aboutsummaryrefslogtreecommitdiff
path: root/src/audio_core/sink/sink_stream.cpp
blob: 9dfd3bf747a5fbd46b7d636ae6cf6e7d015df09a (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
// SPDX-FileCopyrightText: Copyright 2026 Eden Emulator Project
// SPDX-License-Identifier: GPL-3.0-or-later

// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later

#include <algorithm>
#include <array>
#include <atomic>
#include <cstdlib>
#include <memory>
#include <span>
#include <vector>

#include "audio_core/audio_core.h"
#include "audio_core/common/common.h"
#include "audio_core/sink/sink_stream.h"
#include "common/common_types.h"
#include "common/fixed_point.h"
#include "common/scope_exit.h"
#include "common/settings.h"
#include "core/core.h"
#include "core/core_timing.h"

namespace AudioCore::Sink {

void SinkStream::AppendBuffer(SinkBuffer& buffer, std::span<s16> samples) {
    if (type == StreamType::In)
        return;

    constexpr s32 min = (std::numeric_limits<s16>::min)();
    constexpr s32 max = (std::numeric_limits<s16>::max)();

    auto yuzu_volume = Settings::Volume();
    if (yuzu_volume > 1.0f)
        yuzu_volume = 0.6f + 20.0f * std::log10(yuzu_volume);
    auto const volume = system_volume * device_volume * yuzu_volume;

    if (system_channels > device_channels) {
        static constexpr std::array<f32, 4> tcoeff{1.0f, 0.596f, 0.354f, 0.707f};
        for (u32 r_offs = 0, w_offs = 0; r_offs < samples.size(); r_offs += system_channels, w_offs += device_channels) {
            std::array<f32, 6> ccoeff{0.f};
            for (u32 i = 0; i < system_channels; ++i)
                ccoeff[i] = f32(samples[r_offs + i]);

            std::array<f32, 6> rcoeff{
                ccoeff[u32(Channels::FrontLeft)],
                ccoeff[u32(Channels::BackLeft)],
                ccoeff[u32(Channels::Center)],
                ccoeff[u32(Channels::LFE)],
                ccoeff[u32(Channels::BackRight)],
                ccoeff[u32(Channels::FrontRight)],
            };

            const f32 left  = rcoeff[0] * tcoeff[0] + rcoeff[2] * tcoeff[1] + rcoeff[3] * tcoeff[2] + rcoeff[1] * tcoeff[3];
            const f32 right = rcoeff[5] * tcoeff[0] + rcoeff[2] * tcoeff[1] + rcoeff[3] * tcoeff[2] + rcoeff[4] * tcoeff[3];

            samples[w_offs + 0] = s16(std::clamp(s32(left * volume),  min, max));
            samples[w_offs + 1] = s16(std::clamp(s32(right * volume), min, max));
        }

        queue.EmplaceWait(buffer);
        samples_buffer.Push(samples.subspan(0, samples.size() / system_channels * device_channels));
    } else if (system_channels < device_channels) {
        std::vector<s16> new_samples(samples.size() / system_channels * device_channels);
        for (u32 r_offs = 0, w_offs = 0; r_offs < samples.size(); r_offs += system_channels, w_offs += device_channels)
            for (u32 channel = 0; channel < system_channels; ++channel)
                new_samples[w_offs + channel] = s16(std::clamp(s32(f32(samples[r_offs + channel]) * volume), min, max));

        queue.EmplaceWait(buffer);
        samples_buffer.Push(new_samples);
    } else {
        if (volume != 1.0f) {
            for (u32 i = 0; i < samples.size(); ++i)
                samples[i] = s16(std::clamp(s32(f32(samples[i]) * volume), min, max));
        }

        queue.EmplaceWait(buffer);
        samples_buffer.Push(samples);
    }

    ++queued_buffers;
}

std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) {
    auto samples{samples_buffer.Pop(num_samples)};

    // TODO: Up-mix to 6 channels if the game expects it.
    // For audio input this is unlikely to ever be the case though.

    // Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
    // TODO: Play with this and find something that works better.
    constexpr s32 min = (std::numeric_limits<s16>::min)();
    constexpr s32 max = (std::numeric_limits<s16>::max)();
    auto volume{system_volume * device_volume * 8};
    for (u32 i = 0; i < samples.size(); i++)
        samples[i] = s16(std::clamp(s32(f32(samples[i]) * volume), min, max));

    if (samples.size() < num_samples)
        samples.resize(num_samples, 0);
    return samples;
}

void SinkStream::ClearQueue() {
    std::scoped_lock lk{release_mutex};

    samples_buffer.Pop();
    SinkBuffer tmp;
    while (queue.TryPop(tmp));

    queued_buffers = 0;
    playing_buffer = {};
    playing_buffer.consumed = true;
}

void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) {
    const std::size_t num_channels = GetDeviceChannels();
    const std::size_t frame_size = num_channels;
    const std::size_t frame_size_bytes = frame_size * sizeof(s16);
    size_t frames_written{0};

    // If we're paused or going to shut down, we don't want to consume buffers as coretiming is
    // paused and we'll desync, so just return.
    if (system.IsPaused() || system.IsShuttingDown())
        return;

    while (frames_written < num_frames) {
        // If the playing buffer has been consumed or has no frames, we need a new one
        if (playing_buffer.consumed || playing_buffer.frames == 0) {
            if (!queue.TryPop(playing_buffer)) {
                // If no buffer was available we've underrun, just push the samples and
                // continue.
                samples_buffer.Push(&input_buffer[frames_written * frame_size], (num_frames - frames_written) * frame_size);
                frames_written = num_frames;
                continue;
            }
            // Successfully dequeued a new buffer.
            queued_buffers--;
        }

        // Get the minimum frames available between the currently playing buffer, and the
        // amount we have left to fill
        size_t frames_available{std::min<u64>(playing_buffer.frames - playing_buffer.frames_played, num_frames - frames_written)};

        samples_buffer.Push(&input_buffer[frames_written * frame_size], frames_available * frame_size);

        frames_written += frames_available;
        playing_buffer.frames_played += frames_available;

        // If that's all the frames in the current buffer, add its samples and mark it as
        // consumed
        if (playing_buffer.frames_played >= playing_buffer.frames)
            playing_buffer.consumed = true;
    }

    if (frames_written > 0) {
        std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes);
    }

    // update sample counts fuer audio-ins
    {
        std::scoped_lock lk{sample_count_lock};
        last_sample_count_update_time = system.CoreTiming().GetGlobalTimeNs();
        min_played_sample_count = max_played_sample_count;
        max_played_sample_count += frames_written;
    }
}

void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) {
    const std::size_t num_channels = GetDeviceChannels();
    const std::size_t frame_size = num_channels;
    const std::size_t frame_size_bytes = frame_size * sizeof(s16);
    size_t frames_written{0};
    size_t actual_frames_written{0};

    if (system.IsPaused() || system.IsShuttingDown()) {
        if (system.IsShuttingDown()) {
            std::scoped_lock lk{release_mutex};
            queued_buffers.store(0);
            release_cv.notify_one();
        }
        for (size_t i = 0; i < num_frames; i++)
            std::memset(&output_buffer[i * frame_size], 0, frame_size_bytes);
        return;
    }

    while (frames_written < num_frames) {
        if (playing_buffer.consumed || playing_buffer.frames == 0) {
            std::unique_lock lk{release_mutex};

            if (!queue.TryPop(playing_buffer)) {
                lk.unlock();
                for (size_t i = frames_written; i < num_frames; i++)
                    std::memcpy(&output_buffer[i * frame_size], last_frame.data(), frame_size_bytes);
                frames_written = num_frames;
                continue;
            }

            --queued_buffers;
            lk.unlock();
            release_cv.notify_one();
        }

        const size_t frames_available = std::min<u64>(playing_buffer.frames - playing_buffer.frames_played, num_frames - frames_written);

        samples_buffer.Pop(&output_buffer[frames_written * frame_size], frames_available * frame_size);

        frames_written += frames_available;
        actual_frames_written += frames_available;
        playing_buffer.frames_played += frames_available;

        if (playing_buffer.frames_played >= playing_buffer.frames)
            playing_buffer.consumed = true;
    }

    std::memcpy(last_frame.data(), &output_buffer[(frames_written - 1) * frame_size], frame_size_bytes);

    // EDEN_NX2_AUDIO_LEVEL: log the output peak about every two seconds (is a game audible?).
    if (static const bool meter = std::getenv("EDEN_NX2_AUDIO_LEVEL") != nullptr; meter) {
        for (size_t i = 0; i < num_frames * frame_size; ++i) {
            level_peak = std::max<s32>(level_peak, std::abs(static_cast<s32>(output_buffer[i])));
        }
        level_frames += num_frames;
        if (level_frames >= TargetSampleRate * 2) {
            LOG_INFO(Audio, "Audio output peak over {} frames: {} of 32767", level_frames, level_peak);
            level_frames = 0;
            level_peak = 0;
        }
    }

    {
        std::scoped_lock lk{sample_count_lock};
        last_sample_count_update_time = system.CoreTiming().GetGlobalTimeNs();
        min_played_sample_count = max_played_sample_count;
        max_played_sample_count += actual_frames_written;
    }
}

u64 SinkStream::GetExpectedPlayedSampleCount() {
    std::scoped_lock lk{sample_count_lock};
    auto cur_time{system.CoreTiming().GetGlobalTimeNs()};
    auto time_delta{cur_time - last_sample_count_update_time};
    auto exp_played_sample_count{min_played_sample_count + (TargetSampleRate * time_delta) / std::chrono::seconds{1}};

    // Add 25ms of latency in sample reporting to allow for some leeway in scheduler timings
    return std::min<u64>(exp_played_sample_count, max_played_sample_count) + TargetSampleCount * 5;
}

void SinkStream::WaitFreeSpace(std::stop_token stop_token) {
    std::unique_lock lk{release_mutex};

    auto can_continue = [this]() {
        return paused || queued_buffers < max_queue_size;
    };

    release_cv.wait_for(lk, std::chrono::milliseconds(7), can_continue);

    if (queued_buffers > max_queue_size + 3) {
        release_cv.wait(lk, stop_token, can_continue);
    }
}

void SinkStream::SignalPause() {
    {
        std::scoped_lock lk{release_mutex};
        paused = true;
    }
    release_cv.notify_one();
}

} // namespace AudioCore::Sink