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Merge pull request #6498 from Kelebek1/Audio

[audio_core] Decouple audio update and processing, and process at variable rate
This commit is contained in:
bunnei 2021-07-03 00:24:33 -07:00 committed by GitHub
commit 2fc0a760f0
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GPG key ID: 4AEE18F83AFDEB23
8 changed files with 180 additions and 88 deletions

View file

@ -12,6 +12,7 @@
#include "audio_core/voice_context.h"
#include "common/logging/log.h"
#include "common/settings.h"
#include "core/core_timing.h"
#include "core/memory.h"
namespace {
@ -68,7 +69,9 @@ namespace {
} // namespace
namespace AudioCore {
AudioRenderer::AudioRenderer(Core::Timing::CoreTiming& core_timing, Core::Memory::Memory& memory_,
constexpr s32 NUM_BUFFERS = 2;
AudioRenderer::AudioRenderer(Core::Timing::CoreTiming& core_timing_, Core::Memory::Memory& memory_,
AudioCommon::AudioRendererParameter params,
Stream::ReleaseCallback&& release_callback,
std::size_t instance_number)
@ -77,7 +80,8 @@ AudioRenderer::AudioRenderer(Core::Timing::CoreTiming& core_timing, Core::Memory
sink_context(params.sink_count), splitter_context(),
voices(params.voice_count), memory{memory_},
command_generator(worker_params, voice_context, mix_context, splitter_context, effect_context,
memory) {
memory),
core_timing{core_timing_} {
behavior_info.SetUserRevision(params.revision);
splitter_context.Initialize(behavior_info, params.splitter_count,
params.num_splitter_send_channels);
@ -86,16 +90,27 @@ AudioRenderer::AudioRenderer(Core::Timing::CoreTiming& core_timing, Core::Memory
stream = audio_out->OpenStream(
core_timing, params.sample_rate, AudioCommon::STREAM_NUM_CHANNELS,
fmt::format("AudioRenderer-Instance{}", instance_number), std::move(release_callback));
audio_out->StartStream(stream);
QueueMixedBuffer(0);
QueueMixedBuffer(1);
QueueMixedBuffer(2);
QueueMixedBuffer(3);
process_event = Core::Timing::CreateEvent(
fmt::format("AudioRenderer-Instance{}-Process", instance_number),
[this](std::uintptr_t, std::chrono::nanoseconds) { ReleaseAndQueueBuffers(); });
for (s32 i = 0; i < NUM_BUFFERS; ++i) {
QueueMixedBuffer(i);
}
}
AudioRenderer::~AudioRenderer() = default;
ResultCode AudioRenderer::Start() {
audio_out->StartStream(stream);
ReleaseAndQueueBuffers();
return ResultSuccess;
}
ResultCode AudioRenderer::Stop() {
audio_out->StopStream(stream);
return ResultSuccess;
}
u32 AudioRenderer::GetSampleRate() const {
return worker_params.sample_rate;
}
@ -114,7 +129,7 @@ Stream::State AudioRenderer::GetStreamState() const {
ResultCode AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params,
std::vector<u8>& output_params) {
std::scoped_lock lock{mutex};
InfoUpdater info_updater{input_params, output_params, behavior_info};
if (!info_updater.UpdateBehaviorInfo(behavior_info)) {
@ -194,9 +209,6 @@ ResultCode AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_param
LOG_ERROR(Audio, "Audio buffers were not consumed!");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
ReleaseAndQueueBuffers();
return ResultSuccess;
}
@ -220,10 +232,8 @@ void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
command_generator.PostCommand();
// Base sample size
std::size_t BUFFER_SIZE{worker_params.sample_count};
// Samples
std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels());
// Make sure to clear our samples
std::memset(buffer.data(), 0, buffer.size() * sizeof(s16));
// Samples, making sure to clear
std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels(), 0);
if (sink_context.InUse()) {
const auto stream_channel_count = stream->GetNumChannels();
@ -315,10 +325,24 @@ void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
}
void AudioRenderer::ReleaseAndQueueBuffers() {
const auto released_buffers{audio_out->GetTagsAndReleaseBuffers(stream)};
for (const auto& tag : released_buffers) {
QueueMixedBuffer(tag);
if (!stream->IsPlaying()) {
return;
}
{
std::scoped_lock lock{mutex};
const auto released_buffers{audio_out->GetTagsAndReleaseBuffers(stream)};
for (const auto& tag : released_buffers) {
QueueMixedBuffer(tag);
}
}
const f32 sample_rate = static_cast<f32>(GetSampleRate());
const f32 sample_count = static_cast<f32>(GetSampleCount());
const f32 consume_rate = sample_rate / (sample_count * (sample_count / 240));
const s32 ms = (1000 / static_cast<s32>(consume_rate)) - 1;
const std::chrono::milliseconds next_event_time(std::max(ms / NUM_BUFFERS, 1));
core_timing.ScheduleEvent(next_event_time, process_event, {});
}
} // namespace AudioCore

View file

@ -6,6 +6,7 @@
#include <array>
#include <memory>
#include <mutex>
#include <vector>
#include "audio_core/behavior_info.h"
@ -45,6 +46,8 @@ public:
[[nodiscard]] ResultCode UpdateAudioRenderer(const std::vector<u8>& input_params,
std::vector<u8>& output_params);
[[nodiscard]] ResultCode Start();
[[nodiscard]] ResultCode Stop();
void QueueMixedBuffer(Buffer::Tag tag);
void ReleaseAndQueueBuffers();
[[nodiscard]] u32 GetSampleRate() const;
@ -68,6 +71,9 @@ private:
Core::Memory::Memory& memory;
CommandGenerator command_generator;
std::size_t elapsed_frame_count{};
Core::Timing::CoreTiming& core_timing;
std::shared_ptr<Core::Timing::EventType> process_event;
std::mutex mutex;
};
} // namespace AudioCore

View file

@ -795,7 +795,7 @@ void CommandGenerator::UpdateI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbSta
state.lowpass_1 = 0.0f;
} else {
const auto a = 1.0f - hf_gain;
const auto b = 2.0f * (1.0f - hf_gain * CosD(256.0f * info.hf_reference /
const auto b = 2.0f * (2.0f - hf_gain * CosD(256.0f * info.hf_reference /
static_cast<f32>(info.sample_rate)));
const auto c = std::sqrt(b * b - 4.0f * a * a);
@ -843,7 +843,7 @@ void CommandGenerator::UpdateI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbSta
}
const auto max_early_delay = state.early_delay_line.GetMaxDelay();
const auto reflection_time = 1000.0f * (0.0098f * info.reverb_delay + 0.02f);
const auto reflection_time = 1000.0f * (0.9998f * info.reverb_delay + 0.02f);
for (std::size_t tap = 0; tap < AudioCommon::I3DL2REVERB_TAPS; tap++) {
const auto length = AudioCommon::CalculateDelaySamples(
sample_rate, 1000.0f * info.reflection_delay + reflection_time * EARLY_TAP_TIMES[tap]);
@ -1004,7 +1004,8 @@ void CommandGenerator::GenerateFinalMixCommand() {
}
s32 CommandGenerator::DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
s32 sample_count, s32 channel, std::size_t mix_offset) {
s32 sample_start_offset, s32 sample_end_offset, s32 sample_count,
s32 channel, std::size_t mix_offset) {
const 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) {
@ -1013,14 +1014,12 @@ s32 CommandGenerator::DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_s
if (wave_buffer.buffer_size == 0) {
return 0;
}
if (wave_buffer.end_sample_offset < wave_buffer.start_sample_offset) {
if (sample_end_offset < sample_start_offset) {
return 0;
}
const auto samples_remaining =
(wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
const auto samples_remaining = (sample_end_offset - sample_start_offset) - dsp_state.offset;
const auto start_offset =
((wave_buffer.start_sample_offset + dsp_state.offset) * in_params.channel_count) *
sizeof(s16);
((dsp_state.offset + sample_start_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);
@ -1044,8 +1043,8 @@ s32 CommandGenerator::DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_s
}
s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
s32 sample_count, [[maybe_unused]] s32 channel,
std::size_t mix_offset) {
s32 sample_start_offset, s32 sample_end_offset, s32 sample_count,
[[maybe_unused]] s32 channel, std::size_t mix_offset) {
const 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) {
@ -1054,7 +1053,7 @@ s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_s
if (wave_buffer.buffer_size == 0) {
return 0;
}
if (wave_buffer.end_sample_offset < wave_buffer.start_sample_offset) {
if (sample_end_offset < sample_start_offset) {
return 0;
}
@ -1079,10 +1078,9 @@ s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_s
s32 coef1 = coeffs[idx * 2];
s32 coef2 = coeffs[idx * 2 + 1];
const auto samples_remaining =
(wave_buffer.end_sample_offset - wave_buffer.start_sample_offset) - dsp_state.offset;
const auto samples_remaining = (sample_end_offset - sample_start_offset) - dsp_state.offset;
const auto samples_processed = std::min(sample_count, samples_remaining);
const auto sample_pos = wave_buffer.start_sample_offset + dsp_state.offset;
const auto sample_pos = dsp_state.offset + sample_start_offset;
const auto samples_remaining_in_frame = sample_pos % SAMPLES_PER_FRAME;
auto position_in_frame = ((sample_pos / SAMPLES_PER_FRAME) * NIBBLES_PER_SAMPLE) +
@ -1210,9 +1208,8 @@ void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* o
}
std::size_t temp_mix_offset{};
bool is_buffer_completed{false};
auto samples_remaining = sample_count;
while (samples_remaining > 0 && !is_buffer_completed) {
while (samples_remaining > 0) {
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;
@ -1229,24 +1226,38 @@ void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* o
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
memory.ReadBlock(wave_buffer.context_address, &dsp_state.context,
sizeof(ADPCMContext));
}
s32 samples_offset_start;
s32 samples_offset_end;
if (dsp_state.loop_count > 0 && wave_buffer.loop_start_sample != 0 &&
wave_buffer.loop_end_sample != 0 &&
wave_buffer.loop_start_sample <= wave_buffer.loop_end_sample) {
samples_offset_start = wave_buffer.loop_start_sample;
samples_offset_end = wave_buffer.loop_end_sample;
} else {
samples_offset_start = wave_buffer.start_sample_offset;
samples_offset_end = wave_buffer.end_sample_offset;
}
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);
samples_decoded =
DecodePcm16(voice_info, dsp_state, samples_offset_start, samples_offset_end,
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);
samples_decoded =
DecodeAdpcm(voice_info, dsp_state, samples_offset_start, samples_offset_end,
samples_to_read - samples_read, channel, temp_mix_offset);
break;
default:
UNREACHABLE_MSG("Unimplemented sample format={}", in_params.sample_format);
@ -1257,15 +1268,19 @@ void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* o
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) ||
if (dsp_state.offset >= (samples_offset_end - samples_offset_start) ||
samples_decoded == 0) {
// Reset our sample offset
dsp_state.offset = 0;
if (wave_buffer.is_looping) {
if (samples_decoded == 0) {
dsp_state.loop_count++;
if (wave_buffer.loop_count > 0 &&
(dsp_state.loop_count > wave_buffer.loop_count || samples_decoded == 0)) {
// End of our buffer
is_buffer_completed = true;
voice_info.SetWaveBufferCompleted(dsp_state, wave_buffer);
}
if (samples_decoded == 0) {
break;
}
@ -1273,15 +1288,8 @@ void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* o
dsp_state.played_sample_count = 0;
}
} else {
// 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;
}
voice_info.SetWaveBufferCompleted(dsp_state, wave_buffer);
}
}
}

View file

@ -86,10 +86,10 @@ private:
std::vector<u8>& work_buffer);
void UpdateI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbState& state, bool should_clear);
// DSP Code
s32 DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_count,
s32 channel, std::size_t mix_offset);
s32 DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_count,
s32 channel, std::size_t mix_offset);
s32 DecodePcm16(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_start_offset,
s32 sample_end_offset, s32 sample_count, s32 channel, std::size_t mix_offset);
s32 DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_start_offset,
s32 sample_end_offset, s32 sample_count, s32 channel, std::size_t mix_offset);
void DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, s32* output, VoiceState& dsp_state,
s32 channel, s32 target_sample_rate, s32 sample_count, s32 node_id);

View file

@ -189,9 +189,6 @@ bool InfoUpdater::UpdateVoices(VoiceContext& voice_context,
if (voice_in_params.is_new) {
// Default our values for our voice
voice_info.Initialize();
if (channel_count == 0 || channel_count > AudioCommon::MAX_CHANNEL_COUNT) {
continue;
}
// Zero out our voice states
for (std::size_t channel = 0; channel < channel_count; channel++) {

View file

@ -66,7 +66,7 @@ void ServerVoiceInfo::Initialize() {
in_params.last_volume = 0.0f;
in_params.biquad_filter.fill({});
in_params.wave_buffer_count = 0;
in_params.wave_bufffer_head = 0;
in_params.wave_buffer_head = 0;
in_params.mix_id = AudioCommon::NO_MIX;
in_params.splitter_info_id = AudioCommon::NO_SPLITTER;
in_params.additional_params_address = 0;
@ -75,7 +75,7 @@ void ServerVoiceInfo::Initialize() {
out_params.played_sample_count = 0;
out_params.wave_buffer_consumed = 0;
in_params.voice_drop_flag = false;
in_params.buffer_mapped = false;
in_params.buffer_mapped = true;
in_params.wave_buffer_flush_request_count = 0;
in_params.was_biquad_filter_enabled.fill(false);
@ -126,7 +126,7 @@ void ServerVoiceInfo::UpdateParameters(const VoiceInfo::InParams& voice_in,
in_params.volume = voice_in.volume;
in_params.biquad_filter = voice_in.biquad_filter;
in_params.wave_buffer_count = voice_in.wave_buffer_count;
in_params.wave_bufffer_head = voice_in.wave_buffer_head;
in_params.wave_buffer_head = voice_in.wave_buffer_head;
if (behavior_info.IsFlushVoiceWaveBuffersSupported()) {
const auto in_request_count = in_params.wave_buffer_flush_request_count;
const auto voice_request_count = voice_in.wave_buffer_flush_request_count;
@ -185,14 +185,16 @@ void ServerVoiceInfo::UpdateWaveBuffers(
wave_buffer.buffer_size = 0;
wave_buffer.context_address = 0;
wave_buffer.context_size = 0;
wave_buffer.loop_start_sample = 0;
wave_buffer.loop_end_sample = 0;
wave_buffer.sent_to_dsp = true;
}
// Mark all our wave buffers as invalid
for (std::size_t channel = 0; channel < static_cast<std::size_t>(in_params.channel_count);
channel++) {
for (auto& is_valid : voice_states[channel]->is_wave_buffer_valid) {
is_valid = false;
for (std::size_t i = 0; i < AudioCommon::MAX_WAVE_BUFFERS; ++i) {
voice_states[channel]->is_wave_buffer_valid[i] = false;
}
}
}
@ -211,7 +213,7 @@ void ServerVoiceInfo::UpdateWaveBuffer(ServerWaveBuffer& out_wavebuffer,
const WaveBuffer& in_wave_buffer, SampleFormat sample_format,
bool is_buffer_valid,
[[maybe_unused]] BehaviorInfo& behavior_info) {
if (!is_buffer_valid && out_wavebuffer.sent_to_dsp) {
if (!is_buffer_valid && out_wavebuffer.sent_to_dsp && out_wavebuffer.buffer_address != 0) {
out_wavebuffer.buffer_address = 0;
out_wavebuffer.buffer_size = 0;
}
@ -219,11 +221,40 @@ void ServerVoiceInfo::UpdateWaveBuffer(ServerWaveBuffer& out_wavebuffer,
if (!in_wave_buffer.sent_to_server || !in_params.buffer_mapped) {
// Validate sample offset sizings
if (sample_format == SampleFormat::Pcm16) {
const auto buffer_size = in_wave_buffer.buffer_size;
if (in_wave_buffer.start_sample_offset < 0 || in_wave_buffer.end_sample_offset < 0 ||
(buffer_size < (sizeof(s16) * in_wave_buffer.start_sample_offset)) ||
(buffer_size < (sizeof(s16) * in_wave_buffer.end_sample_offset))) {
const s64 buffer_size = static_cast<s64>(in_wave_buffer.buffer_size);
const s64 start = sizeof(s16) * in_wave_buffer.start_sample_offset;
const s64 end = sizeof(s16) * in_wave_buffer.end_sample_offset;
if (0 > start || start > buffer_size || 0 > end || end > buffer_size) {
// TODO(ogniK): Write error info
LOG_ERROR(Audio,
"PCM16 wavebuffer has an invalid size. Buffer has size 0x{:08X}, but "
"offsets were "
"{:08X} - 0x{:08X}",
buffer_size, sizeof(s16) * in_wave_buffer.start_sample_offset,
sizeof(s16) * in_wave_buffer.end_sample_offset);
return;
}
} else if (sample_format == SampleFormat::Adpcm) {
const s64 buffer_size = static_cast<s64>(in_wave_buffer.buffer_size);
const s64 start_frames = in_wave_buffer.start_sample_offset / 14;
const s64 start_extra = in_wave_buffer.start_sample_offset % 14 == 0
? 0
: (in_wave_buffer.start_sample_offset % 14) / 2 + 1 +
(in_wave_buffer.start_sample_offset % 2);
const s64 start = start_frames * 8 + start_extra;
const s64 end_frames = in_wave_buffer.end_sample_offset / 14;
const s64 end_extra = in_wave_buffer.end_sample_offset % 14 == 0
? 0
: (in_wave_buffer.end_sample_offset % 14) / 2 + 1 +
(in_wave_buffer.end_sample_offset % 2);
const s64 end = end_frames * 8 + end_extra;
if (in_wave_buffer.start_sample_offset < 0 || start > buffer_size ||
in_wave_buffer.end_sample_offset < 0 || end > buffer_size) {
LOG_ERROR(Audio,
"ADPMC wavebuffer has an invalid size. Buffer has size 0x{:08X}, but "
"offsets were "
"{:08X} - 0x{:08X}",
in_wave_buffer.buffer_size, start, end);
return;
}
}
@ -239,29 +270,34 @@ void ServerVoiceInfo::UpdateWaveBuffer(ServerWaveBuffer& out_wavebuffer,
out_wavebuffer.buffer_size = in_wave_buffer.buffer_size;
out_wavebuffer.context_address = in_wave_buffer.context_address;
out_wavebuffer.context_size = in_wave_buffer.context_size;
out_wavebuffer.loop_start_sample = in_wave_buffer.loop_start_sample;
out_wavebuffer.loop_end_sample = in_wave_buffer.loop_end_sample;
in_params.buffer_mapped =
in_wave_buffer.buffer_address != 0 && in_wave_buffer.buffer_size != 0;
// TODO(ogniK): Pool mapper attachment
// TODO(ogniK): IsAdpcmLoopContextBugFixed
if (sample_format == SampleFormat::Adpcm && in_wave_buffer.context_address != 0 &&
in_wave_buffer.context_size != 0 && behavior_info.IsAdpcmLoopContextBugFixed()) {
} else {
out_wavebuffer.context_address = 0;
out_wavebuffer.context_size = 0;
}
}
}
void ServerVoiceInfo::WriteOutStatus(
VoiceInfo::OutParams& voice_out, VoiceInfo::InParams& voice_in,
std::array<VoiceState*, AudioCommon::MAX_CHANNEL_COUNT>& voice_states) {
if (voice_in.is_new) {
if (voice_in.is_new || in_params.is_new) {
in_params.is_new = true;
voice_out.wave_buffer_consumed = 0;
voice_out.played_sample_count = 0;
voice_out.voice_dropped = false;
} else if (!in_params.is_new) {
voice_out.wave_buffer_consumed = voice_states[0]->wave_buffer_consumed;
voice_out.played_sample_count = voice_states[0]->played_sample_count;
voice_out.voice_dropped = in_params.voice_drop_flag;
} else {
voice_out.wave_buffer_consumed = 0;
voice_out.played_sample_count = 0;
voice_out.voice_dropped = false;
const auto& state = voice_states[0];
voice_out.wave_buffer_consumed = state->wave_buffer_consumed;
voice_out.played_sample_count = state->played_sample_count;
voice_out.voice_dropped = state->voice_dropped;
}
}
@ -283,7 +319,8 @@ ServerVoiceInfo::OutParams& ServerVoiceInfo::GetOutParams() {
bool ServerVoiceInfo::ShouldSkip() const {
// TODO(ogniK): Handle unmapped wave buffers or parameters
return !in_params.in_use || (in_params.wave_buffer_count == 0) || in_params.voice_drop_flag;
return !in_params.in_use || in_params.wave_buffer_count == 0 || !in_params.buffer_mapped ||
in_params.voice_drop_flag;
}
bool ServerVoiceInfo::UpdateForCommandGeneration(VoiceContext& voice_context) {
@ -381,7 +418,7 @@ bool ServerVoiceInfo::UpdateParametersForCommandGeneration(
void ServerVoiceInfo::FlushWaveBuffers(
u8 flush_count, std::array<VoiceState*, AudioCommon::MAX_CHANNEL_COUNT>& dsp_voice_states,
s32 channel_count) {
auto wave_head = in_params.wave_bufffer_head;
auto wave_head = in_params.wave_buffer_head;
for (u8 i = 0; i < flush_count; i++) {
in_params.wave_buffer[wave_head].sent_to_dsp = true;
@ -401,6 +438,17 @@ bool ServerVoiceInfo::HasValidWaveBuffer(const VoiceState* state) const {
return std::find(valid_wb.begin(), valid_wb.end(), true) != valid_wb.end();
}
void ServerVoiceInfo::SetWaveBufferCompleted(VoiceState& dsp_state,
const ServerWaveBuffer& wave_buffer) {
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;
dsp_state.loop_count = 0;
if (wave_buffer.end_of_stream) {
dsp_state.played_sample_count = 0;
}
}
VoiceContext::VoiceContext(std::size_t voice_count_) : voice_count{voice_count_} {
for (std::size_t i = 0; i < voice_count; i++) {
voice_channel_resources.emplace_back(static_cast<s32>(i));

View file

@ -60,10 +60,12 @@ struct WaveBuffer {
u8 is_looping{};
u8 end_of_stream{};
u8 sent_to_server{};
INSERT_PADDING_BYTES(5);
INSERT_PADDING_BYTES(1);
s32 loop_count{};
u64 context_address{};
u64 context_size{};
INSERT_PADDING_BYTES(8);
u32 loop_start_sample{};
u32 loop_end_sample{};
};
static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer is an invalid size");
@ -76,6 +78,9 @@ struct ServerWaveBuffer {
bool end_of_stream{};
VAddr context_address{};
std::size_t context_size{};
s32 loop_count{};
u32 loop_start_sample{};
u32 loop_end_sample{};
bool sent_to_dsp{true};
};
@ -108,6 +113,7 @@ struct VoiceState {
u32 external_context_size;
bool is_external_context_used;
bool voice_dropped;
s32 loop_count;
};
class VoiceChannelResource {
@ -206,7 +212,7 @@ public:
float last_volume{};
std::array<BiquadFilterParameter, AudioCommon::MAX_BIQUAD_FILTERS> biquad_filter{};
s32 wave_buffer_count{};
s16 wave_bufffer_head{};
s16 wave_buffer_head{};
INSERT_PADDING_BYTES(2);
BehaviorFlags behavior_flags{};
VAddr additional_params_address{};
@ -252,6 +258,7 @@ public:
void FlushWaveBuffers(u8 flush_count,
std::array<VoiceState*, AudioCommon::MAX_CHANNEL_COUNT>& dsp_voice_states,
s32 channel_count);
void SetWaveBufferCompleted(VoiceState& dsp_state, const ServerWaveBuffer& wave_buffer);
private:
std::vector<s16> stored_samples;

View file

@ -96,7 +96,7 @@ private:
void RequestUpdateImpl(Kernel::HLERequestContext& ctx) {
LOG_DEBUG(Service_Audio, "(STUBBED) called");
std::vector<u8> output_params(ctx.GetWriteBufferSize());
std::vector<u8> output_params(ctx.GetWriteBufferSize(), 0);
auto result = renderer->UpdateAudioRenderer(ctx.ReadBuffer(), output_params);
if (result.IsSuccess()) {
@ -110,17 +110,19 @@ private:
void Start(Kernel::HLERequestContext& ctx) {
LOG_WARNING(Service_Audio, "(STUBBED) called");
IPC::ResponseBuilder rb{ctx, 2};
const auto result = renderer->Start();
rb.Push(ResultSuccess);
IPC::ResponseBuilder rb{ctx, 2};
rb.Push(result);
}
void Stop(Kernel::HLERequestContext& ctx) {
LOG_WARNING(Service_Audio, "(STUBBED) called");
IPC::ResponseBuilder rb{ctx, 2};
const auto result = renderer->Stop();
rb.Push(ResultSuccess);
IPC::ResponseBuilder rb{ctx, 2};
rb.Push(result);
}
void QuerySystemEvent(Kernel::HLERequestContext& ctx) {