_number, const short* samples, unsigned int num_samples)
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{
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m_gba_mixers[device_number].PushSamples(samples, num_samples);
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}
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void Mixer::SetDMAInputSampleRateDivisor(unsigned int rate_divisor)
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{
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m_dma_mixer.SetInputSampleRateDivisor(rate_divisor);
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}
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void Mixer::SetStreamInputSampleRateDivisor(unsigned int rate_divisor)
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{
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m_streaming_mixer.SetInputSampleRateDivisor(rate_divisor);
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}
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void Mixer::SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor)
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{
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m_gba_mixers[device_number].SetInputSampleRateDivisor(rate_divisor);
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}
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void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
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{
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m_streaming_mixer.SetVolume(lvolume, rvolume);
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}
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void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
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{
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m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
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}
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void Mixer::SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume)
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{
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m_gba_mixers[device_number].SetVolume(lvolume, rvolume);
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}
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void Mixer::StartLogDTKAudio(const std::string& filename)
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{
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if (!m_log_dtk_audio)
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{
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bool success = m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRateDivisor());
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if (success)
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{
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m_log_dtk_audio = true;
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m_wave_writer_dtk.SetSkipSilence(false);
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NOTICE_LOG_FMT(AUDIO, "Starting DTK Audio logging");
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}
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else
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{
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m_wave_writer_dtk.Stop();
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NOTICE_LOG_FMT(AUDIO, "Unable to start DTK Audio logging");
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}
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}
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else
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{
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WARN_LOG_FMT(AUDIO, "DTK Audio logging has already been started");
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}
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}
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void Mixer::StopLogDTKAudio()
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{
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if (m_log_dtk_audio)
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{
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m_log_dtk_audio = false;
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m_wave_writer_dtk.Stop();
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NOTICE_LOG_FMT(AUDIO, "Stopping DTK Audio logging");
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}
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else
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{
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WARN_LOG_FMT(AUDIO, "DTK Audio logging has already been stopped");
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}
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}
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void Mixer::StartLogDSPAudio(const std::string& filename)
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{
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if (!m_log_dsp_audio)
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{
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bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRateDivisor());
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if (success)
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{
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m_log_dsp_audio = true;
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m_wave_writer_dsp.SetSkipSilence(false);
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NOTICE_LOG_FMT(AUDIO, "Starting DSP Audio logging");
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}
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else
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{
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m_wave_writer_dsp.Stop();
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NOTICE_LOG_FMT(AUDIO, "Unable to start DSP Audio logging");
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}
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}
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else
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{
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WARN_LOG_FMT(AUDIO, "DSP Audio logging has already been started");
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}
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}
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void Mixer::StopLogDSPAudio()
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{
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if (m_log_dsp_audio)
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{
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m_log_dsp_audio = false;
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m_wave_writer_dsp.Stop();
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NOTICE_LOG_FMT(AUDIO, "Stopping DSP Audio logging");
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}
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else
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{
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WARN_LOG_FMT(AUDIO, "DSP Audio logging has already been stopped");
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}
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}
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void Mixer::RefreshConfig()
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{
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m_config_emulation_speed = Config::Get(Config::MAIN_EMULATION_SPEED);
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m_config_timing_variance = Config::Get(Config::MAIN_TIMING_VARIANCE);
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m_config_audio_stretch = Config::Get(Config::MAIN_AUDIO_STRETCH);
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}
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void Mixer::MixerFifo::DoState(PointerWrap& p)
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{
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p.Do(m_input_sample_rate_divisor);
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p.Do(m_LVolume);
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p.Do(m_RVolume);
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}
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void Mixer::MixerFifo::SetInputSampleRateDivisor(unsigned int rate_divisor)
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{
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m_input_sample_rate_divisor = rate_divisor;
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}
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unsigned int Mixer::MixerFifo::GetInputSampleRateDivisor() const
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{
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return m_input_sample_rate_divisor;
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}
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void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
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{
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m_LVolume.store(lvolume + (lvolume >> 7));
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m_RVolume.store(rvolume + (rvolume >> 7));
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}
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std::pair<s32, s32> Mixer::MixerFifo::GetVolume() const
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{
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return std::make_pair(m_LVolume.load(), m_RVolume.load());
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}
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unsigned int Mixer::MixerFifo::AvailableSamples() const
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{
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unsigned int samples_in_fifo = ((m_indexW.load() - m_indexR.load()) & INDEX_MASK) / 2;
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if (samples_in_fifo <= 1)
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return 0; // Mixer::MixerFifo::Mix always keeps one sample in the buffer.
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return (samples_in_fifo - 1) * static_cast<u64>(m_mixer->m_sampleRate) *
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m_input_sample_rate_divisor / FIXED_SAMPLE_RATE_DIVIDEND;
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}
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