summaryrefslogtreecommitdiffstats
path: root/src/audio_core/command_generator.cpp
blob: 440bfc140685dc0f72443190b62f037ffa102b0a (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
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
// Copyright 2020 yuzu Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.

#include "audio_core/algorithm/interpolate.h"
#include "audio_core/command_generator.h"
#include "audio_core/mix_context.h"
#include "audio_core/voice_context.h"
#include "core/memory.h"

namespace AudioCore {
namespace {
constexpr std::size_t MIX_BUFFER_SIZE = 0x3f00;
constexpr std::size_t SCALED_MIX_BUFFER_SIZE = MIX_BUFFER_SIZE << 15ULL;

template <std::size_t N>
void ApplyMix(s32* output, const s32* input, s32 gain, s32 sample_count) {
    for (std::size_t i = 0; i < static_cast<std::size_t>(sample_count); i += N) {
        for (std::size_t j = 0; j < N; j++) {
            output[i + j] +=
                static_cast<s32>((static_cast<s64>(input[i + j]) * gain + 0x4000) >> 15);
        }
    }
}

s32 ApplyMixRamp(s32* output, const s32* input, float gain, float delta, s32 sample_count) {
    s32 x = 0;
    for (s32 i = 0; i < sample_count; i++) {
        x = static_cast<s32>(static_cast<float>(input[i]) * gain);
        output[i] += x;
        gain += delta;
    }
    return x;
}

void ApplyGain(s32* output, const s32* input, s32 gain, s32 delta, s32 sample_count) {
    for (s32 i = 0; i < sample_count; i++) {
        output[i] = static_cast<s32>((static_cast<s64>(input[i]) * gain + 0x4000) >> 15);
        gain += delta;
    }
}

void ApplyGainWithoutDelta(s32* output, const s32* input, s32 gain, s32 sample_count) {
    for (s32 i = 0; i < sample_count; i++) {
        output[i] = static_cast<s32>((static_cast<s64>(input[i]) * gain + 0x4000) >> 15);
    }
}

} // namespace

CommandGenerator::CommandGenerator(AudioCommon::AudioRendererParameter& worker_params,
                                   VoiceContext& voice_context, MixContext& mix_context,
                                   SplitterContext& splitter_context, Core::Memory::Memory& memory)
    : worker_params(worker_params), voice_context(voice_context), mix_context(mix_context),
      splitter_context(splitter_context), memory(memory),
      mix_buffer((worker_params.mix_buffer_count + AudioCommon::MAX_CHANNEL_COUNT) *
                 worker_params.sample_count),
      sample_buffer(MIX_BUFFER_SIZE) {}
CommandGenerator::~CommandGenerator() = default;

void CommandGenerator::ClearMixBuffers() {
    std::fill(mix_buffer.begin(), mix_buffer.end(), 0);
    std::fill(sample_buffer.begin(), sample_buffer.end(), 0);
}

void CommandGenerator::GenerateVoiceCommands() {
    if (dumping_frame) {
        LOG_DEBUG(Audio, "(DSP_TRACE) GenerateVoiceCommands");
    }
    // Grab all our voices
    const auto voice_count = voice_context.GetVoiceCount();
    for (std::size_t i = 0; i < voice_count; i++) {
        auto& voice_info = voice_context.GetSortedInfo(i);
        // Update voices and check if we should queue them
        if (voice_info.ShouldSkip() || !voice_info.UpdateForCommandGeneration(voice_context)) {
            continue;
        }

        // Queue our voice
        GenerateVoiceCommand(voice_info);
    }
    // Update our splitters
    splitter_context.UpdateInternalState();
}

void CommandGenerator::GenerateVoiceCommand(ServerVoiceInfo& voice_info) {
    auto& in_params = voice_info.GetInParams();
    const auto channel_count = in_params.channel_count;

    for (s32 channel = 0; channel < channel_count; channel++) {
        const auto resource_id = in_params.voice_channel_resource_id[channel];
        auto& dsp_state = voice_context.GetDspSharedState(resource_id);
        auto& channel_resource = voice_context.GetChannelResource(resource_id);

        // Decode our samples for our channel
        GenerateDataSourceCommand(voice_info, dsp_state, channel);

        if (in_params.should_depop) {
            in_params.last_volume = 0.0f;
        } else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER ||
                   in_params.mix_id != AudioCommon::NO_MIX) {
            // Apply a biquad filter if needed
            GenerateBiquadFilterCommandForVoice(voice_info, dsp_state,
                                                worker_params.mix_buffer_count, channel);
            // Base voice volume ramping
            GenerateVolumeRampCommand(in_params.last_volume, in_params.volume, channel,
                                      in_params.node_id);
            in_params.last_volume = in_params.volume;

            if (in_params.mix_id != AudioCommon::NO_MIX) {
                // If we're using a mix id
                auto& mix_info = mix_context.GetInfo(in_params.mix_id);
                const auto& dest_mix_params = mix_info.GetInParams();

                // Voice Mixing
                GenerateVoiceMixCommand(
                    channel_resource.GetCurrentMixVolume(), channel_resource.GetLastMixVolume(),
                    dsp_state, dest_mix_params.buffer_offset, dest_mix_params.buffer_count,
                    worker_params.mix_buffer_count + channel, in_params.node_id);

                // Update last mix volumes
                channel_resource.UpdateLastMixVolumes();
            } else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER) {
                s32 base = channel;
                while (auto* destination_data =
                           GetDestinationData(in_params.splitter_info_id, base)) {
                    base += channel_count;

                    if (!destination_data->IsConfigured()) {
                        continue;
                    }
                    if (destination_data->GetMixId() >= mix_context.GetCount()) {
                        continue;
                    }

                    const auto& mix_info = mix_context.GetInfo(destination_data->GetMixId());
                    const auto& dest_mix_params = mix_info.GetInParams();
                    GenerateVoiceMixCommand(
                        destination_data->CurrentMixVolumes(), destination_data->LastMixVolumes(),
                        dsp_state, dest_mix_params.buffer_offset, dest_mix_params.buffer_count,
                        worker_params.mix_buffer_count + channel, in_params.node_id);
                    destination_data->MarkDirty();
                }
            }
        }

        // Update biquad filter enabled states
        for (std::size_t i = 0; i < AudioCommon::MAX_BIQUAD_FILTERS; i++) {
            in_params.was_biquad_filter_enabled[i] = in_params.biquad_filter[i].enabled;
        }
    }
}

void CommandGenerator::GenerateSubMixCommands() {
    const auto mix_count = mix_context.GetCount();
    for (std::size_t i = 0; i < mix_count; i++) {
        auto& mix_info = mix_context.GetSortedInfo(i);
        const auto& in_params = mix_info.GetInParams();
        if (!in_params.in_use || in_params.mix_id == AudioCommon::FINAL_MIX) {
            continue;
        }
        GenerateSubMixCommand(mix_info);
    }
}

void CommandGenerator::GenerateFinalMixCommands() {
    GenerateFinalMixCommand();
}

void CommandGenerator::PreCommand() {
    if (!dumping_frame) {
        return;
    }
    for (std::size_t i = 0; i < splitter_context.GetInfoCount(); i++) {
        const auto& base = splitter_context.GetInfo(i);
        std::string graph = fmt::format("b[{}]", i);
        auto* head = base.GetHead();
        while (head != nullptr) {
            graph += fmt::format("->{}", head->GetMixId());
            head = head->GetNextDestination();
        }
        LOG_DEBUG(Audio, "(DSP_TRACE) SplitterGraph splitter_info={}, {}", i, graph);
    }
}

void CommandGenerator::PostCommand() {
    if (!dumping_frame) {
        return;
    }
    dumping_frame = false;
}

void CommandGenerator::GenerateDataSourceCommand(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
                                                 s32 channel) {
    auto& in_params = voice_info.GetInParams();
    const auto depop = in_params.should_depop;

    if (in_params.mix_id != AudioCommon::NO_MIX) {
        [[maybe_unused]] auto& mix_info = mix_context.GetInfo(in_params.mix_id);
        // mix_info.
        // TODO(ogniK): Depop to destination mix
    } else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER) {
        // TODO(ogniK): Depop to splitter
    }

    if (depop) {
        return;
    }

    switch (in_params.sample_format) {
    case SampleFormat::Pcm16:
        DecodeFromWaveBuffers(voice_info, GetChannelMixBuffer(channel), dsp_state, channel,
                              worker_params.sample_rate, worker_params.sample_count,
                              in_params.node_id);
        break;
    case SampleFormat::Adpcm:
        ASSERT(channel == 0 && in_params.channel_count == 1);
        DecodeFromWaveBuffers(voice_info, GetChannelMixBuffer(0), dsp_state, 0,
                              worker_params.sample_rate, worker_params.sample_count,
                              in_params.node_id);
        break;
    default:
        UNREACHABLE_MSG("Unimplemented sample format={}", in_params.sample_format);
    }
}

void CommandGenerator::GenerateBiquadFilterCommandForVoice(ServerVoiceInfo& voice_info,
                                                           VoiceState& dsp_state,
                                                           s32 mix_buffer_count, s32 channel) {
    for (std::size_t i = 0; i < AudioCommon::MAX_BIQUAD_FILTERS; i++) {
        const auto& in_params = voice_info.GetInParams();
        auto& biquad_filter = in_params.biquad_filter[i];
        // Check if biquad filter is actually used
        if (!biquad_filter.enabled) {
            continue;
        }

        // Reinitialize our biquad filter state if it was enabled previously
        if (!in_params.was_biquad_filter_enabled[i]) {
            dsp_state.biquad_filter_state.fill(0);
        }

        // Generate biquad filter
        GenerateBiquadFilterCommand(mix_buffer_count, biquad_filter, dsp_state.biquad_filter_state,
                                    mix_buffer_count + channel, mix_buffer_count + channel,
                                    worker_params.sample_count, voice_info.GetInParams().node_id);
    }
}

void AudioCore::CommandGenerator::GenerateBiquadFilterCommand(
    s32 mix_buffer, const BiquadFilterParameter& params, std::array<s64, 2>& state,
    std::size_t input_offset, std::size_t output_offset, s32 sample_count, s32 node_id) {
    if (dumping_frame) {
        LOG_DEBUG(Audio,
                  "(DSP_TRACE) GenerateBiquadFilterCommand node_id={}, "
                  "input_mix_buffer={}, output_mix_buffer={}",
                  node_id, input_offset, output_offset);
    }
    const auto* input = GetMixBuffer(input_offset);
    auto* output = GetMixBuffer(output_offset);

    // Biquad filter parameters
    const auto [n0, n1, n2] = params.numerator;
    const auto [d0, d1] = params.denominator;

    // Biquad filter states
    auto [s0, s1] = state;

    constexpr s64 int32_min = std::numeric_limits<s32>::min();
    constexpr s64 int32_max = std::numeric_limits<s32>::max();

    for (int i = 0; i < sample_count; ++i) {
        const auto sample = static_cast<s64>(input[i]);
        const auto f = (sample * n0 + s0 + 0x4000) >> 15;
        const auto y = std::clamp(f, int32_min, int32_max);
        s0 = sample * n1 + y * d0 + s1;
        s1 = sample * n2 + y * d1;
        output[i] = static_cast<s32>(y);
    }

    state = {s0, s1};
}

ServerSplitterDestinationData* CommandGenerator::GetDestinationData(s32 splitter_id, s32 index) {
    if (splitter_id == AudioCommon::NO_SPLITTER) {
        return nullptr;
    }
    return splitter_context.GetDestinationData(splitter_id, index);
}

void CommandGenerator::GenerateVolumeRampCommand(float last_volume, float current_volume,
                                                 s32 channel, s32 node_id) {
    const auto last = static_cast<s32>(last_volume * 32768.0f);
    const auto current = static_cast<s32>(current_volume * 32768.0f);
    const auto delta = static_cast<s32>((static_cast<float>(current) - static_cast<float>(last)) /
                                        static_cast<float>(worker_params.sample_count));

    if (dumping_frame) {
        LOG_DEBUG(Audio,
                  "(DSP_TRACE) GenerateVolumeRampCommand node_id={}, input={}, output={}, "
                  "last_volume={}, current_volume={}",
                  node_id, GetMixChannelBufferOffset(channel), GetMixChannelBufferOffset(channel),
                  last_volume, current_volume);
    }
    // Apply generic gain on samples
    ApplyGain(GetChannelMixBuffer(channel), GetChannelMixBuffer(channel), last, delta,
              worker_params.sample_count);
}

void CommandGenerator::GenerateVoiceMixCommand(const MixVolumeBuffer& mix_volumes,
                                               const MixVolumeBuffer& last_mix_volumes,
                                               VoiceState& dsp_state, s32 mix_buffer_offset,
                                               s32 mix_buffer_count, s32 voice_index, s32 node_id) {
    // Loop all our mix buffers
    for (s32 i = 0; i < mix_buffer_count; i++) {
        if (last_mix_volumes[i] != 0.0f || mix_volumes[i] != 0.0f) {
            const auto delta = static_cast<float>((mix_volumes[i] - last_mix_volumes[i])) /
                               static_cast<float>(worker_params.sample_count);

            if (dumping_frame) {
                LOG_DEBUG(Audio,
                          "(DSP_TRACE) GenerateVoiceMixCommand node_id={}, input={}, "
                          "output={}, last_volume={}, current_volume={}",
                          node_id, voice_index, mix_buffer_offset + i, last_mix_volumes[i],
                          mix_volumes[i]);
            }

            dsp_state.previous_samples[i] =
                ApplyMixRamp(GetMixBuffer(mix_buffer_offset + i), GetMixBuffer(voice_index),
                             last_mix_volumes[i], delta, worker_params.sample_count);
        } else {
            dsp_state.previous_samples[i] = 0;
        }
    }
}

void CommandGenerator::GenerateSubMixCommand(ServerMixInfo& mix_info) {
    if (dumping_frame) {
        LOG_DEBUG(Audio, "(DSP_TRACE) GenerateSubMixCommand");
    }
    // TODO(ogniK): Depop
    // TODO(ogniK): Effects
    GenerateMixCommands(mix_info);
}

void CommandGenerator::GenerateMixCommands(ServerMixInfo& mix_info) {
    if (!mix_info.HasAnyConnection()) {
        return;
    }
    const auto& in_params = mix_info.GetInParams();
    if (in_params.dest_mix_id != AudioCommon::NO_MIX) {
        const auto& dest_mix = mix_context.GetInfo(in_params.dest_mix_id);
        const auto& dest_in_params = dest_mix.GetInParams();

        const auto buffer_count = in_params.buffer_count;

        for (s32 i = 0; i < buffer_count; i++) {
            for (s32 j = 0; j < dest_in_params.buffer_count; j++) {
                const auto mixed_volume = in_params.volume * in_params.mix_volume[i][j];
                if (mixed_volume != 0.0f) {
                    GenerateMixCommand(dest_in_params.buffer_offset + j,
                                       in_params.buffer_offset + i, mixed_volume,
                                       in_params.node_id);
                }
            }
        }
    } else if (in_params.splitter_id != AudioCommon::NO_SPLITTER) {
        s32 base{};
        while (const auto* destination_data = GetDestinationData(in_params.splitter_id, base++)) {
            if (!destination_data->IsConfigured()) {
                continue;
            }

            const auto& dest_mix = mix_context.GetInfo(destination_data->GetMixId());
            const auto& dest_in_params = dest_mix.GetInParams();
            const auto mix_index = (base - 1) % in_params.buffer_count + in_params.buffer_offset;
            for (std::size_t i = 0; i < dest_in_params.buffer_count; i++) {
                const auto mixed_volume = in_params.volume * destination_data->GetMixVolume(i);
                if (mixed_volume != 0.0f) {
                    GenerateMixCommand(dest_in_params.buffer_offset + i, mix_index, mixed_volume,
                                       in_params.node_id);
                }
            }
        }
    }
}

void CommandGenerator::GenerateMixCommand(std::size_t output_offset, std::size_t input_offset,
                                          float volume, s32 node_id) {

    if (dumping_frame) {
        LOG_DEBUG(Audio,
                  "(DSP_TRACE) GenerateMixCommand node_id={}, input={}, output={}, volume={}",
                  node_id, input_offset, output_offset, volume);
    }

    auto* output = GetMixBuffer(output_offset);
    const auto* input = GetMixBuffer(input_offset);

    const s32 gain = static_cast<s32>(volume * 32768.0f);
    // Mix with loop unrolling
    if (worker_params.sample_count % 4 == 0) {
        ApplyMix<4>(output, input, gain, worker_params.sample_count);
    } else if (worker_params.sample_count % 2 == 0) {
        ApplyMix<2>(output, input, gain, worker_params.sample_count);
    } else {
        ApplyMix<1>(output, input, gain, worker_params.sample_count);
    }
}

void CommandGenerator::GenerateFinalMixCommand() {
    if (dumping_frame) {
        LOG_DEBUG(Audio, "(DSP_TRACE) GenerateFinalMixCommand");
    }
    // TODO(ogniK): Depop
    // TODO(ogniK): Effects
    auto& mix_info = mix_context.GetFinalMixInfo();
    const auto in_params = mix_info.GetInParams();
    for (s32 i = 0; i < in_params.buffer_count; i++) {
        const s32 gain = static_cast<s32>(in_params.volume * 32768.0f);
        if (dumping_frame) {
            LOG_DEBUG(
                Audio,
                "(DSP_TRACE) ApplyGainWithoutDelta node_id={}, input={}, output={}, volume={}",
                in_params.node_id, in_params.buffer_offset + i, in_params.buffer_offset + i,
                in_params.volume);
        }
        ApplyGainWithoutDelta(GetMixBuffer(in_params.buffer_offset + i),
                              GetMixBuffer(in_params.buffer_offset + i), gain,
                              worker_params.sample_count);
    }
}

s32 CommandGenerator::DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
                                  s32 sample_count, s32 channel, std::size_t mix_offset) {
    auto& in_params = voice_info.GetInParams();
    const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
    if (wave_buffer.buffer_address == 0) {
        return 0;
    }
    if (wave_buffer.buffer_size == 0) {
        return 0;
    }
    if (wave_buffer.end_sample_offset < wave_buffer.start_sample_offset) {
        return 0;
    }
    const auto samples_remaining =
        (wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
    const auto start_offset =
        ((wave_buffer.start_sample_offset + dsp_state.offset) * in_params.channel_count) *
        sizeof(s16);
    const auto buffer_pos = wave_buffer.buffer_address + start_offset;
    const auto samples_processed = std::min(sample_count, samples_remaining);

    if (in_params.channel_count == 1) {
        std::vector<s16> buffer(samples_processed);
        memory.ReadBlock(buffer_pos, buffer.data(), buffer.size() * sizeof(s16));
        for (std::size_t i = 0; i < buffer.size(); i++) {
            sample_buffer[mix_offset + i] = buffer[i];
        }
    } else {
        const auto channel_count = in_params.channel_count;
        std::vector<s16> buffer(samples_processed * channel_count);
        memory.ReadBlock(buffer_pos, buffer.data(), buffer.size() * sizeof(s16));

        for (std::size_t i = 0; i < samples_processed; i++) {
            sample_buffer[mix_offset + i] = buffer[i * channel_count + channel];
        }
    }

    return samples_processed;
}
s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
                                  s32 sample_count, s32 channel, std::size_t mix_offset) {
    auto& in_params = voice_info.GetInParams();
    const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
    if (wave_buffer.buffer_address == 0) {
        return 0;
    }
    if (wave_buffer.buffer_size == 0) {
        return 0;
    }
    if (wave_buffer.end_sample_offset < wave_buffer.start_sample_offset) {
        return 0;
    }

    const auto samples_remaining =
        (wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
    const auto start_offset =
        ((wave_buffer.start_sample_offset + dsp_state.offset) * in_params.channel_count);
    const auto buffer_pos = wave_buffer.buffer_address + start_offset;

    const auto samples_processed = std::min(sample_count, samples_remaining);

    if (start_offset > dsp_state.adpcm_samples.size()) {
        dsp_state.adpcm_samples.clear();
    }

    // TODO(ogniK): Proper ADPCM streaming
    if (dsp_state.adpcm_samples.empty()) {
        Codec::ADPCM_Coeff coeffs;
        memory.ReadBlock(in_params.additional_params_address, coeffs.data(),
                         sizeof(Codec::ADPCM_Coeff));
        std::vector<u8> buffer(wave_buffer.buffer_size);
        memory.ReadBlock(wave_buffer.buffer_address, buffer.data(), buffer.size());
        dsp_state.adpcm_samples =
            std::move(Codec::DecodeADPCM(buffer.data(), buffer.size(), coeffs, dsp_state.context));
    }

    for (std::size_t i = 0; i < samples_processed; i++) {
        const auto sample_offset = i + start_offset;
        sample_buffer[mix_offset + i] =
            dsp_state.adpcm_samples[sample_offset * in_params.channel_count + channel];
    }

    return samples_processed;
}

s32* CommandGenerator::GetMixBuffer(std::size_t index) {
    return mix_buffer.data() + (index * worker_params.sample_count);
}

const s32* CommandGenerator::GetMixBuffer(std::size_t index) const {
    return mix_buffer.data() + (index * worker_params.sample_count);
}

std::size_t CommandGenerator::GetMixChannelBufferOffset(s32 channel) const {
    return worker_params.mix_buffer_count + channel;
}

s32* CommandGenerator::GetChannelMixBuffer(s32 channel) {
    return GetMixBuffer(worker_params.mix_buffer_count + channel);
}

const s32* CommandGenerator::GetChannelMixBuffer(s32 channel) const {
    return GetMixBuffer(worker_params.mix_buffer_count + channel);
}

void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* output,
                                             VoiceState& dsp_state, s32 channel,
                                             s32 target_sample_rate, s32 sample_count,
                                             s32 node_id) {
    auto& in_params = voice_info.GetInParams();
    if (dumping_frame) {
        LOG_DEBUG(Audio,
                  "(DSP_TRACE) DecodeFromWaveBuffers, node_id={}, channel={}, "
                  "format={}, sample_count={}, sample_rate={}, mix_id={}, splitter_id={}",
                  node_id, channel, in_params.sample_format, sample_count, in_params.sample_rate,
                  in_params.mix_id, in_params.splitter_info_id);
    }
    ASSERT_OR_EXECUTE(output != nullptr, { return; });

    const auto resample_rate = static_cast<s32>(
        static_cast<float>(in_params.sample_rate) / static_cast<float>(target_sample_rate) *
        static_cast<float>(static_cast<s32>(in_params.pitch * 32768.0f)));
    auto* output_base = output;
    if ((dsp_state.fraction + sample_count * resample_rate) > (SCALED_MIX_BUFFER_SIZE - 4ULL)) {
        return;
    }

    auto min_required_samples =
        std::min(static_cast<s32>(SCALED_MIX_BUFFER_SIZE) - dsp_state.fraction, resample_rate);
    if (min_required_samples >= sample_count) {
        min_required_samples = sample_count;
    }

    std::size_t temp_mix_offset{};
    bool is_buffer_completed{false};
    auto samples_remaining = sample_count;
    while (samples_remaining > 0 && !is_buffer_completed) {
        const auto samples_to_output = std::min(samples_remaining, min_required_samples);
        const auto samples_to_read = (samples_to_output * resample_rate + dsp_state.fraction) >> 15;

        if (!in_params.behavior_flags.is_pitch_and_src_skipped) {
            // Append sample histtory for resampler
            for (std::size_t i = 0; i < AudioCommon::MAX_SAMPLE_HISTORY; i++) {
                sample_buffer[temp_mix_offset + i] = dsp_state.sample_history[i];
            }
            temp_mix_offset += 4;
        }

        s32 samples_read{};
        while (samples_read < samples_to_read) {
            const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
            // No more data can be read
            if (!dsp_state.is_wave_buffer_valid[dsp_state.wave_buffer_index]) {
                is_buffer_completed = true;
                break;
            }

            if (in_params.sample_format == SampleFormat::Adpcm && dsp_state.offset == 0 &&
                wave_buffer.context_address != 0 && wave_buffer.context_size != 0) {
                // TODO(ogniK): ADPCM loop context
            }

            s32 samples_decoded{0};
            switch (in_params.sample_format) {
            case SampleFormat::Pcm16:
                samples_decoded = DecodePcm16(voice_info, dsp_state, samples_to_read - samples_read,
                                              channel, temp_mix_offset);
                break;
            case SampleFormat::Adpcm:
                samples_decoded = DecodeAdpcm(voice_info, dsp_state, samples_to_read - samples_read,
                                              channel, temp_mix_offset);
                break;
            default:
                UNREACHABLE_MSG("Unimplemented sample format={}", in_params.sample_format);
            }

            temp_mix_offset += samples_decoded;
            samples_read += samples_decoded;
            dsp_state.offset += samples_decoded;
            dsp_state.played_sample_count += samples_decoded;

            if (dsp_state.offset >=
                    (wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) ||
                samples_decoded == 0) {
                // Reset our sample offset
                dsp_state.offset = 0;
                if (wave_buffer.is_looping) {
                    if (samples_decoded == 0) {
                        // End of our buffer
                        is_buffer_completed = true;
                        break;
                    }

                    if (in_params.behavior_flags.is_played_samples_reset_at_loop_point.Value()) {
                        dsp_state.played_sample_count = 0;
                    }
                } else {
                    if (in_params.sample_format == SampleFormat::Adpcm) {
                        // TODO(ogniK): Remove this when ADPCM streaming implemented
                        dsp_state.adpcm_samples.clear();
                    }

                    // Update our wave buffer states
                    dsp_state.is_wave_buffer_valid[dsp_state.wave_buffer_index] = false;
                    dsp_state.wave_buffer_consumed++;
                    dsp_state.wave_buffer_index =
                        (dsp_state.wave_buffer_index + 1) % AudioCommon::MAX_WAVE_BUFFERS;
                    if (wave_buffer.end_of_stream) {
                        dsp_state.played_sample_count = 0;
                    }
                }
            }
        }

        if (in_params.behavior_flags.is_pitch_and_src_skipped.Value()) {
            // No need to resample
            std::memcpy(output, sample_buffer.data(), samples_read * sizeof(s32));
        } else {
            std::fill(sample_buffer.begin() + temp_mix_offset,
                      sample_buffer.begin() + temp_mix_offset + (samples_to_read - samples_read),
                      0);
            AudioCore::Resample(output, sample_buffer.data(), resample_rate, dsp_state.fraction,
                                samples_to_output);
            // Resample
            for (std::size_t i = 0; i < AudioCommon::MAX_SAMPLE_HISTORY; i++) {
                dsp_state.sample_history[i] = sample_buffer[samples_to_read + i];
            }
        }
        output += samples_to_output;
        samples_remaining -= samples_to_output;
    }
}

} // namespace AudioCore