Mercurial > sdl-ios-xcode
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(¶ms); 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); }