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lxdream.org :: lxdream/src/drivers/audio_alsa.c
lxdream 0.9.1
released Jun 29
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filename src/drivers/audio_alsa.c
changeset 669:ab344e42bca9
prev643:653b0a70f173
next697:479b8c213f61
author nkeynes
date Mon May 12 10:00:13 2008 +0000 (13 years ago)
permissions -rwxr-xr-x
last change Cleanup most of the -Wall warnings (getting a bit sloppy...)
Convert FP code to use fixed banks rather than indirect pointer
(3-4% faster this way now)
file annotate diff log raw
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/**
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 * $Id: audio_esd.c 602 2008-01-15 20:50:23Z nkeynes $
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 * 
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 * The asla  audio driver
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 *
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 * Copyright (c) 2008 Jonathan Muller
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 *
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 * This program is free software; you can redistribute it and/or modify
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 * it under the terms of the GNU General Public License as published by
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 * the Free Software Foundation; either version 2 of the License, or
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 * (at your option) any later version.
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 *
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 * This program 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
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 * GNU General Public License for more details.
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 */
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#include <stdio.h>
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#include <unistd.h>
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/* Use the newer ALSA API */
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#include <alsa/asoundlib.h>
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#include "config.h"
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#include "aica/audio.h"
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#include "dream.h"
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static snd_pcm_t *_soundDevice = NULL;
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static int frame_bytes;
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struct lxdream_config_entry alsa_config[] = {
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    {"device", CONFIG_TYPE_FILE, "default"},
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    {NULL, CONFIG_TYPE_NONE}
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};
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gboolean audio_alsa_init(  )
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{
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    return TRUE;
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}
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gboolean audio_alsa_set_format( uint32_t rate, uint32_t format )
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{
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    int err;
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    snd_pcm_hw_params_t *hw_params;
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    snd_pcm_sw_params_t *sw_params;
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    snd_pcm_uframes_t frames;
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    snd_pcm_uframes_t bufferSize;
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    int dir;
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    // Open the device we were told to open.
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    err = snd_pcm_open( &_soundDevice, alsa_config[0].value,
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                        SND_PCM_STREAM_PLAYBACK, 0 );
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    // Check for error on open.
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    if ( err < 0 ) {
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        ERROR( "Init: cannot open audio device %s (%s)\n",
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               alsa_config[0].value, snd_strerror( err ) );
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        return FALSE;
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    } else {
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        DEBUG( "Audio device opened successfully." );
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    }
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    frame_bytes = ( 2 * ( snd_pcm_format_width( SND_PCM_FORMAT_S16_LE ) / 8 ) );
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    //snd_pcm_hw_params_alloca (&hw_params);
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    // Allocate the hardware parameter structure.
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    if ( ( err = snd_pcm_hw_params_malloc( &hw_params ) ) < 0 ) {
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        ERROR( "Init: cannot allocate hardware parameter structure (%s)\n",
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               snd_strerror( err ) );
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        return FALSE;
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    }
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    if ( ( err = snd_pcm_hw_params_any( _soundDevice, hw_params ) ) < 0 ) {
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        ERROR( "Init: cannot allocate hardware parameter structure (%s)\n",
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               snd_strerror( err ) );
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        return FALSE;
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    }
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    // Set access to RW interleaved.
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    if ( ( err = snd_pcm_hw_params_set_access( _soundDevice, hw_params,
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                                               SND_PCM_ACCESS_RW_INTERLEAVED ) )
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         < 0 ) {
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        ERROR( " Init: cannot set access type (%s)\n", snd_strerror( err ) );
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        return FALSE;
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    }
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    if ( ( err = snd_pcm_hw_params_set_format( _soundDevice, hw_params,
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                                               SND_PCM_FORMAT_S16_LE ) ) <
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         0 ) {
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        ERROR( "Init: cannot set sample format (%s)\n", snd_strerror( err ) );
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        return FALSE;
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    }
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    err = snd_pcm_hw_params_set_rate_near( _soundDevice, hw_params, &rate, 0 );
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    if ( err < 0 ) {
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        ERROR( "Init: Resampling setup failed for playback: %s\n",
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               snd_strerror( err ) );
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        return err;
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    }
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    // Set channels to stereo (2).
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    err = snd_pcm_hw_params_set_channels( _soundDevice, hw_params, 2 );
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    if ( err < 0 ) {
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        ERROR( "Init: cannot set channel count (%s)\n", snd_strerror( err ) );
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        return FALSE;
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    }
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    // frames = 4410;
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    // snd_pcm_hw_params_set_period_size_near( _soundDevice, hw_params, &frames,
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    //                                     &dir );
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    // Apply the hardware parameters that we've set.
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    err = snd_pcm_hw_params( _soundDevice, hw_params );
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    if ( err < 0 ) {
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        DEBUG( "Init: cannot set parameters (%s)\n", snd_strerror( err ) );
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        return FALSE;
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    } else {
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        DEBUG( "Audio device parameters have been set successfully." );
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    }
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    snd_pcm_hw_params_get_period_size( hw_params, &frames, &dir );
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    DEBUG( "period size = %d\n", frames );
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    // Get the buffer size.
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    snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );
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    DEBUG("Buffer Size = %d\n", bufferSize);
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    // If we were going to do more with our sound device we would want to store
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    // the buffer size so we know how much data we will need to fill it with.
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    //cout << "Init: Buffer size = " << bufferSize << " frames." << endl;
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    // Display the bit size of samples.
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    //cout << "Init: Significant bits for linear samples = " << snd_pcm_hw_params_get_sbits(hw_params) << endl;
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    // Free the hardware parameters now that we're done with them.
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    snd_pcm_hw_params_free( hw_params );
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    // Set the start threshold to reduce inter-buffer gaps
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    snd_pcm_sw_params_alloca( &sw_params );
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    snd_pcm_sw_params_current( _soundDevice, sw_params );
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    snd_pcm_sw_params_set_start_threshold( _soundDevice, sw_params, bufferSize/2 );
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    err = snd_pcm_sw_params( _soundDevice, sw_params );
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    if( err < 0 ) {
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	ERROR("Unable to set sw params for alsa driver: %s\n", snd_strerror(err));
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	return FALSE;
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    } 
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    err = snd_pcm_prepare( _soundDevice );
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    if ( err < 0 ) {
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        ERROR( "Init: cannot prepare audio interface for use (%s)\n",
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               snd_strerror( err ) );
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        return FALSE;
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    }
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    return TRUE;
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}
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gboolean audio_alsa_process_buffer( audio_buffer_t buffer )
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{
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    int err;
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    int length;
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    length = buffer->length / frame_bytes;
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    err = snd_pcm_writei( _soundDevice, buffer->data, length );
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    if( err == -EPIPE ) {
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	snd_pcm_prepare( _soundDevice );
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    } else if( err == -ESTRPIPE ) {
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	snd_pcm_resume( _soundDevice );
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    }
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    return TRUE;
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}
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gboolean audio_alsa_close(  )
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{
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    return TRUE;
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}
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struct audio_driver audio_alsa_driver = { 
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    "alsa",
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    audio_alsa_init,
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    audio_alsa_set_format,
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    audio_alsa_process_buffer,
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    audio_alsa_close
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};
.