dolphin/Source/Core/AudioCommon/AlsaSoundStream.cpp

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// Copyright 2009 Dolphin Emulator Project
2015-05-17 23:08:10 +00:00
// Licensed under GPLv2+
// Refer to the license.txt file included.
#include <mutex>
#include "AudioCommon/AlsaSoundStream.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
#include "Common/Thread.h"
AlsaSound::AlsaSound()
: m_thread_status(ALSAThreadStatus::STOPPED), handle(nullptr),
frames_to_deliver(FRAME_COUNT_MIN)
{
}
AlsaSound::~AlsaSound()
{
m_thread_status.store(ALSAThreadStatus::STOPPING);
// Give the opportunity to the audio thread
// to realize we are stopping the emulation
cv.notify_one();
thread.join();
}
bool AlsaSound::Init()
{
m_thread_status.store(ALSAThreadStatus::PAUSED);
if (!AlsaInit())
{
m_thread_status.store(ALSAThreadStatus::STOPPED);
return false;
}
thread = std::thread(&AlsaSound::SoundLoop, this);
return true;
}
void AlsaSound::Update()
{
// don't need to do anything here.
}
// Called on audio thread.
void AlsaSound::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - alsa");
while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
{
while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
{
m_mixer->Mix(mix_buffer, frames_to_deliver);
int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
if (rc == -EPIPE)
{
// Underrun
snd_pcm_prepare(handle);
}
else if (rc < 0)
{
ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
}
}
if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
{
snd_pcm_drop(handle); // Stop sound output
// Block until thread status changes.
std::unique_lock<std::mutex> lock(cv_m);
cv.wait(lock, [this] { return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
snd_pcm_prepare(handle); // resume sound output
}
}
AlsaShutdown();
m_thread_status.store(ALSAThreadStatus::STOPPED);
}
bool AlsaSound::SetRunning(bool running)
{
m_thread_status.store(running ? ALSAThreadStatus::RUNNING : ALSAThreadStatus::PAUSED);
cv.notify_one(); // Notify thread that status has changed
return true;
}
bool AlsaSound::AlsaInit()
{
unsigned int sample_rate = m_mixer->GetSampleRate();
int err;
int dir;
snd_pcm_sw_params_t* swparams;
snd_pcm_hw_params_t* hwparams;
snd_pcm_uframes_t buffer_size, buffer_size_max;
unsigned int periods;
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0)
{
ERROR_LOG(AUDIO, "Audio open error: %s", snd_strerror(err));
return false;
}
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
{
ERROR_LOG(AUDIO, "Access type not available: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
if (err < 0)
{
ERROR_LOG(AUDIO, "Sample format not available: %s", snd_strerror(err));
return false;
}
dir = 0;
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Rate not available: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
if (err < 0)
{
ERROR_LOG(AUDIO, "Channels count not available: %s", snd_strerror(err));
return false;
}
periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s", snd_strerror(err));
return false;
}
buffer_size_max = BUFFER_SIZE_MAX;
err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to install hw params: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get buffer size: %s", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot get periods: %s", snd_strerror(err));
return false;
}
// periods is the number of fragments alsa can wait for during one
// buffer_size
frames_to_deliver = buffer_size / periods;
// limit the minimum size. pulseaudio advertises a minimum of 32 samples.
if (frames_to_deliver < FRAME_COUNT_MIN)
frames_to_deliver = FRAME_COUNT_MIN;
// it is probably a bad idea to try to send more than one buffer of data
if ((unsigned int)frames_to_deliver > buffer_size)
frames_to_deliver = buffer_size;
NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d "
"samples per fragments.",
buffer_size, periods, frames_to_deliver);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot init sw params: %s", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set start thresh: %s", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set sw params: %s", snd_strerror(err));
return false;
}
err = snd_pcm_prepare(handle);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to prepare: %s", snd_strerror(err));
return false;
}
NOTICE_LOG(AUDIO, "ALSA successfully initialized.");
return true;
}
void AlsaSound::AlsaShutdown()
{
if (handle != nullptr)
{
snd_pcm_drop(handle);
snd_pcm_close(handle);
handle = nullptr;
}
}