Initial revision
--HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%401
This commit is contained in:
Sam Lantinga
commit
2f110628a7
892 changed files with 170313 additions and 0 deletions
521
src/audio/alsa/SDL_alsa_audio.c
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521
src/audio/alsa/SDL_alsa_audio.c
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@ -0,0 +1,521 @@
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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public
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License along with this library; if not, write to the Free
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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Sam Lantinga
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slouken@devolution.com
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*/
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/* Allow access to a raw mixing buffer */
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <errno.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <signal.h>
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#include <sys/types.h>
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#include <sys/time.h>
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#include "SDL_audio.h"
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#include "SDL_error.h"
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#include "SDL_audiomem.h"
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#include "SDL_audio_c.h"
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#include "SDL_timer.h"
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#include "SDL_alsa_audio.h"
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/* The tag name used by ALSA audio */
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#define DRIVER_NAME "alsa"
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/* default card and device numbers as listed in dev/snd */
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static int card_no = 0;
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static int device_no = 0;
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/* default channel communication parameters */
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#define DEFAULT_CPARAMS_RATE 22050
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#define DEFAULT_CPARAMS_VOICES 1
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#define DEFAULT_CPARAMS_FRAG_SIZE 512
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#define DEFAULT_CPARAMS_FRAGS_MIN 1
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#define DEFAULT_CPARAMS_FRAGS_MAX -1
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/* Open the audio device for playback, and don't block if busy */
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#define OPEN_FLAGS (SND_PCM_OPEN_PLAYBACK|SND_PCM_OPEN_NONBLOCK)
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/* Audio driver functions */
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static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec);
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static void PCM_WaitAudio(_THIS);
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static void PCM_PlayAudio(_THIS);
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static Uint8 *PCM_GetAudioBuf(_THIS);
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static void PCM_CloseAudio(_THIS);
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/* PCM transfer channel parameters initialize function */
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static void init_pcm_cparams(snd_pcm_channel_params_t* cparams)
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{
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memset(cparams,0,sizeof(snd_pcm_channel_params_t));
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cparams->channel = SND_PCM_CHANNEL_PLAYBACK;
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cparams->mode = SND_PCM_MODE_BLOCK;
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cparams->start_mode = SND_PCM_START_DATA; //_FULL
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cparams->stop_mode = SND_PCM_STOP_STOP;
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cparams->format.format = SND_PCM_SFMT_S16_LE;
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cparams->format.interleave = 1;
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cparams->format.rate = DEFAULT_CPARAMS_RATE;
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cparams->format.voices = DEFAULT_CPARAMS_VOICES;
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cparams->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
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cparams->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
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cparams->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
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}
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/* Audio driver bootstrap functions */
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static int Audio_Available(void)
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/*
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See if we can open a nonblocking channel.
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Return value '1' means we can.
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Return value '0' means we cannot.
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*/
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{
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int available;
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int rval;
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snd_pcm_t *handle;
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snd_pcm_channel_params_t cparams;
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#ifdef DEBUG_AUDIO
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snd_pcm_channel_status_t cstatus;
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#endif
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available = 0;
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handle = NULL;
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init_pcm_cparams(&cparams);
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rval = snd_pcm_open(&handle, card_no, device_no, OPEN_FLAGS);
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if (rval >= 0)
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{
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rval = snd_pcm_plugin_params(handle, &cparams);
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#ifdef DEBUG_AUDIO
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snd_pcm_plugin_status(handle, &cstatus);
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printf("status after snd_pcm_plugin_params call = %d\n",cstatus.status);
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#endif
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if (rval >= 0)
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{
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available = 1;
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}
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else
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{
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SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
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}
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if ((rval = snd_pcm_close(handle)) < 0)
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{
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SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
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available = 0;
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}
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}
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else
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{
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SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
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}
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return(available);
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}
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static void Audio_DeleteDevice(SDL_AudioDevice *device)
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{
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free(device->hidden);
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free(device);
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}
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static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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{
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SDL_AudioDevice *this;
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/* Initialize all variables that we clean on shutdown */
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this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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if ( this ) {
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memset(this, 0, (sizeof *this));
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this->hidden = (struct SDL_PrivateAudioData *)
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malloc((sizeof *this->hidden));
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}
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if ( (this == NULL) || (this->hidden == NULL) ) {
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SDL_OutOfMemory();
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if ( this ) {
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free(this);
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}
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return(0);
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}
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memset(this->hidden, 0, (sizeof *this->hidden));
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audio_handle = NULL;
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/* Set the function pointers */
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this->OpenAudio = PCM_OpenAudio;
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this->WaitAudio = PCM_WaitAudio;
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this->PlayAudio = PCM_PlayAudio;
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this->GetAudioBuf = PCM_GetAudioBuf;
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this->CloseAudio = PCM_CloseAudio;
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this->free = Audio_DeleteDevice;
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return this;
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}
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AudioBootStrap ALSA_bootstrap = {
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DRIVER_NAME, "ALSA PCM audio",
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Audio_Available, Audio_CreateDevice
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};
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/* This function waits until it is possible to write a full sound buffer */
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static void PCM_WaitAudio(_THIS)
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{
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/* Check to see if the thread-parent process is still alive */
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{ static int cnt = 0;
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/* Note that this only works with thread implementations
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that use a different process id for each thread.
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*/
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if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
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if ( kill(parent, 0) < 0 ) {
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this->enabled = 0;
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}
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}
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}
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/* See if we need to use timed audio synchronization */
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if ( frame_ticks )
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{
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/* Use timer for general audio synchronization */
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Sint32 ticks;
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ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
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if ( ticks > 0 )
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{
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SDL_Delay(ticks);
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}
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}
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else
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{
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/* Use select() for audio synchronization */
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fd_set fdset;
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struct timeval timeout;
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FD_ZERO(&fdset);
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FD_SET(audio_fd, &fdset);
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timeout.tv_sec = 10;
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timeout.tv_usec = 0;
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Waiting for audio to get ready\n");
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#endif
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if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 )
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{
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const char *message =
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"Audio timeout - buggy audio driver? (disabled)";
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/* In general we should never print to the screen,
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but in this case we have no other way of letting
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the user know what happened.
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*/
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fprintf(stderr, "SDL: %s\n", message);
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this->enabled = 0;
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/* Don't try to close - may hang */
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audio_fd = -1;
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Done disabling audio\n");
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#endif
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}
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Ready!\n");
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#endif
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}
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}
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static snd_pcm_channel_status_t cstatus;
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static void PCM_PlayAudio(_THIS)
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{
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int written, rval;
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/* Write the audio data, checking for EAGAIN (buffer full) and underrun */
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do {
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written = snd_pcm_plugin_write(audio_handle, pcm_buf, pcm_len);
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "written = %d pcm_len = %d\n",written,pcm_len);
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#endif
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if (written != pcm_len)
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{
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if (errno == EAGAIN)
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{
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SDL_Delay(1); /* Let a little CPU time go by and try to write again */
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "errno == EAGAIN\n");
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#endif
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}
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else
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{
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if( (rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0 )
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{
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SDL_SetError("snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
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return;
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}
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if ( (cstatus.status == SND_PCM_STATUS_UNDERRUN)
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||(cstatus.status == SND_PCM_STATUS_READY) )
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{
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "buffer underrun\n");
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#endif
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if ( (rval = snd_pcm_plugin_prepare (audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0 )
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{
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SDL_SetError("snd_pcm_plugin_prepare failed: %s\n",snd_strerror(rval) );
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return;
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}
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/* if we reach here, try to write again */
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}
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}
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}
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} while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
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/* Set the next write frame */
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if ( frame_ticks ) {
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next_frame += frame_ticks;
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}
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/* If we couldn't write, assume fatal error for now */
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if ( written < 0 ) {
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this->enabled = 0;
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}
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return;
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}
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static Uint8 *PCM_GetAudioBuf(_THIS)
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{
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return(pcm_buf);
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}
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static void PCM_CloseAudio(_THIS)
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{
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int rval;
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if ( pcm_buf != NULL ) {
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free(pcm_buf);
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pcm_buf = NULL;
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}
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if ( audio_handle != NULL ) {
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if ((rval = snd_pcm_plugin_flush(audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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SDL_SetError("snd_pcm_plugin_flush failed: %s\n",snd_strerror(rval));
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return;
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}
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if ((rval = snd_pcm_close(audio_handle)) < 0)
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{
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SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
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return;
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}
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audio_handle = NULL;
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}
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}
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static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec)
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{
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int rval;
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snd_pcm_channel_params_t cparams;
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snd_pcm_channel_setup_t csetup;
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int format;
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Uint16 test_format;
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int twidth;
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/* initialize channel transfer parameters to default */
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init_pcm_cparams(&cparams);
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/* Reset the timer synchronization flag */
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frame_ticks = 0.0;
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/* Open the audio device */
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rval = snd_pcm_open(&audio_handle, card_no, device_no, OPEN_FLAGS);
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if ( rval < 0 ) {
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SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
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return(-1);
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}
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#ifdef PLUGIN_DISABLE_MMAP /* This is gone in newer versions of ALSA? */
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/* disable count status parameter */
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if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
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{
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SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
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return(-1);
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||||
}
|
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#endif
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|
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pcm_buf = NULL;
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/* Try for a closest match on audio format */
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format = 0;
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for ( test_format = SDL_FirstAudioFormat(spec->format);
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! format && test_format; )
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{
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#ifdef DEBUG_AUDIO
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fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
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#endif
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/* if match found set format to equivalent ALSA format */
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switch ( test_format ) {
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case AUDIO_U8:
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format = SND_PCM_SFMT_U8;
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cparams.buf.block.frag_size = spec->samples * spec->channels;
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break;
|
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case AUDIO_S8:
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format = SND_PCM_SFMT_S8;
|
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cparams.buf.block.frag_size = spec->samples * spec->channels;
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||||
break;
|
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case AUDIO_S16LSB:
|
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format = SND_PCM_SFMT_S16_LE;
|
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cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
|
||||
break;
|
||||
case AUDIO_S16MSB:
|
||||
format = SND_PCM_SFMT_S16_BE;
|
||||
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
|
||||
break;
|
||||
case AUDIO_U16LSB:
|
||||
format = SND_PCM_SFMT_U16_LE;
|
||||
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
|
||||
break;
|
||||
case AUDIO_U16MSB:
|
||||
format = SND_PCM_SFMT_U16_BE;
|
||||
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
if ( ! format ) {
|
||||
test_format = SDL_NextAudioFormat();
|
||||
}
|
||||
}
|
||||
if ( format == 0 ) {
|
||||
SDL_SetError("Couldn't find any hardware audio formats");
|
||||
return(-1);
|
||||
}
|
||||
spec->format = test_format;
|
||||
|
||||
/* Set the audio format */
|
||||
cparams.format.format = format;
|
||||
|
||||
/* Set mono or stereo audio (currently only two channels supported) */
|
||||
cparams.format.voices = spec->channels;
|
||||
|
||||
#ifdef DEBUG_AUDIO
|
||||
printf("intializing channels %d\n", cparams.format.voices);
|
||||
#endif
|
||||
|
||||
/* Set rate */
|
||||
cparams.format.rate = spec->freq ;
|
||||
|
||||
/* Setup the transfer parameters according to cparams */
|
||||
rval = snd_pcm_plugin_params(audio_handle, &cparams);
|
||||
if (rval < 0) {
|
||||
SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
|
||||
return(-1);
|
||||
}
|
||||
|
||||
/* Make sure channel is setup right one last time */
|
||||
memset( &csetup, 0, sizeof( csetup ) );
|
||||
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
|
||||
if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
|
||||
{
|
||||
SDL_SetError("Unable to setup playback channel\n" );
|
||||
return(-1);
|
||||
}
|
||||
|
||||
#ifdef DEBUG_AUDIO
|
||||
else
|
||||
{
|
||||
fprintf(stderr,"requested format: %d\n",cparams.format.format);
|
||||
fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
|
||||
fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
|
||||
|
||||
fprintf(stderr,"real format: %d\n", csetup.format.format );
|
||||
fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
|
||||
fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
|
||||
}
|
||||
#endif // DEBUG_AUDIO
|
||||
|
||||
/* Allocate memory to the audio buffer and initialize with silence
|
||||
(Note that buffer size must be a multiple of fragment size, so find closest multiple)
|
||||
*/
|
||||
|
||||
twidth = snd_pcm_format_width(format);
|
||||
if (twidth < 0) {
|
||||
printf("snd_pcm_format_width failed\n");
|
||||
twidth = 0;
|
||||
}
|
||||
#ifdef DEBUG_AUDIO
|
||||
printf("format is %d bits wide\n",twidth);
|
||||
#endif
|
||||
|
||||
pcm_len = csetup.buf.block.frag_size * (twidth/8) * csetup.format.voices ;
|
||||
|
||||
#ifdef DEBUG_AUDIO
|
||||
printf("pcm_len set to %d\n", pcm_len);
|
||||
#endif
|
||||
|
||||
if (pcm_len == 0)
|
||||
{
|
||||
pcm_len = csetup.buf.block.frag_size;
|
||||
}
|
||||
|
||||
pcm_buf = (Uint8*)malloc(pcm_len);
|
||||
if (pcm_buf == NULL) {
|
||||
SDL_SetError("pcm_buf malloc failed\n");
|
||||
return(-1);
|
||||
}
|
||||
memset(pcm_buf,spec->silence,pcm_len);
|
||||
|
||||
#ifdef DEBUG_AUDIO
|
||||
fprintf(stderr,"pcm_buf malloced and silenced.\n");
|
||||
#endif
|
||||
|
||||
/* get the file descriptor */
|
||||
if( (audio_fd = snd_pcm_file_descriptor(audio_handle, device_no)) < 0)
|
||||
{
|
||||
fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
|
||||
}
|
||||
|
||||
/* Trigger audio playback */
|
||||
rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
|
||||
if (rval < 0) {
|
||||
SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
|
||||
return(-1);
|
||||
}
|
||||
rval = snd_pcm_playback_go(audio_handle);
|
||||
if (rval < 0) {
|
||||
SDL_SetError("snd_pcm_playback_go failed: %s\n", snd_strerror (rval));
|
||||
return(-1);
|
||||
}
|
||||
|
||||
/* Check to see if we need to use select() workaround */
|
||||
{ char *workaround;
|
||||
workaround = getenv("SDL_DSP_NOSELECT");
|
||||
if ( workaround ) {
|
||||
frame_ticks = (float)(spec->samples*1000)/spec->freq;
|
||||
next_frame = SDL_GetTicks()+frame_ticks;
|
||||
}
|
||||
}
|
||||
|
||||
/* Get the parent process id (we're the parent of the audio thread) */
|
||||
parent = getpid();
|
||||
|
||||
/* We're ready to rock and roll. :-) */
|
||||
return(0);
|
||||
}
|
||||
Loading…
Add table
Add a link
Reference in a new issue