diff options
Diffstat (limited to 'src/audio_core')
-rw-r--r-- | src/audio_core/CMakeLists.txt | 15 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.cpp | 75 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.h | 8 | ||||
-rw-r--r-- | src/audio_core/hle/mixers.cpp | 201 | ||||
-rw-r--r-- | src/audio_core/hle/mixers.h | 63 | ||||
-rw-r--r-- | src/audio_core/hle/pipe.cpp | 9 | ||||
-rw-r--r-- | src/audio_core/hle/pipe.h | 12 | ||||
-rw-r--r-- | src/audio_core/hle/source.cpp | 4 | ||||
-rw-r--r-- | src/audio_core/sdl2_sink.cpp | 126 | ||||
-rw-r--r-- | src/audio_core/sdl2_sink.h | 30 | ||||
-rw-r--r-- | src/audio_core/sink.h | 2 | ||||
-rw-r--r-- | src/audio_core/sink_details.cpp | 7 | ||||
-rw-r--r-- | src/audio_core/time_stretch.cpp | 144 | ||||
-rw-r--r-- | src/audio_core/time_stretch.h | 57 |
14 files changed, 730 insertions, 23 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index 4cd7aba67..a72a907ef 100644 --- a/src/audio_core/CMakeLists.txt +++ b/src/audio_core/CMakeLists.txt @@ -3,10 +3,12 @@ set(SRCS codec.cpp hle/dsp.cpp hle/filter.cpp + hle/mixers.cpp hle/pipe.cpp hle/source.cpp interpolate.cpp sink_details.cpp + time_stretch.cpp ) set(HEADERS @@ -15,17 +17,30 @@ set(HEADERS hle/common.h hle/dsp.h hle/filter.h + hle/mixers.h hle/pipe.h hle/source.h interpolate.h null_sink.h sink.h sink_details.h + time_stretch.h ) include_directories(../../externals/soundtouch/include) +if(SDL2_FOUND) + set(SRCS ${SRCS} sdl2_sink.cpp) + set(HEADERS ${HEADERS} sdl2_sink.h) + include_directories(${SDL2_INCLUDE_DIR}) +endif() + create_directory_groups(${SRCS} ${HEADERS}) add_library(audio_core STATIC ${SRCS} ${HEADERS}) target_link_libraries(audio_core SoundTouch) + +if(SDL2_FOUND) + target_link_libraries(audio_core ${SDL2_LIBRARY}) + set_property(TARGET audio_core APPEND PROPERTY COMPILE_DEFINITIONS HAVE_SDL2) +endif() diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp index 0cdbdb06a..0640e1eff 100644 --- a/src/audio_core/hle/dsp.cpp +++ b/src/audio_core/hle/dsp.cpp @@ -6,13 +6,17 @@ #include <memory> #include "audio_core/hle/dsp.h" +#include "audio_core/hle/mixers.h" #include "audio_core/hle/pipe.h" #include "audio_core/hle/source.h" #include "audio_core/sink.h" +#include "audio_core/time_stretch.h" namespace DSP { namespace HLE { +// Region management + std::array<SharedMemory, 2> g_regions; static size_t CurrentRegionIndex() { @@ -40,43 +44,96 @@ static SharedMemory& WriteRegion() { return g_regions[1 - CurrentRegionIndex()]; } +// Audio processing and mixing + static std::array<Source, num_sources> sources = { Source(0), Source(1), Source(2), Source(3), Source(4), Source(5), Source(6), Source(7), Source(8), Source(9), Source(10), Source(11), Source(12), Source(13), Source(14), Source(15), Source(16), Source(17), Source(18), Source(19), Source(20), Source(21), Source(22), Source(23) }; +static Mixers mixers; + +static StereoFrame16 GenerateCurrentFrame() { + SharedMemory& read = ReadRegion(); + SharedMemory& write = WriteRegion(); + + std::array<QuadFrame32, 3> intermediate_mixes = {}; + + // Generate intermediate mixes + for (size_t i = 0; i < num_sources; i++) { + write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]); + for (size_t mix = 0; mix < 3; mix++) { + sources[i].MixInto(intermediate_mixes[mix], mix); + } + } + + // Generate final mix + write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples, write.intermediate_mix_samples, intermediate_mixes); + + StereoFrame16 output_frame = mixers.GetOutput(); + + // Write current output frame to the shared memory region + for (size_t samplei = 0; samplei < output_frame.size(); samplei++) { + for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) { + write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]); + } + } + + return output_frame; +} + +// Audio output static std::unique_ptr<AudioCore::Sink> sink; +static AudioCore::TimeStretcher time_stretcher; + +static void OutputCurrentFrame(const StereoFrame16& frame) { + time_stretcher.AddSamples(&frame[0][0], frame.size()); + sink->EnqueueSamples(time_stretcher.Process(sink->SamplesInQueue())); +} + +// Public Interface void Init() { DSP::HLE::ResetPipes(); + for (auto& source : sources) { source.Reset(); } + + mixers.Reset(); + + time_stretcher.Reset(); + if (sink) { + time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate()); + } } void Shutdown() { + time_stretcher.Flush(); + while (true) { + std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue()); + if (residual_audio.empty()) + break; + sink->EnqueueSamples(residual_audio); + } } bool Tick() { - SharedMemory& read = ReadRegion(); - SharedMemory& write = WriteRegion(); + StereoFrame16 current_frame = {}; - std::array<QuadFrame32, 3> intermediate_mixes = {}; + // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region) + current_frame = GenerateCurrentFrame(); - for (size_t i = 0; i < num_sources; i++) { - write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]); - for (size_t mix = 0; mix < 3; mix++) { - sources[i].MixInto(intermediate_mixes[mix], mix); - } - } + OutputCurrentFrame(current_frame); return true; } void SetSink(std::unique_ptr<AudioCore::Sink> sink_) { sink = std::move(sink_); + time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate()); } } // namespace HLE diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h index 4459a5668..9275cd7de 100644 --- a/src/audio_core/hle/dsp.h +++ b/src/audio_core/hle/dsp.h @@ -33,13 +33,9 @@ namespace HLE { // double-buffer. The frame counter is located as the very last u16 of each region and is incremented // each audio tick. -struct SharedMemory; - constexpr VAddr region0_base = 0x1FF50000; constexpr VAddr region1_base = 0x1FF70000; -extern std::array<SharedMemory, 2> g_regions; - /** * The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from * its memory regions, the higher and lower 16-bit halves are swapped compared to the little-endian @@ -432,7 +428,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32); /// Final mixed output in PCM16 stereo format, what you hear out of the speakers. /// When the application writes to this region it has no effect. struct FinalMixSamples { - s16_le pcm16[2 * samples_per_frame]; + s16_le pcm16[samples_per_frame][2]; }; ASSERT_DSP_STRUCT(FinalMixSamples, 640); @@ -507,6 +503,8 @@ struct SharedMemory { }; ASSERT_DSP_STRUCT(SharedMemory, 0x8000); +extern std::array<SharedMemory, 2> g_regions; + // Structures must have an offset that is a multiple of two. static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp new file mode 100644 index 000000000..18335f7f0 --- /dev/null +++ b/src/audio_core/hle/mixers.cpp @@ -0,0 +1,201 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <cstddef> + +#include "audio_core/hle/common.h" +#include "audio_core/hle/dsp.h" +#include "audio_core/hle/mixers.h" + +#include "common/assert.h" +#include "common/logging/log.h" +#include "common/math_util.h" + +namespace DSP { +namespace HLE { + +void Mixers::Reset() { + current_frame.fill({}); + state = {}; +} + +DspStatus Mixers::Tick(DspConfiguration& config, + const IntermediateMixSamples& read_samples, + IntermediateMixSamples& write_samples, + const std::array<QuadFrame32, 3>& input) +{ + ParseConfig(config); + + AuxReturn(read_samples); + AuxSend(write_samples, input); + + MixCurrentFrame(); + + return GetCurrentStatus(); +} + +void Mixers::ParseConfig(DspConfiguration& config) { + if (!config.dirty_raw) { + return; + } + + if (config.mixer1_enabled_dirty) { + config.mixer1_enabled_dirty.Assign(0); + state.mixer1_enabled = config.mixer1_enabled != 0; + LOG_TRACE(Audio_DSP, "mixers mixer1_enabled = %hu", config.mixer1_enabled); + } + + if (config.mixer2_enabled_dirty) { + config.mixer2_enabled_dirty.Assign(0); + state.mixer2_enabled = config.mixer2_enabled != 0; + LOG_TRACE(Audio_DSP, "mixers mixer2_enabled = %hu", config.mixer2_enabled); + } + + if (config.volume_0_dirty) { + config.volume_0_dirty.Assign(0); + state.intermediate_mixer_volume[0] = config.volume[0]; + LOG_TRACE(Audio_DSP, "mixers volume[0] = %f", config.volume[0]); + } + + if (config.volume_1_dirty) { + config.volume_1_dirty.Assign(0); + state.intermediate_mixer_volume[1] = config.volume[1]; + LOG_TRACE(Audio_DSP, "mixers volume[1] = %f", config.volume[1]); + } + + if (config.volume_2_dirty) { + config.volume_2_dirty.Assign(0); + state.intermediate_mixer_volume[2] = config.volume[2]; + LOG_TRACE(Audio_DSP, "mixers volume[2] = %f", config.volume[2]); + } + + if (config.output_format_dirty) { + config.output_format_dirty.Assign(0); + state.output_format = config.output_format; + LOG_TRACE(Audio_DSP, "mixers output_format = %zu", static_cast<size_t>(config.output_format)); + } + + if (config.headphones_connected_dirty) { + config.headphones_connected_dirty.Assign(0); + // Do nothing. + // (Note: Whether headphones are connected does affect coefficients used for surround sound.) + LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected); + } + + if (config.dirty_raw) { + LOG_DEBUG(Audio_DSP, "mixers remaining_dirty=%x", config.dirty_raw); + } + + config.dirty_raw = 0; +} + +static s16 ClampToS16(s32 value) { + return static_cast<s16>(MathUtil::Clamp(value, -32768, 32767)); +} + +static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a, const std::array<s16, 2>& b) { + return { + ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])), + ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1])) + }; +} + +void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) { + // TODO(merry): Limiter. (Currently we're performing final mixing assuming a disabled limiter.) + + switch (state.output_format) { + case OutputFormat::Mono: + std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(), + [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> { + // Downmix to mono + s16 mono = ClampToS16(static_cast<s32>((gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) / 2)); + // Mix into current frame + return AddAndClampToS16(accumulator, { mono, mono }); + }); + return; + + case OutputFormat::Surround: + // TODO(merry): Implement surround sound. + // fallthrough + + case OutputFormat::Stereo: + std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(), + [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> { + // Downmix to stereo + s16 left = ClampToS16(static_cast<s32>(gain * sample[0] + gain * sample[2])); + s16 right = ClampToS16(static_cast<s32>(gain * sample[1] + gain * sample[3])); + // Mix into current frame + return AddAndClampToS16(accumulator, { left, right }); + }); + return; + } + + UNREACHABLE_MSG("Invalid output_format %zu", static_cast<size_t>(state.output_format)); +} + +void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) { + // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32. + + if (state.mixer1_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + state.intermediate_mix_buffer[1][sample][channel] = read_samples.mix1.pcm32[channel][sample]; + } + } + } + + if (state.mixer2_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + state.intermediate_mix_buffer[2][sample][channel] = read_samples.mix2.pcm32[channel][sample]; + } + } + } +} + +void Mixers::AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input) { + // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32. + + state.intermediate_mix_buffer[0] = input[0]; + + if (state.mixer1_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + write_samples.mix1.pcm32[channel][sample] = input[1][sample][channel]; + } + } + } else { + state.intermediate_mix_buffer[1] = input[1]; + } + + if (state.mixer2_enabled) { + for (size_t sample = 0; sample < samples_per_frame; sample++) { + for (size_t channel = 0; channel < 4; channel++) { + write_samples.mix2.pcm32[channel][sample] = input[2][sample][channel]; + } + } + } else { + state.intermediate_mix_buffer[2] = input[2]; + } +} + +void Mixers::MixCurrentFrame() { + current_frame.fill({}); + + for (size_t mix = 0; mix < 3; mix++) { + DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix], state.intermediate_mix_buffer[mix]); + } + + // TODO(merry): Compressor. (We currently assume a disabled compressor.) +} + +DspStatus Mixers::GetCurrentStatus() const { + DspStatus status; + status.unknown = 0; + status.dropped_frames = 0; + return status; +} + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/mixers.h b/src/audio_core/hle/mixers.h new file mode 100644 index 000000000..b52952eb5 --- /dev/null +++ b/src/audio_core/hle/mixers.h @@ -0,0 +1,63 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <array> + +#include "audio_core/hle/common.h" +#include "audio_core/hle/dsp.h" + +namespace DSP { +namespace HLE { + +class Mixers final { +public: + Mixers() { + Reset(); + } + + void Reset(); + + DspStatus Tick(DspConfiguration& config, + const IntermediateMixSamples& read_samples, + IntermediateMixSamples& write_samples, + const std::array<QuadFrame32, 3>& input); + + StereoFrame16 GetOutput() const { + return current_frame; + } + +private: + StereoFrame16 current_frame = {}; + + using OutputFormat = DspConfiguration::OutputFormat; + + struct { + std::array<float, 3> intermediate_mixer_volume = {}; + + bool mixer1_enabled = false; + bool mixer2_enabled = false; + std::array<QuadFrame32, 3> intermediate_mix_buffer = {}; + + OutputFormat output_format = OutputFormat::Stereo; + + } state; + + /// INTERNAL: Update our internal state based on the current config. + void ParseConfig(DspConfiguration& config); + /// INTERNAL: Read samples from shared memory that have been modified by the ARM11. + void AuxReturn(const IntermediateMixSamples& read_samples); + /// INTERNAL: Write samples to shared memory for the ARM11 to modify. + void AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input); + /// INTERNAL: Mix current_frame. + void MixCurrentFrame(); + /// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate into current_frame. + void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples); + /// INTERNAL: Generate DspStatus based on internal state. + DspStatus GetCurrentStatus() const; +}; + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp index 03280780f..44dff1345 100644 --- a/src/audio_core/hle/pipe.cpp +++ b/src/audio_core/hle/pipe.cpp @@ -36,12 +36,17 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) { return {}; } + if (length > UINT16_MAX) { // Can only read at most UINT16_MAX from the pipe + LOG_ERROR(Audio_DSP, "length of %u greater than max of %u", length, UINT16_MAX); + return {}; + } + std::vector<u8>& data = pipe_data[pipe_index]; if (length > data.size()) { LOG_WARNING(Audio_DSP, "pipe_number = %zu is out of data, application requested read of %u but %zu remain", pipe_index, length, data.size()); - length = data.size(); + length = static_cast<u32>(data.size()); } if (length == 0) @@ -94,7 +99,7 @@ static void AudioPipeWriteStructAddresses() { }; // Begin with a u16 denoting the number of structs. - WriteU16(DspPipe::Audio, struct_addresses.size()); + WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size())); // Then write the struct addresses. for (u16 addr : struct_addresses) { WriteU16(DspPipe::Audio, addr); diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h index 64d97f8ba..b714c0496 100644 --- a/src/audio_core/hle/pipe.h +++ b/src/audio_core/hle/pipe.h @@ -24,10 +24,14 @@ enum class DspPipe { constexpr size_t NUM_DSP_PIPE = 8; /** - * Read a DSP pipe. - * @param pipe_number The Pipe ID - * @param length How much data to request. - * @return The data read from the pipe. The size of this vector can be less than the length requested. + * Reads `length` bytes from the DSP pipe identified with `pipe_number`. + * @note Can read up to the maximum value of a u16 in bytes (65,535). + * @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty vector will be returned. + * @note IF `length` is set to 0, an empty vector will be returned. + * @note IF `length` is greater than the amount of data available, this function will only read the available amount. + * @param pipe_number a `DspPipe` + * @param length the number of bytes to read. The max is 65,535 (max of u16). + * @returns a vector of bytes from the specified pipe. On error, will be empty. */ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length); diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp index daaf6e3f3..30552fe26 100644 --- a/src/audio_core/hle/source.cpp +++ b/src/audio_core/hle/source.cpp @@ -126,13 +126,13 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_l if (config.simple_filter_dirty) { config.simple_filter_dirty.Assign(0); state.filters.Configure(config.simple_filter); - LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update"); + LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update", source_id); } if (config.biquad_filter_dirty) { config.biquad_filter_dirty.Assign(0); state.filters.Configure(config.biquad_filter); - LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update"); + LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update", source_id); } if (config.interpolation_dirty) { diff --git a/src/audio_core/sdl2_sink.cpp b/src/audio_core/sdl2_sink.cpp new file mode 100644 index 000000000..dc75c04ee --- /dev/null +++ b/src/audio_core/sdl2_sink.cpp @@ -0,0 +1,126 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <list> +#include <vector> + +#include <SDL.h> + +#include "audio_core/audio_core.h" +#include "audio_core/sdl2_sink.h" + +#include "common/assert.h" +#include "common/logging/log.h" +#include <numeric> + +namespace AudioCore { + +struct SDL2Sink::Impl { + unsigned int sample_rate = 0; + + SDL_AudioDeviceID audio_device_id = 0; + + std::list<std::vector<s16>> queue; + + static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes); +}; + +SDL2Sink::SDL2Sink() : impl(std::make_unique<Impl>()) { + if (SDL_Init(SDL_INIT_AUDIO) < 0) { + LOG_CRITICAL(Audio_Sink, "SDL_Init(SDL_INIT_AUDIO) failed"); + impl->audio_device_id = 0; + return; + } + + SDL_AudioSpec desired_audiospec; + SDL_zero(desired_audiospec); + desired_audiospec.format = AUDIO_S16; + desired_audiospec.channels = 2; + desired_audiospec.freq = native_sample_rate; + desired_audiospec.samples = 1024; + desired_audiospec.userdata = impl.get(); + desired_audiospec.callback = &Impl::Callback; + + SDL_AudioSpec obtained_audiospec; + SDL_zero(obtained_audiospec); + + impl->audio_device_id = SDL_OpenAudioDevice(nullptr, false, &desired_audiospec, &obtained_audiospec, 0); + if (impl->audio_device_id <= 0) { + LOG_CRITICAL(Audio_Sink, "SDL_OpenAudioDevice failed"); + return; + } + + impl->sample_rate = obtained_audiospec.freq; + + // SDL2 audio devices start out paused, unpause it: + SDL_PauseAudioDevice(impl->audio_device_id, 0); +} + +SDL2Sink::~SDL2Sink() { + if (impl->audio_device_id <= 0) + return; + + SDL_CloseAudioDevice(impl->audio_device_id); +} + +unsigned int SDL2Sink::GetNativeSampleRate() const { + if (impl->audio_device_id <= 0) + return native_sample_rate; + + return impl->sample_rate; +} + +void SDL2Sink::EnqueueSamples(const std::vector<s16>& samples) { + if (impl->audio_device_id <= 0) + return; + + ASSERT_MSG(samples.size() % 2 == 0, "Samples must be in interleaved stereo PCM16 format (size must be a multiple of two)"); + + SDL_LockAudioDevice(impl->audio_device_id); + impl->queue.emplace_back(samples); + SDL_UnlockAudioDevice(impl->audio_device_id); +} + +size_t SDL2Sink::SamplesInQueue() const { + if (impl->audio_device_id <= 0) + return 0; + + SDL_LockAudioDevice(impl->audio_device_id); + + size_t total_size = std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<size_t>(0), + [](size_t sum, const auto& buffer) { + // Division by two because each stereo sample is made of two s16. + return sum + buffer.size() / 2; + }); + + SDL_UnlockAudioDevice(impl->audio_device_id); + + return total_size; +} + +void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) { + Impl* impl = reinterpret_cast<Impl*>(impl_); + + size_t remaining_size = static_cast<size_t>(buffer_size_in_bytes) / sizeof(s16); // Keep track of size in 16-bit increments. + + while (remaining_size > 0 && !impl->queue.empty()) { + if (impl->queue.front().size() <= remaining_size) { + memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16)); + buffer += impl->queue.front().size() * sizeof(s16); + remaining_size -= impl->queue.front().size(); + impl->queue.pop_front(); + } else { + memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16)); + buffer += remaining_size * sizeof(s16); + impl->queue.front().erase(impl->queue.front().begin(), impl->queue.front().begin() + remaining_size); + remaining_size = 0; + } + } + + if (remaining_size > 0) { + memset(buffer, 0, remaining_size * sizeof(s16)); + } +} + +} // namespace AudioCore diff --git a/src/audio_core/sdl2_sink.h b/src/audio_core/sdl2_sink.h new file mode 100644 index 000000000..0f296b673 --- /dev/null +++ b/src/audio_core/sdl2_sink.h @@ -0,0 +1,30 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <cstddef> +#include <memory> + +#include "audio_core/sink.h" + +namespace AudioCore { + +class SDL2Sink final : public Sink { +public: + SDL2Sink(); + ~SDL2Sink() override; + + unsigned int GetNativeSampleRate() const override; + + void EnqueueSamples(const std::vector<s16>& samples) override; + + size_t SamplesInQueue() const override; + +private: + struct Impl; + std::unique_ptr<Impl> impl; +}; + +} // namespace AudioCore diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h index cad21a85e..1c881c3d2 100644 --- a/src/audio_core/sink.h +++ b/src/audio_core/sink.h @@ -19,7 +19,7 @@ public: virtual ~Sink() = default; /// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec) - virtual unsigned GetNativeSampleRate() const = 0; + virtual unsigned int GetNativeSampleRate() const = 0; /** * Feed stereo samples to sink. diff --git a/src/audio_core/sink_details.cpp b/src/audio_core/sink_details.cpp index d2cc74103..ba5e83d17 100644 --- a/src/audio_core/sink_details.cpp +++ b/src/audio_core/sink_details.cpp @@ -8,10 +8,17 @@ #include "audio_core/null_sink.h" #include "audio_core/sink_details.h" +#ifdef HAVE_SDL2 +#include "audio_core/sdl2_sink.h" +#endif + namespace AudioCore { // g_sink_details is ordered in terms of desirability, with the best choice at the top. const std::vector<SinkDetails> g_sink_details = { +#ifdef HAVE_SDL2 + { "sdl2", []() { return std::make_unique<SDL2Sink>(); } }, +#endif { "null", []() { return std::make_unique<NullSink>(); } }, }; diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp new file mode 100644 index 000000000..ea38f40d0 --- /dev/null +++ b/src/audio_core/time_stretch.cpp @@ -0,0 +1,144 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <chrono> +#include <cmath> +#include <vector> + +#include <SoundTouch.h> + +#include "audio_core/audio_core.h" +#include "audio_core/time_stretch.h" + +#include "common/common_types.h" +#include "common/logging/log.h" +#include "common/math_util.h" + +using steady_clock = std::chrono::steady_clock; + +namespace AudioCore { + +constexpr double MIN_RATIO = 0.1; +constexpr double MAX_RATIO = 100.0; + +static double ClampRatio(double ratio) { + return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO); +} + +constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds +constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds +constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples + +constexpr double SMOOTHING_FACTOR = 0.007; + +struct TimeStretcher::Impl { + soundtouch::SoundTouch soundtouch; + + steady_clock::time_point frame_timer = steady_clock::now(); + size_t samples_queued = 0; + + double smoothed_ratio = 1.0; + + double sample_rate = static_cast<double>(native_sample_rate); +}; + +std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) { + // This is a very simple algorithm without any fancy control theory. It works and is stable. + + double ratio = CalculateCurrentRatio(); + ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue); + impl->smoothed_ratio = (1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio; + impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio); + + // SoundTouch's tempo definition the inverse of our ratio definition. + impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio); + + std::vector<s16> samples = GetSamples(); + if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) { + samples.clear(); + LOG_DEBUG(Audio, "Dropping frames!"); + } + return samples; +} + +TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) { + impl->soundtouch.setPitch(1.0); + impl->soundtouch.setChannels(2); + impl->soundtouch.setSampleRate(native_sample_rate); + Reset(); +} + +TimeStretcher::~TimeStretcher() { + impl->soundtouch.clear(); +} + +void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) { + impl->sample_rate = static_cast<double>(sample_rate); + impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate); +} + +void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) { + impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples)); + impl->samples_queued += num_samples; +} + +void TimeStretcher::Flush() { + impl->soundtouch.flush(); +} + +void TimeStretcher::Reset() { + impl->soundtouch.setTempo(1.0); + impl->soundtouch.clear(); + impl->smoothed_ratio = 1.0; + impl->frame_timer = steady_clock::now(); + impl->samples_queued = 0; + SetOutputSampleRate(native_sample_rate); +} + +double TimeStretcher::CalculateCurrentRatio() { + const steady_clock::time_point now = steady_clock::now(); + const std::chrono::duration<double> duration = now - impl->frame_timer; + + const double expected_time = static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate); + const double actual_time = duration.count(); + + double ratio; + if (expected_time != 0) { + ratio = ClampRatio(actual_time / expected_time); + } else { + ratio = impl->smoothed_ratio; + } + + impl->frame_timer = now; + impl->samples_queued = 0; + + return ratio; +} + +double TimeStretcher::CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const { + const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate); + const size_t max_sample_delay = static_cast<size_t>(MAX_DELAY_TIME * impl->sample_rate); + + if (sample_delay < min_sample_delay) { + // Make the ratio bigger. + ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio); + } else if (sample_delay > max_sample_delay) { + // Make the ratio smaller. + ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio; + } + + return ClampRatio(ratio); +} + +std::vector<s16> TimeStretcher::GetSamples() { + uint available = impl->soundtouch.numSamples(); + + std::vector<s16> output(static_cast<size_t>(available) * 2); + + impl->soundtouch.receiveSamples(output.data(), available); + + return output; +} + +} // namespace AudioCore diff --git a/src/audio_core/time_stretch.h b/src/audio_core/time_stretch.h new file mode 100644 index 000000000..1fde3f72a --- /dev/null +++ b/src/audio_core/time_stretch.h @@ -0,0 +1,57 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include <cstddef> +#include <memory> +#include <vector> + +#include "common/common_types.h" + +namespace AudioCore { + +class TimeStretcher final { +public: + TimeStretcher(); + ~TimeStretcher(); + + /** + * Set sample rate for the samples that Process returns. + * @param sample_rate The sample rate. + */ + void SetOutputSampleRate(unsigned int sample_rate); + + /** + * Add samples to be processed. + * @param sample_buffer Buffer of samples in interleaved stereo PCM16 format. + * @param num_sample Number of samples. + */ + void AddSamples(const s16* sample_buffer, size_t num_samples); + + /// Flush audio remaining in internal buffers. + void Flush(); + + /// Resets internal state and clears buffers. + void Reset(); + + /** + * Does audio stretching and produces the time-stretched samples. + * Timer calculations use sample_delay to determine how much of a margin we have. + * @param sample_delay How many samples are buffered downstream of this module and haven't been played yet. + * @return Samples to play in interleaved stereo PCM16 format. + */ + std::vector<s16> Process(size_t sample_delay); + +private: + struct Impl; + std::unique_ptr<Impl> impl; + + /// INTERNAL: ratio = wallclock time / emulated time + double CalculateCurrentRatio(); + /// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate direction. + double CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const; + /// INTERNAL: Gets the time-stretched samples from SoundTouch. + std::vector<s16> GetSamples(); +}; + +} // namespace AudioCore |