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// Copyright 2009 Dolphin Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <cstring>
#include "AudioCommon/PulseAudioStream.h"
#include "Common/Assert.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
#include "Common/Thread.h"
#include "Core/Config/MainSettings.h"
namespace
{
const size_t BUFFER_SAMPLES = 512; // ~10 ms - needs to be at least 240 for surround
}
PulseAudio::PulseAudio() = default;
bool PulseAudio::Init()
{
m_stereo = !Config::ShouldUseDPL2Decoder();
m_channels = m_stereo ? 2 : 6; // will tell PA we use a Stereo or 5.0 channel setup
NOTICE_LOG_FMT(AUDIO, "PulseAudio backend using {} channels", m_channels);
m_run_thread.Set();
m_thread = std::thread(&PulseAudio::SoundLoop, this);
return true;
}
PulseAudio::~PulseAudio()
{
m_run_thread.Clear();
m_thread.join();
}
// Called on audio thread.
void PulseAudio::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - pulse");
if (PulseInit())
{
while (m_run_thread.IsSet() && m_pa_connected == 1 && m_pa_error >= 0)
m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
if (m_pa_error < 0)
ERROR_LOG_FMT(AUDIO, "PulseAudio error: {}", pa_strerror(m_pa_error));
PulseShutdown();
}
}
bool PulseAudio::PulseInit()
{
m_pa_error = 0;
m_pa_connected = 0;
// create pulseaudio main loop and context
// also register the async state callback which is called when the connection to the pa server has
// changed
m_pa_ml = pa_mainloop_new();
m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
// wait until we're connected to the pulseaudio server
while (m_pa_connected == 0 && m_pa_error >= 0)
m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
if (m_pa_connected == 2 || m_pa_error < 0)
{
ERROR_LOG_FMT(AUDIO, "PulseAudio failed to initialize: {}", pa_strerror(m_pa_error));
return false;
}
// create a new audio stream with our sample format
// also connect the callbacks for this stream
pa_sample_spec ss;
pa_channel_map channel_map;
pa_channel_map* channel_map_p = nullptr; // auto channel map
if (m_stereo)
{
ss.format = PA_SAMPLE_S16LE;
m_bytespersample = sizeof(s16);
}
else
{
// surround is remixed in floats, use a float PA buffer to save another conversion
ss.format = PA_SAMPLE_FLOAT32NE;
m_bytespersample = sizeof(float);
channel_map_p = &channel_map; // explicit channel map:
channel_map.channels = 6;
channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
channel_map.map[3] = PA_CHANNEL_POSITION_LFE;
channel_map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
channel_map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
}
ss.channels = m_channels;
ss.rate = m_mixer->GetSampleRate();
ASSERT(pa_sample_spec_valid(&ss));
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, channel_map_p);
pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
// connect this audio stream to the default audio playback
// limit buffersize to reduce latency
m_pa_ba.fragsize = -1;
m_pa_ba.maxlength = -1; // max buffer, so also max latency
m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_
m_pa_ba.prebuf = -1; // start as early as possible
m_pa_ba.tlength =
BUFFER_SAMPLES * m_channels *
m_bytespersample; // designed latency, only change this flag for low latency output
pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_AUTO_TIMING_UPDATE);
m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
if (m_pa_error < 0)
{
ERROR_LOG_FMT(AUDIO, "PulseAudio failed to initialize: {}", pa_strerror(m_pa_error));
return false;
}
INFO_LOG_FMT(AUDIO, "Pulse successfully initialized");
return true;
}
void PulseAudio::PulseShutdown()
{
pa_context_disconnect(m_pa_ctx);
pa_context_unref(m_pa_ctx);
pa_mainloop_free(m_pa_ml);
}
void PulseAudio::StateCallback(pa_context* c)
{
pa_context_state_t state = pa_context_get_state(c);
switch (state)
{
case PA_CONTEXT_FAILED:
case PA_CONTEXT_TERMINATED:
m_pa_connected = 2;
break;
case PA_CONTEXT_READY:
m_pa_connected = 1;
break;
default:
break;
}
}
// on underflow, increase pulseaudio latency in ~10ms steps
void PulseAudio::UnderflowCallback(pa_stream* s)
{
m_pa_ba.tlength += BUFFER_SAMPLES * m_channels * m_bytespersample;
pa_operation* op = pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
pa_operation_unref(op);
WARN_LOG_FMT(AUDIO, "pulseaudio underflow, new latency: {} bytes", m_pa_ba.tlength);
}
void PulseAudio::WriteCallback(pa_stream* s, size_t length)
{
int bytes_per_frame = m_channels * m_bytespersample;
int frames = (length / bytes_per_frame);
size_t trunc_length = frames * bytes_per_frame;
// fetch dst buffer directly from pulseaudio, so no memcpy is needed
void* buffer;
m_pa_error = pa_stream_begin_write(s, &buffer, &trunc_length);
if (!buffer || m_pa_error < 0)
return; // error will be printed from main loop
if (m_stereo)
{
// use the raw s16 stereo mix
m_mixer->Mix((s16*)buffer, frames);
}
else
{
if (m_channels == 6) // Extract dpl2/5.1 Surround
{
m_mixer->MixSurround((float*)buffer, frames);
}
else
{
ERROR_LOG_FMT(AUDIO, "Unsupported number of PA channels requested: {}", m_channels);
return;
}
}
m_pa_error = pa_stream_write(s, buffer, trunc_length, nullptr, 0, PA_SEEK_RELATIVE);
}
// Callbacks that forward to internal methods (required because PulseAudio is a C API).
void PulseAudio::StateCallback(pa_context* c, void* userdata)
{
PulseAudio* p = (PulseAudio*)userdata;
p->StateCallback(c);
}
void PulseAudio::UnderflowCallback(pa_stream* s, void* userdata)
{
PulseAudio* p = (PulseAudio*)userdata;
p->UnderflowCallback(s);
}
void PulseAudio::WriteCallback(pa_stream* s, size_t length, void* userdata)
{
PulseAudio* p = (PulseAudio*)userdata;
p->WriteCallback(s, length);
}
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