From 380658c21d39cf05ac765a9284da246388cca2a4 Mon Sep 17 00:00:00 2001 From: David Marcec Date: Sun, 12 Jul 2020 21:59:14 +1000 Subject: audio_core: Apollo Part 1, AudioRenderer refactor --- src/audio_core/command_generator.cpp | 668 +++++++++++++++++++++++++++++++++++ 1 file changed, 668 insertions(+) create mode 100644 src/audio_core/command_generator.cpp (limited to 'src/audio_core/command_generator.cpp') diff --git a/src/audio_core/command_generator.cpp b/src/audio_core/command_generator.cpp new file mode 100644 index 000000000..722f9b6c5 --- /dev/null +++ b/src/audio_core/command_generator.cpp @@ -0,0 +1,668 @@ +// 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 { +static constexpr std::size_t MIX_BUFFER_SIZE = 0x3f00; +static constexpr std::size_t SCALED_MIX_BUFFER_SIZE = MIX_BUFFER_SIZE << 15ULL; + +template +void ApplyMix(s32* output, const s32* input, s32 gain, s32 sample_count) { + for (s32 i = 0; i < sample_count; i += N) { + for (std::size_t j = 0; j < N; j++) { + output[i + j] += + static_cast((static_cast(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(static_cast(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((static_cast(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((static_cast(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::memset(mix_buffer.data(), 0, mix_buffer.size() * sizeof(s32)); + std::memset(sample_buffer.data(), 0, sample_buffer.size() * sizeof(s32)); +} + +void CommandGenerator::GenerateVoiceCommands() { + if (dumping_frame) { + LOG_CRITICAL(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) { + 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_CRITICAL(Audio, "(DSP_TRACE) SplitterGraph splitter_info={}, {}", i, graph); + } + } +} + +void CommandGenerator::PostCommand() { + if (dumping_frame) { + 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) { + 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) { + 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]) { + std::memset(dsp_state.biquad_filter_state.data(), 0, + dsp_state.biquad_filter_state.size() * sizeof(s64)); + } + + // 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& state, + std::size_t input_offset, std::size_t output_offset, s32 sample_count, s32 node_id) { + if (dumping_frame) { + LOG_CRITICAL(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 = params.numerator[0]; + const auto n1 = params.numerator[1]; + const auto n2 = params.numerator[2]; + const auto d0 = params.denominator[0]; + const auto d1 = params.denominator[1]; + + // Biquad filter states + auto s0 = state[0]; + auto s1 = state[1]; + + constexpr s64 MIN = std::numeric_limits::min(); + constexpr s64 MAX = std::numeric_limits::max(); + + for (int i = 0; i < sample_count; ++i) { + const auto sample = static_cast(input[i]); + const auto f = (sample * n0 + s0 + 0x4000) >> 15; + const auto y = std::clamp(f, MIN, MAX); + s0 = sample * n1 + y * d0 + s1; + s1 = sample * n2 + y * d1; + output[i] = static_cast(y); + } + + state[0] = s0; + state[1] = 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(last_volume * 32768.0f); + const auto current = static_cast(current_volume * 32768.0f); + const auto delta = static_cast((static_cast(current) - static_cast(last)) / + static_cast(worker_params.sample_count)); + + if (dumping_frame) { + LOG_CRITICAL(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((mix_volumes[i] - last_mix_volumes[i])) / + static_cast(worker_params.sample_count); + + if (dumping_frame) { + LOG_CRITICAL(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_CRITICAL(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_CRITICAL(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(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_CRITICAL(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(in_params.volume * 32768.0f); + if (dumping_frame) { + LOG_CRITICAL( + 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 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 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 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_CRITICAL(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( + static_cast(in_params.sample_rate) / static_cast(target_sample_rate) * + static_cast(static_cast(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(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 + memcpy(output, sample_buffer.data(), samples_read * sizeof(s32)); + } else { + std::memset(sample_buffer.data() + temp_mix_offset, 0, + sizeof(s32) * (samples_to_read - samples_read)); + 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 -- cgit v1.2.3