First sounds from linux (see followup post to devs list)

This commit is contained in:
Aaron VonderHaar
2003-02-23 08:39:29 +00:00
parent 996f89885d
commit 7c7356ed7d
2 changed files with 415 additions and 0 deletions
@@ -0,0 +1,340 @@
#include "global.h"
#include "RageSoundDriver_ALSA9.h"
#include "RageTimer.h"
#include "RageLog.h"
#include "RageSound.h"
#include "RageUtil.h"
#include "SDL_utils.h"
const int channels = 2;
const int samplesize = channels*2; /* 16-bit */
const int samplerate = 44100;
const int buffersize_frames = 1024*8; /* in frames */
const int buffersize = buffersize_frames * samplesize; /* in bytes */
/**
* int err; must be defined before using this macro
*/
#define ALSA_ASSERT(x) \
if (err < 0) \
{ \
LOG->Trace("RageSound_ALSA9: ASSERT %s: %s", \
x, snd_strerror(err)); \
}
int RageSound_ALSA9::MixerThread_start(void *p)
{
((RageSound_ALSA9 *) p)->MixerThread();
return 0;
}
void RageSound_ALSA9::MixerThread()
{
/* SOUNDMAN will be set once RageSoundManager's ctor returns and
* assigns it; we might get here before that happens, though. */
while(!SOUNDMAN && !shutdown) SDL_Delay(10);
while(!shutdown) {
GetData();
//SDL_Delay(10);
}
}
bool RageSound_ALSA9::GetData()
{
LockMutex L(SOUNDMAN->lock);
int err;
/* Create a 32-bit buffer to mix sounds. */
static Sint16 *buf = NULL, *buf2 = NULL;
int bufsize = buffersize_frames * channels;
if (!buf)
{
buf = new Sint16[bufsize];
buf2 = new Sint16[bufsize];
}
memset(buf, 0, bufsize*sizeof(Uint16));
memset(buf2, 0, bufsize*sizeof(Uint16));
SoundMixBuffer mix;
for(unsigned i = 0; i < sounds.size(); ++i)
{
if(sounds[i]->stopping)
continue;
/* Call the callback. */
unsigned got = sounds[i]->snd->GetPCM((char *) buf, bufsize, last_cursor_pos);
mix.write((Sint16 *) buf, got/2);
if(got < bufsize)
{
/* This sound is finishing. */
sounds[i]->stopping = true;
sounds[i]->flush_pos = last_cursor_pos + (got / samplesize);
}
}
mix.read((Sint16*)buf2);
int r;
Sint16 *start = buf2;
/* it doesn't make sense to me why we must divide by two
* but we must */
snd_pcm_uframes_t remaining = buffersize_frames/2;
while (remaining > 0)
{
r = snd_pcm_mmap_writei(pcm, start, remaining);
if (r == -EAGAIN) {
continue;
}
if (r <= 0)
{
LOG->Trace("RageSoundDriver_ALSA9::GetData: snd_pcm_mmap_writei: %s", snd_strerror(r));
Recover(r);
break;
}
start += r*channels;
remaining -= r;
last_cursor_pos += r;
}
return true;
}
/**
* When the play buffer underruns, subsequent writes to the buffer
* return -EPIPE. When this happens, call Recover() to restart
* playback.
*/
void RageSound_ALSA9::Recover(int r)
{
int err;
if (r == -EPIPE)
{
LOG->Trace("RageSound_ALSA9::Recover (prepare)");
err = snd_pcm_prepare(pcm);
ALSA_ASSERT("snd_pcm_prepare (Recover)");
}
else /* r == -ESTRPIPE */
{
LOG->Trace("RageSound_ALSA9::Recover (resume)");
while ((err = snd_pcm_resume(pcm)) == -EAGAIN)
SDL_Delay(10);
ALSA_ASSERT("snd_pcm_resume (Recover)");
}
}
void RageSound_ALSA9::StartMixing(RageSound *snd)
{
sound *s = new sound;
s->snd = snd;
SDL_LockAudio();
sounds.push_back(s);
SDL_UnlockAudio();
}
void RageSound_ALSA9::Update(float delta)
{
LockMutex L(SOUNDMAN->lock);
/* SoundStopped might erase sounds out from under us, so make a copy
* of the sound list. */
vector<sound *> snds = sounds;
for(unsigned i = 0; i < snds.size(); ++i)
{
if(!sounds[i]->stopping) continue;
if(GetPosition(snds[i]->snd) < sounds[i]->flush_pos)
continue; /* stopping but still flushing */
/* This sound is done. */
snds[i]->snd->StopPlaying();
}
}
void RageSound_ALSA9::StopMixing(RageSound *snd)
{
LockMutex L(SOUNDMAN->lock);
/* Find the sound. */
unsigned i;
for(i = 0; i < sounds.size(); ++i)
if(sounds[i]->snd == snd) break;
if(i == sounds.size())
{
LOG->Trace("not stopping a sound because it's not playing");
return;
}
delete sounds[i];
sounds.erase(sounds.begin()+i, sounds.begin()+i+1);
}
int RageSound_ALSA9::GetPosition(const RageSound *snd) const
{
/* what's below tries to use the status time of the pcm device
* but since we use last_cursor_pos for the GetPCM() call,
* it doesn't match up with what RageSound is expecting
*/
LockMutex L(SOUNDMAN->lock);
int err;
snd_pcm_status_t *status;
//snd_timestamp_t now;
snd_pcm_sframes_t delay;
snd_pcm_status_alloca(&status);
err = snd_pcm_status(pcm, status);
ALSA_ASSERT("snd_pcm_status");
snd_pcm_state_t state = snd_pcm_status_get_state(status);
if (state == SND_PCM_STATE_PREPARED) return 0;
//snd_pcm_status_get_tstamp(status, &now);
delay = snd_pcm_status_get_delay(status);
//LOG->Trace("RageSound_ALSA9::GetPosition %d", last_cursor_pos - delay);
return last_cursor_pos - delay;
/* we need startup_time else we will be returning
* samples-since-1970, which will surely overflow int
*/
//float this_sec = (now.tv_sec - startup_time.tv_sec)
// + ((float)now.tv_usec)/1000000;
//LOG->Trace("RageSound_ALSA9::GetPosition %d", (int)(this_sec * samplerate)+800000);
/* the +800000 was a fudge to compensate for the difference
* between this and last_cursor_pos */
//return (int)(this_sec * samplerate)+800000;
}
RageSound_ALSA9::RageSound_ALSA9()
{
shutdown = false;
int err;
last_cursor_pos = 0;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
/* open the device */
// if we instead use plughw: then our requested format WILL be provided
err = snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0);
ALSA_ASSERT("snd_pcm_open");
/* allocate the hardware parameters structure */
err = snd_pcm_hw_params_malloc(&hwparams);
ALSA_ASSERT("snd_pcm_hw_params_malloc");
/* not exactly sure what this does */
err = snd_pcm_hw_params_any(pcm, hwparams);
ALSA_ASSERT("snd_pcm_hw_params_any");
/* set to mmap mode (with channels interleaved) */
err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED);
ALSA_ASSERT("snd_pcm_hw_params_set_access");
/* set PCM format (signed 16bit, little endian) */
err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE);
ALSA_ASSERT("snd_pcm_hw_params_set_format");
/* set number of channels */
err = snd_pcm_hw_params_set_channels(pcm, hwparams, 2);
ALSA_ASSERT("snd_pcm_hw_params_set_channels");
unsigned int rate = samplerate;
err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0);
// check if we got the rate we desire
/* write the hardware parameters to the device */
err = snd_pcm_hw_params(pcm, hwparams);
ALSA_ASSERT("snd_pcm_hw_params");
snd_pcm_hw_params_free(hwparams);
/* prepare the device to reveice data */
err = snd_pcm_prepare(pcm);
ALSA_ASSERT("snd_pcm_prepare");
//XXX should RageException::ThrowNonfatal if something went wrong
/* prepare a snd_output_t for use with LOG->Trace */
errout = NULL;
snd_output_buffer_open(&errout);
snd_pcm_dump(pcm, errout);
char *errstring;
snd_output_buffer_string(errout, &errstring);
LOG->Trace("%s", errstring);
snd_output_flush(errout);
gettimeofday(&startup_time, NULL);
/* add error checking before Jan 2038 */
MixerThreadPtr = SDL_CreateThread(MixerThread_start, this);
}
RageSound_ALSA9::~RageSound_ALSA9()
{
/* Signal the mixing thread to quit. */
shutdown = true;
LOG->Trace("Shutting down mixer thread ...");
SDL_WaitThread(MixerThreadPtr, NULL);
LOG->Trace("Mixer thread shut down.");
snd_pcm_close(pcm);
}
float RageSound_ALSA9::GetPlayLatency() const
{
int err;
snd_pcm_status_t *status;
snd_pcm_sframes_t delay;
snd_pcm_status_alloca(&status);
err = snd_pcm_status(pcm, status);
ALSA_ASSERT("snd_pcm_status");
delay = snd_pcm_status_get_delay(status);
/*XXX not sure if delay is the thing to use for
* calculating latency
*
* delay is the number of frames between where
* the next write will begin and what is being
* played now
*/
/*XXX I'm assuming GetPlayLatency() is supposed to return seconds */
return ((float)delay)/samplerate;
}
/*
* Copyright (c) 2002 by the person(s) listed below. All rights reserved.
*
* Glenn Maynard (RageSoundDriver_WaveOut)
*/
@@ -0,0 +1,75 @@
#ifndef RAGE_SOUND_ALSA9
#define RAGE_SOUND_ALSA9
/* RageSound_ALSA9 fakes playing sounds having GetPosition()
* return seconds since the constructor was called.
*
* Only tested in linux, but intended to work across the globe.
* ( uses time_t, sleep() and nanosleep() from <time.h> )
*
* The timing probably isn't accurate, but at least it is fairly
* steady. Someone with more knowledge of RageSound, feel free
* to play around with this (not that it's any sort of priority).
*/
#include "RageSound.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <alsa/asoundlib.h>
/* needed because ALSA returns struct timeval info */
#include <sys/time.h>
class RageSound_ALSA9: public RageSoundDriver
{
public:
struct sound {
RageSound *snd;
bool stopping;
int samples_buffered;
int flush_pos; /* state == STOPPING only */
sound() { snd = NULL; stopping=false; samples_buffered=0; }
};
/* List of currently playing sounds: */
vector<sound *> sounds;
bool shutdown;
struct timeval startup_time;
int last_cursor_pos;
snd_pcm_t *pcm;
/* this is messy since it avoids LOG->Trace, but for now the
* quickest way to get snd_pcm_dump() info */
snd_output_t *errout;
static int MixerThread_start(void *p);
void MixerThread();
SDL_Thread *MixerThreadPtr;
bool GetData();
void Recover(int r);
/* virtuals: */
void StartMixing(RageSound *snd);
void StopMixing(RageSound *snd);
int GetPosition(const RageSound *snd) const;
float GetPlayLatency() const;
void Update(float delta);
RageSound_ALSA9();
~RageSound_ALSA9();
};
#endif
/*
* Copyright (c) 2003 by the person(s) listed below. All rights reserved.
*
* Glenn Maynard
*
* 2003-02 Modified to fake playing sound Aaron VonderHaar
*
*/