dolphin-emulator/Source/Core/AudioCommon/PulseAudioStream.cpp
degasus 5c646d334a Pulseaudio: rewrite the pa backend with the async api
The default async api allow us to set some latency options. The old one (simple API) was the lazy way to go for usual audio where latency doesn't matter.

This also streams audio, so it should be a bit faster then the old one.
2014-02-07 00:20:10 +01:00

175 lines
4.7 KiB
C++

// Copyright 2013 Dolphin Emulator Project
// Licensed under GPLv2
// Refer to the license.txt file included.
#include <functional>
#include "Common.h"
#include "Thread.h"
#include "PulseAudioStream.h"
namespace
{
const size_t BUFFER_SAMPLES = 512; // ~10 ms
const size_t CHANNEL_COUNT = 2;
const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT * sizeof(s16);
}
PulseAudio::PulseAudio(CMixer *mixer)
: SoundStream(mixer)
, m_thread()
, m_run_thread()
{}
bool PulseAudio::Start()
{
m_run_thread = true;
m_thread = std::thread(std::mem_fun(&PulseAudio::SoundLoop), this);
return true;
}
void PulseAudio::Stop()
{
m_run_thread = false;
m_thread.join();
}
void PulseAudio::Update()
{
// don't need to do anything here.
}
// Called on audio thread.
void PulseAudio::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - pulse");
if (PulseInit())
{
while (m_run_thread && m_pa_connected == 1 && m_pa_error >= 0)
m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, NULL);
if(m_pa_error < 0)
ERROR_LOG(AUDIO, "PulseAudio error: %s", 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, NULL, PA_CONTEXT_NOFLAGS, NULL);
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, NULL);
if (m_pa_connected == 2 || m_pa_error < 0)
{
ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", 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;
ss.format = PA_SAMPLE_S16LE;
ss.channels = 2;
ss.rate = m_mixer->GetSampleRate();
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, NULL);
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_SIZE; // 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, NULL, &m_pa_ba, flags, NULL, NULL);
if (m_pa_error < 0)
{
ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
return false;
}
INFO_LOG(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_SIZE;
pa_stream_set_buffer_attr(s, &m_pa_ba, NULL, NULL);
WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
}
void PulseAudio::WriteCallback(pa_stream* s, size_t length)
{
// fetch dst buffer directly from pulseaudio, so no memcpy is needed
void* buffer;
m_pa_error = pa_stream_begin_write(s, &buffer, &length);
if (!buffer || m_pa_error < 0)
return; // error will be printed from main loop
m_mixer->Mix((s16*) buffer, length / sizeof(s16) / CHANNEL_COUNT);
m_pa_error = pa_stream_write(s, buffer, length, NULL, 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);
}