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audio_core: Simplify sink interface

This commit is contained in:
MerryMage 2018-09-08 21:07:28 +01:00
parent 761ef78408
commit f34711219a
8 changed files with 42 additions and 131 deletions
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40
src/audio_core/cubeb_sink.cpp
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@ -13,13 +13,11 @@ namespace AudioCore {
struct CubebSink::Impl {
unsigned int sample_rate = 0;
std::vector<std::string> device_list;
cubeb* ctx = nullptr;
cubeb_stream* stream = nullptr;
std::mutex queue_mutex;
std::vector<s16> queue;
std::function<void(s16*, std::size_t)> cb;
static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames);
@ -95,45 +93,19 @@ unsigned int CubebSink::GetNativeSampleRate() const {
return impl->sample_rate;
}
void CubebSink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
if (!impl->ctx)
return;
std::lock_guard lock{impl->queue_mutex};
impl->queue.reserve(impl->queue.size() + sample_count * 2);
std::copy(samples, samples + sample_count * 2, std::back_inserter(impl->queue));
}
size_t CubebSink::SamplesInQueue() const {
if (!impl->ctx)
return 0;
std::lock_guard lock{impl->queue_mutex};
return impl->queue.size() / 2;
void CubebSink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
impl->cb = cb;
}
long CubebSink::Impl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames) {
Impl* impl = static_cast<Impl*>(user_data);
u8* buffer = reinterpret_cast<u8*>(output_buffer);
s16* buffer = reinterpret_cast<s16*>(output_buffer);
if (!impl)
if (!impl || !impl->cb)
return 0;
std::lock_guard lock{impl->queue_mutex};
std::size_t frames_to_write =
std::min(impl->queue.size() / 2, static_cast<std::size_t>(num_frames));
memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * 2);
impl->queue.erase(impl->queue.begin(), impl->queue.begin() + frames_to_write * 2);
if (frames_to_write < num_frames) {
// Fill the rest of the frames with silence
memset(buffer + frames_to_write * sizeof(s16) * 2, 0,
(num_frames - frames_to_write) * sizeof(s16) * 2);
}
impl->cb(buffer, num_frames);
return num_frames;
}

4
src/audio_core/cubeb_sink.h
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@ -17,9 +17,7 @@ public:
unsigned int GetNativeSampleRate() const override;
void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
std::size_t SamplesInQueue() const override;
void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
private:
struct Impl;

44
src/audio_core/dsp_interface.cpp
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@ -12,16 +12,13 @@
namespace AudioCore {
DspInterface::DspInterface() = default;
DspInterface::~DspInterface() {
if (perform_time_stretching) {
FlushResidualStretcherAudio();
}
}
DspInterface::~DspInterface() = default;
void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) {
const SinkDetails& sink_details = GetSinkDetails(sink_id);
sink = sink_details.factory(audio_device);
sink->SetCallback(
[this](s16* buffer, std::size_t num_frames) { OutputCallback(buffer, num_frames); });
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
@ -51,32 +48,21 @@ void DspInterface::OutputFrame(StereoFrame16& frame) {
frame[i][1] = static_cast<s16>(frame[i][1] * volume_scale_factor);
}
if (perform_time_stretching) {
time_stretcher.AddSamples(&frame[0][0], frame.size());
std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
} else {
constexpr std::size_t maximum_sample_latency = 2048; // about 64 miliseconds
if (sink->SamplesInQueue() > maximum_sample_latency) {
// This can occur if we're running too fast and samples are starting to back up.
// Just drop the samples.
return;
}
sink->EnqueueSamples(&frame[0][0], frame.size());
}
fifo.Push(frame.data(), frame.size());
}
void DspInterface::FlushResidualStretcherAudio() {
if (!sink)
return;
void DspInterface::FlushResidualStretcherAudio() {}
time_stretcher.Flush();
while (true) {
std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
if (residual_audio.empty())
break;
sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
void DspInterface::OutputCallback(s16* buffer, size_t num_frames) {
const size_t frames_written = fifo.Pop(buffer, num_frames);
if (frames_written > 0) {
std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
}
// Hold last emitted frame; this prevents popping.
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16));
}
}

4
src/audio_core/dsp_interface.h
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@ -9,6 +9,7 @@
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/ring_buffer.h"
#include "core/memory.h"
namespace Service {
@ -81,9 +82,12 @@ protected:
private:
void FlushResidualStretcherAudio();
void OutputCallback(s16* buffer, std::size_t num_frames);
std::unique_ptr<Sink> sink;
bool perform_time_stretching = false;
Common::RingBuffer<s16, 0x2000, 2> fifo;
std::array<s16, 2> last_frame{};
TimeStretcher time_stretcher;
};

6
src/audio_core/null_sink.h
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@ -19,11 +19,7 @@ public:
return native_sample_rate;
}
void EnqueueSamples(const s16*, std::size_t) override {}
std::size_t SamplesInQueue() const override {
return 0;
}
void SetCallback(std::function<void(s16*, std::size_t)>) override {}
};
} // namespace AudioCore

58
src/audio_core/sdl2_sink.cpp
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@ -2,8 +2,8 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <list>
#include <numeric>
#include <string>
#include <vector>
#include <SDL.h>
#include "audio_core/audio_types.h"
#include "audio_core/sdl2_sink.h"
@ -17,7 +17,7 @@ struct SDL2Sink::Impl {
SDL_AudioDeviceID audio_device_id = 0;
std::list<std::vector<s16>> queue;
std::function<void(s16*, std::size_t)> cb;
static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes);
};
@ -74,58 +74,18 @@ unsigned int SDL2Sink::GetNativeSampleRate() const {
return impl->sample_rate;
}
void SDL2Sink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
if (impl->audio_device_id <= 0)
return;
SDL_LockAudioDevice(impl->audio_device_id);
impl->queue.emplace_back(samples, samples + sample_count * 2);
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);
std::size_t total_size =
std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<std::size_t>(0),
[](std::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::SetCallback(std::function<void(s16*, std::size_t)> cb) {
impl->cb = cb;
}
void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
Impl* impl = reinterpret_cast<Impl*>(impl_);
if (!impl || !impl->cb)
return;
std::size_t remaining_size = static_cast<std::size_t>(buffer_size_in_bytes) /
sizeof(s16); // Keep track of size in 16-bit increments.
const size_t num_frames = buffer_size_in_bytes / (2 * sizeof(s16));
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));
}
impl->cb(reinterpret_cast<s16*>(buffer), num_frames);
}
std::vector<std::string> ListSDL2SinkDevices() {

4
src/audio_core/sdl2_sink.h
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@ -17,9 +17,7 @@ public:
unsigned int GetNativeSampleRate() const override;
void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
std::size_t SamplesInQueue() const override;
void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
private:
struct Impl;

13
src/audio_core/sink.h
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@ -4,7 +4,7 @@
#pragma once
#include <vector>
#include <functional>
#include "common/common_types.h"
namespace AudioCore {
@ -20,19 +20,16 @@ class Sink {
public:
virtual ~Sink() = default;
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units:
/// samples/sec)
/// The native rate of this sink. The sink expects to be fed samples that respect this.
/// (Units: samples/sec)
virtual unsigned int GetNativeSampleRate() const = 0;
/**
* Feed stereo samples to sink.
* Set callback for samples
* @param samples Samples in interleaved stereo PCM16 format.
* @param sample_count Number of samples.
*/
virtual void EnqueueSamples(const s16* samples, std::size_t sample_count) = 0;
/// Samples enqueued that have not been played yet.
virtual std::size_t SamplesInQueue() const = 0;
virtual void SetCallback(std::function<void(s16*, std::size_t)> cb) = 0;
};
} // namespace AudioCore