dolphin/Source/Core/AudioCommon/Src/AlsaSoundStream.cpp

231 lines
5.7 KiB
C++

// Copyright (C) 2003 Dolphin Project.
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License 2.0 for more details.
// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/
// Official SVN repository and contact information can be found at
// http://code.google.com/p/dolphin-emu/
#include "Common.h"
#include "Thread.h"
#include "AlsaSoundStream.h"
#define FRAME_COUNT_MIN 256
#define BUFFER_SIZE_MAX 8192
#define BUFFER_SIZE_BYTES (BUFFER_SIZE_MAX*2*2)
AlsaSound::AlsaSound(CMixer *mixer) : SoundStream(mixer), thread_data(0), handle(NULL), frames_to_deliver(FRAME_COUNT_MIN)
{
mix_buffer = new u8[BUFFER_SIZE_BYTES];
}
AlsaSound::~AlsaSound()
{
delete [] mix_buffer;
}
static void *ThreadTrampoline(void *args)
{
reinterpret_cast<AlsaSound *>(args)->SoundLoop();
return NULL;
}
bool AlsaSound::Start()
{
thread = new Common::Thread(&ThreadTrampoline, this);
thread_data = 0;
return true;
}
void AlsaSound::Stop()
{
thread_data = 1;
delete thread;
thread = NULL;
}
void AlsaSound::Update()
{
// don't need to do anything here.
}
// Called on audio thread.
void AlsaSound::SoundLoop()
{
if (!AlsaInit()) {
thread_data = 2;
return;
}
while (!thread_data)
{
m_mixer->Mix(reinterpret_cast<short *>(mix_buffer), frames_to_deliver);
int rc = m_muted ? 1337 : 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));
}
}
AlsaShutdown();
thread_data = 2;
}
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\n", 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\n", 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\n", 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\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
if (err < 0)
{
ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, 2);
if (err < 0)
{
ERROR_LOG(AUDIO, "Channels count not available: %s\n", 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 Minimum periods: %s\n", 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 minimum buffer size: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", 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\n", 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\n", 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 %d sample \"hardware\" buffer with %d periods. Will send %d samples per fragments.\n", 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\n", 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\n", snd_strerror(err));
return false;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0)
{
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
return false;
}
err = snd_pcm_prepare(handle);
if (err < 0)
{
ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
return false;
}
NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
return true;
}
void AlsaSound::AlsaShutdown()
{
if (handle != NULL)
{
snd_pcm_drop(handle);
snd_pcm_close(handle);
handle = NULL;
}
}