diff options
-rw-r--r-- | src/audio_core/CMakeLists.txt | 2 | ||||
-rw-r--r-- | src/audio_core/hle/dsp.cpp | 16 | ||||
-rw-r--r-- | src/audio_core/time_stretch.cpp | 144 | ||||
-rw-r--r-- | src/audio_core/time_stretch.h | 57 | ||||
-rw-r--r-- | src/core/hle/kernel/memory.cpp | 3 | ||||
-rw-r--r-- | src/core/hle/service/dsp_dsp.cpp | 4 |
6 files changed, 224 insertions, 2 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index 13b5e400e..eba0a5697 100644 --- a/src/audio_core/CMakeLists.txt +++ b/src/audio_core/CMakeLists.txt @@ -7,6 +7,7 @@ set(SRCS hle/source.cpp interpolate.cpp sink_details.cpp + time_stretch.cpp ) set(HEADERS @@ -21,6 +22,7 @@ set(HEADERS null_sink.h sink.h sink_details.h + time_stretch.h ) include_directories(../../externals/soundtouch/include) diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp index 0cdbdb06a..5113ad8ca 100644 --- a/src/audio_core/hle/dsp.cpp +++ b/src/audio_core/hle/dsp.cpp @@ -9,6 +9,7 @@ #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 { @@ -48,15 +49,29 @@ static std::array<Source, num_sources> sources = { }; static std::unique_ptr<AudioCore::Sink> sink; +static AudioCore::TimeStretcher time_stretcher; void Init() { DSP::HLE::ResetPipes(); + for (auto& source : sources) { source.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() { @@ -77,6 +92,7 @@ bool Tick() { 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/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 diff --git a/src/core/hle/kernel/memory.cpp b/src/core/hle/kernel/memory.cpp index 4be20db22..17ae87aef 100644 --- a/src/core/hle/kernel/memory.cpp +++ b/src/core/hle/kernel/memory.cpp @@ -55,6 +55,9 @@ void MemoryInit(u32 mem_type) { memory_regions[i].size = memory_region_sizes[mem_type][i]; memory_regions[i].used = 0; memory_regions[i].linear_heap_memory = std::make_shared<std::vector<u8>>(); + // Reserve enough space for this region of FCRAM. + // We do not want this block of memory to be relocated when allocating from it. + memory_regions[i].linear_heap_memory->reserve(memory_regions[i].size); base += memory_regions[i].size; } diff --git a/src/core/hle/service/dsp_dsp.cpp b/src/core/hle/service/dsp_dsp.cpp index 274fc751a..10730d7ac 100644 --- a/src/core/hle/service/dsp_dsp.cpp +++ b/src/core/hle/service/dsp_dsp.cpp @@ -440,9 +440,9 @@ static void GetHeadphoneStatus(Service::Interface* self) { cmd_buff[0] = IPC::MakeHeader(0x1F, 2, 0); cmd_buff[1] = RESULT_SUCCESS.raw; // No error - cmd_buff[2] = 0; // Not using headphones? + cmd_buff[2] = 0; // Not using headphones - LOG_WARNING(Service_DSP, "(STUBBED) called"); + LOG_DEBUG(Service_DSP, "called"); } /** |