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Linux: Better ALSA support

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This removes some bad behaviour, like the audio thread busy waiting inside ALSA code,
missing begins of voice samples(due to DSP_HLE code not getting called often enough).
Turns out my understanding of ALSA API was lacking.

Tested with pulse plugin and directly on my intel-hda/realtek hardware.



git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@5461 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
pierre
2010-05-21 22:48:57 +00:00
parent 3ff6a8bd68
commit ab89decac5
2 changed files with 39 additions and 20 deletions
Download Patch File Download Diff File Expand all files Collapse all files

58
Source/Core/AudioCommon/Src/AlsaSoundStream.cpp
View File

@ -19,10 +19,11 @@
#include "Thread.h" #include "Thread.h"
#include "AlsaSoundStream.h" #include "AlsaSoundStream.h"
#define BUFFER_SIZE 4096 #define FRAME_COUNT_MIN 256
#define BUFFER_SIZE_BYTES (BUFFER_SIZE*2*2) #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) AlsaSound::AlsaSound(CMixer *mixer) : SoundStream(mixer), thread_data(0), handle(NULL), frames_to_deliver(FRAME_COUNT_MIN)
{ {
mix_buffer = new u8[BUFFER_SIZE_BYTES]; mix_buffer = new u8[BUFFER_SIZE_BYTES];
} }
@ -60,9 +61,10 @@ void AlsaSound::Update()
// Called on audio thread. // Called on audio thread.
void AlsaSound::SoundLoop() void AlsaSound::SoundLoop()
{ {
AlsaInit(); if (!AlsaInit()) {
// nakee: What is the optimal value? thread_data = 2;
int frames_to_deliver = BUFFER_SIZE; return;
}
while (!thread_data) while (!thread_data)
{ {
m_mixer->Mix(reinterpret_cast<short *>(mix_buffer), frames_to_deliver); m_mixer->Mix(reinterpret_cast<short *>(mix_buffer), frames_to_deliver);
@ -88,7 +90,7 @@ bool AlsaSound::AlsaInit()
int dir; int dir;
snd_pcm_sw_params_t *swparams; snd_pcm_sw_params_t *swparams;
snd_pcm_hw_params_t *hwparams; snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t buffer_size,buffer_size_max;
unsigned int periods; unsigned int periods;
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
@ -135,31 +137,22 @@ bool AlsaSound::AlsaInit()
return false; return false;
} }
periods = 2; periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &periods, &dir); err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
if (err < 0) if (err < 0)
{ {
ERROR_LOG(AUDIO, "Cannot set Minimum periods: %s\n", snd_strerror(err)); ERROR_LOG(AUDIO, "Cannot set Minimum periods: %s\n", snd_strerror(err));
return false; return false;
} }
buffer_size = BUFFER_SIZE; buffer_size_max = BUFFER_SIZE_MAX;
err = snd_pcm_hw_params_set_buffer_size_min(handle, hwparams, &buffer_size); err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
if (err < 0) if (err < 0)
{ {
ERROR_LOG(AUDIO, "Cannot set minimum buffer size: %s\n", snd_strerror(err)); ERROR_LOG(AUDIO, "Cannot set minimum buffer size: %s\n", snd_strerror(err));
return false; return false;
} }
buffer_size = BUFFER_SIZE*2;
err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
if (err < 0)
{
ERROR_LOG(AUDIO, "Cannot set buffer size: %s\n", snd_strerror(err));
return false;
}
NOTICE_LOG(AUDIO, "ALSA gave us %d \"hardware\" buffers of %d samples.\n", periods, buffer_size);
err = snd_pcm_hw_params(handle, hwparams); err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) if (err < 0)
{ {
@ -167,6 +160,31 @@ bool AlsaSound::AlsaInit()
return false; 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); snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams); err = snd_pcm_sw_params_current(handle, swparams);

1
Source/Core/AudioCommon/Src/AlsaSoundStream.h
View File

@ -59,6 +59,7 @@ private:
volatile int thread_data; volatile int thread_data;
snd_pcm_t *handle; snd_pcm_t *handle;
int frames_to_deliver;
#else #else
public: public:
AlsaSound(CMixer *mixer) : SoundStream(mixer) {} AlsaSound(CMixer *mixer) : SoundStream(mixer) {}