dolphin/Source/Core/AudioCommon/PulseAudioStream.cpp

174 lines
4.7 KiB
C++
Raw Normal View History

// Copyright 2013 Dolphin Emulator Project
// Licensed under GPLv2
// Refer to the license.txt file included.
#include "AudioCommon/PulseAudioStream.h"
#include "Common/CommonTypes.h"
#include "Common/Thread.h"
2013-01-09 23:39:19 +00:00
namespace
{
const size_t BUFFER_SAMPLES = 512; // ~10 ms
2013-01-09 23:39:19 +00:00
const size_t CHANNEL_COUNT = 2;
const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT * sizeof(s16);
}
2013-01-09 23:39:19 +00:00
PulseAudio::PulseAudio(CMixer *mixer)
: SoundStream(mixer)
, m_thread()
, m_run_thread()
2014-08-30 20:29:15 +00:00
{
}
bool PulseAudio::Start()
{
m_run_thread = true;
m_thread = std::thread(&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");
2013-01-09 23:39:19 +00:00
if (PulseInit())
{
2014-02-03 21:52:34 +00:00
while (m_run_thread.load() && 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(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
2013-01-09 23:39:19 +00:00
PulseShutdown();
}
}
2013-01-09 23:39:19 +00:00
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(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;
2013-01-09 23:39:19 +00:00
ss.format = PA_SAMPLE_S16LE;
ss.channels = 2;
2013-01-14 04:20:33 +00:00
ss.rate = m_mixer->GetSampleRate();
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, nullptr);
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, nullptr, &m_pa_ba, flags, nullptr, nullptr);
if (m_pa_error < 0)
{
ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
2013-01-09 23:39:19 +00:00
return false;
}
INFO_LOG(AUDIO, "Pulse successfully initialized");
return true;
}
2013-01-09 23:39:19 +00:00
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, nullptr, nullptr);
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, 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);
}