view src/audio/alsa/SDL_alsa_audio.c @ 773:da0a2ad35bf4

Date: Sun, 4 Jan 2004 23:48:19 +0100 From: Max Horn Subject: Re: Again Audio CD patch Am 04.01.2004 um 22:38 schrieb Sam Lantinga: > > Okay, I fixed the buffering problems by simply using a 4 second buffer > instead of a 1 second buffer. However, using your code I can't play an > entire CD - the playback stops after the first song. > Found the problem: FSReadFork returns eofErr when the file is finished. However, we check its return value for errors, and if anything but noErr occurs, the reader thread aborts its current iteration. That is bad, because it aborts before it can ever set the flag which tells that the file is over (also, any remaining data which FSRead did return is lost - so you'd not hear about to 4 seconds from the end of the file. Furthermore, the computed data size was 8 bytes to high (I forgot to account for the fact that the size of an (A)IFF chunk always contains the chunk header & size fields, too). This is enough to make it work. However, the end condition is rather fragile, so I tuned some other things to be pessimistic (check for <= 0 instead of == 0, when eofErr is encountered enforce mReadFilePosition == mFileLength). You never know... The attached patch fixes the issue for me.
author Sam Lantinga <slouken@libsdl.org>
date Mon, 05 Jan 2004 00:57:51 +0000
parents b8d311d90021
children 92615154bb68
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/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2004 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Library General Public
    License as published by the Free Software Foundation; either
    version 2 of the License, or (at your option) any later version.

    This library 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
    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public
    License along with this library; if not, write to the Free
    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

    Sam Lantinga
    slouken@libsdl.org
*/



/* Allow access to a raw mixing buffer */

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/types.h>
#include <sys/time.h>

#include "SDL_audio.h"
#include "SDL_error.h"
#include "SDL_audiomem.h"
#include "SDL_audio_c.h"
#include "SDL_timer.h"
#include "SDL_alsa_audio.h"

/* The tag name used by ALSA audio */
#define DRIVER_NAME         "alsa"

/* The default ALSA audio driver */
#define DEFAULT_DEVICE	"default"

/* Audio driver functions */
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
static void ALSA_WaitAudio(_THIS);
static void ALSA_PlayAudio(_THIS);
static Uint8 *ALSA_GetAudioBuf(_THIS);
static void ALSA_CloseAudio(_THIS);

static const char *get_audio_device()
{
	const char *device;
	
	device = getenv("AUDIODEV");	/* Is there a standard variable name? */
	if ( device == NULL ) {
		device = DEFAULT_DEVICE;
	}
	return device;
}

/* Audio driver bootstrap functions */

static int Audio_Available(void)
{
	int available;
	int status;
	snd_pcm_t *handle;

	available = 0;
	status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
	if ( status >= 0 ) {
		available = 1;
        	snd_pcm_close(handle);
	}
	return(available);
}

static void Audio_DeleteDevice(SDL_AudioDevice *device)
{
	free(device->hidden);
	free(device);
}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
	SDL_AudioDevice *this;

	/* Initialize all variables that we clean on shutdown */
	this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
	if ( this ) {
		memset(this, 0, (sizeof *this));
		this->hidden = (struct SDL_PrivateAudioData *)
				malloc((sizeof *this->hidden));
	}
	if ( (this == NULL) || (this->hidden == NULL) ) {
		SDL_OutOfMemory();
		if ( this ) {
			free(this);
		}
		return(0);
	}
	memset(this->hidden, 0, (sizeof *this->hidden));

	/* Set the function pointers */
	this->OpenAudio = ALSA_OpenAudio;
	this->WaitAudio = ALSA_WaitAudio;
	this->PlayAudio = ALSA_PlayAudio;
	this->GetAudioBuf = ALSA_GetAudioBuf;
	this->CloseAudio = ALSA_CloseAudio;

	this->free = Audio_DeleteDevice;

	return this;
}

AudioBootStrap ALSA_bootstrap = {
	DRIVER_NAME, "ALSA 0.9 PCM audio",
	Audio_Available, Audio_CreateDevice
};

/* This function waits until it is possible to write a full sound buffer */
static void ALSA_WaitAudio(_THIS)
{
	/* Check to see if the thread-parent process is still alive */
	{ static int cnt = 0;
		/* Note that this only works with thread implementations 
		   that use a different process id for each thread.
		*/
		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
			if ( kill(parent, 0) < 0 ) {
				this->enabled = 0;
			}
		}
	}
}

static void ALSA_PlayAudio(_THIS)
{
	int           status;
	int           sample_len;
	signed short *sample_buf;

	sample_len = this->spec.samples;
	sample_buf = (signed short *)mixbuf;
	while ( sample_len > 0 ) {
		status = snd_pcm_writei(pcm_handle, sample_buf, sample_len);
		if ( status < 0 ) {
			if ( status == -EAGAIN ) {
				SDL_Delay(1);
				continue;
			}
			if ( status == -ESTRPIPE ) {
				do {
					SDL_Delay(1);
					status = snd_pcm_resume(pcm_handle);
				} while ( status == -EAGAIN );
			}
			if ( status < 0 ) {
				status = snd_pcm_prepare(pcm_handle);
			}
			if ( status < 0 ) {
				/* Hmm, not much we can do - abort */
				this->enabled = 0;
				return;
			}
			continue;
		}
		sample_buf += status * this->spec.channels;
		sample_len -= status;
	}
}

static Uint8 *ALSA_GetAudioBuf(_THIS)
{
	return(mixbuf);
}

static void ALSA_CloseAudio(_THIS)
{
	if ( mixbuf != NULL ) {
		SDL_FreeAudioMem(mixbuf);
		mixbuf = NULL;
	}
	if ( pcm_handle ) {
		snd_pcm_drain(pcm_handle);
		snd_pcm_close(pcm_handle);
		pcm_handle = NULL;
	}
}

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	int                  status;
	snd_pcm_hw_params_t *params;
	snd_pcm_format_t     format;
	snd_pcm_uframes_t    frames;
	Uint16               test_format;

	/* Open the audio device */
	status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
	if ( status < 0 ) {
		SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));
		return(-1);
	}

	/* Figure out what the hardware is capable of */
	snd_pcm_hw_params_alloca(&params);
	status = snd_pcm_hw_params_any(pcm_handle, params);
	if ( status < 0 ) {
		SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* SDL only uses interleaved sample output */
	status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Try for a closest match on audio format */
	status = -1;
	for ( test_format = SDL_FirstAudioFormat(spec->format);
	      test_format && (status < 0); ) {
		switch ( test_format ) {
			case AUDIO_U8:
				format = SND_PCM_FORMAT_U8;
				break;
			case AUDIO_S8:
				format = SND_PCM_FORMAT_S8;
				break;
			case AUDIO_S16LSB:
				format = SND_PCM_FORMAT_S16_LE;
				break;
			case AUDIO_S16MSB:
				format = SND_PCM_FORMAT_S16_BE;
				break;
			case AUDIO_U16LSB:
				format = SND_PCM_FORMAT_U16_LE;
				break;
			case AUDIO_U16MSB:
				format = SND_PCM_FORMAT_U16_BE;
				break;
			default:
				format = 0;
				break;
		}
		if ( format != 0 ) {
			status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
		}
		if ( status < 0 ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( status < 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->format = test_format;

	/* Set the number of channels */
	status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
	if ( status < 0 ) {
		status = snd_pcm_hw_params_get_channels(params);
		if ( (status <= 0) || (status > 2) ) {
			SDL_SetError("Couldn't set audio channels");
			ALSA_CloseAudio(this);
			return(-1);
		}
		spec->channels = status;
	}

	/* Set the audio rate */
	status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->freq = status;

	/* Set the buffer size, in samples */
	frames = spec->samples;
	frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
	spec->samples = frames;
	snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);

	/* "set" the hardware with the desired parameters */
	status = snd_pcm_hw_params(pcm_handle, params);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Calculate the final parameters for this audio specification */
	SDL_CalculateAudioSpec(spec);

	/* Allocate mixing buffer */
	mixlen = spec->size;
	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
	if ( mixbuf == NULL ) {
		ALSA_CloseAudio(this);
		return(-1);
	}
	memset(mixbuf, spec->silence, spec->size);

	/* Get the parent process id (we're the parent of the audio thread) */
	parent = getpid();

	/* Switch to blocking mode for playback */
	snd_pcm_nonblock(pcm_handle, 0);

	/* We're ready to rock and roll. :-) */
	return(0);
}