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335 lines
9.3 KiB
335 lines
9.3 KiB
15 years ago
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/*
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* Driver for Advanced Linux Sound Architecture, http://alsa.jcu.cz
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*
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* mpg123 comments:
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* Code by Anders Semb Hermansen <ahermans@vf.telia.no>
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* Cleanups by Jaroslav Kysela <perex@jcu.cz>
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* Ville Syrjala <syrjala@sci.fi>
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*
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* adopted for libworkman cdda audio backend from Alexander Kern alex.kern@gmx.de
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*
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* Adapted to support both ALSA V0.x and V1.x APIs for PCM calls
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* (Philip Nelson <teamdba@scotdb.com> 2004-03-15)
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*
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* This file comes under GPL license.
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*/
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#include <config.h>
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#if defined(HAVE_ARTS_LIBASOUND2)
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#include <alsa/asoundlib.h>
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#include "audio.h"
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char* device = NULL;
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snd_pcm_t *handle;
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snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */
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int rate = 44100; /* stream rate */
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int channels = 2; /* count of channels */
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int buffer_time = 2000000; /* ring buffer length in us */
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int period_time = 100000; /* period time in us */
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snd_pcm_sframes_t buffer_size;
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snd_pcm_sframes_t period_size;
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int alsa_open(void);
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int alsa_close(void);
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int alsa_stop(void);
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int alsa_play(struct cdda_block *blk);
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int alsa_state(struct cdda_block *blk);
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struct audio_oops* setup_alsa(const char *dev, const char *ctl);
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static int set_hwparams(snd_pcm_hw_params_t *params,
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snd_pcm_access_t accesspar)
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{
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int err, dir, new_rate;
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/* choose all parameters */
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err = snd_pcm_hw_params_any(handle, params);
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if (err < 0) {
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ERRORLOG("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
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return err;
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}
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/* set the interleaved read/write format */
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err = snd_pcm_hw_params_set_access(handle, params, accesspar);
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if (err < 0) {
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ERRORLOG("Access type not available for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* set the sample format */
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err = snd_pcm_hw_params_set_format(handle, params, format);
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if (err < 0) {
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ERRORLOG("Sample format not available for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* set the count of channels */
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err = snd_pcm_hw_params_set_channels(handle, params, channels);
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if (err < 0) {
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ERRORLOG("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err));
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return err;
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}
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/* set the stream rate */
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#if (SND_LIB_MAJOR < 1)
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err = new_rate = snd_pcm_hw_params_set_rate_near(handle, params, rate, 0);
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#else
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new_rate = rate;
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err = snd_pcm_hw_params_set_rate_near(handle, params, &new_rate, 0);
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#endif
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if (err < 0) {
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ERRORLOG("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
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return err;
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}
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if (new_rate != rate) {
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ERRORLOG("Rate doesn't match (requested %iHz, get %iHz)\n", rate, new_rate);
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return -EINVAL;
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}
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/* set the buffer time */
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#if (SND_LIB_MAJOR < 1)
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err = snd_pcm_hw_params_set_buffer_time_near(handle, params, buffer_time, &dir);
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#else
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err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir);
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#endif
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if (err < 0) {
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ERRORLOG("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err));
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return err;
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}
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#if (SND_LIB_MAJOR < 1)
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buffer_size = snd_pcm_hw_params_get_buffer_size(params);
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#else
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err = snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
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if (err < 0) {
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ERRORLOG("Unable to get buffer size : %s\n", snd_strerror(err));
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return err;
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}
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#endif
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DEBUGLOG("buffersize %i\n", buffer_size);
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/* set the period time */
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#if (SND_LIB_MAJOR < 1)
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err = snd_pcm_hw_params_set_period_time_near(handle, params, period_time, &dir);
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#else
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err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir);
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#endif
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if (err < 0) {
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ERRORLOG("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err));
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return err;
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}
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#if (SND_LIB_MAJOR < 1)
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period_size = snd_pcm_hw_params_get_period_size(params, &dir);
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#else
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err = snd_pcm_hw_params_get_period_size(params, &period_size, &dir);
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if (err < 0) {
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ERRORLOG("Unable to get hw period size: %s\n", snd_strerror(err));
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}
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#endif
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DEBUGLOG("period_size %i\n", period_size);
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/* write the parameters to device */
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err = snd_pcm_hw_params(handle, params);
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if (err < 0) {
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ERRORLOG("Unable to set hw params for playback: %s\n", snd_strerror(err));
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return err;
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}
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return 0;
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}
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static int set_swparams(snd_pcm_sw_params_t *swparams)
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{
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int err;
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/* get the current swparams */
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err = snd_pcm_sw_params_current(handle, swparams);
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if (err < 0) {
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ERRORLOG("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* start the transfer when the buffer is full */
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err = snd_pcm_sw_params_set_start_threshold(handle, swparams, buffer_size);
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if (err < 0) {
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ERRORLOG("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* allow the transfer when at least period_size samples can be processed */
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err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
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if (err < 0) {
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ERRORLOG("Unable to set avail min for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* align all transfers to 1 sample */
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err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
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if (err < 0) {
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ERRORLOG("Unable to set transfer align for playback: %s\n", snd_strerror(err));
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return err;
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}
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/* write the parameters to the playback device */
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err = snd_pcm_sw_params(handle, swparams);
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if (err < 0) {
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ERRORLOG("Unable to set sw params for playback: %s\n", snd_strerror(err));
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return err;
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}
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return 0;
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}
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int alsa_open( void )
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{
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int err;
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_sw_params_t *swparams;
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DEBUGLOG("alsa_open\n");
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snd_pcm_hw_params_alloca(&hwparams);
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snd_pcm_sw_params_alloca(&swparams);
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if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0/*SND_PCM_NONBLOCK*/)) < 0 ) {
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ERRORLOG("open failed: %s\n", snd_strerror(err));
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return -1;
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}
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if((err = set_hwparams(hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
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ERRORLOG("Setting of hwparams failed: %s\n", snd_strerror(err));
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return -1;
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}
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if((err = set_swparams(swparams)) < 0) {
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ERRORLOG("Setting of swparams failed: %s\n", snd_strerror(err));
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return -1;
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}
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return 0;
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}
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int alsa_close( void )
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{
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int err;
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DEBUGLOG("alsa_close\n");
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err = alsa_stop();
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#if (SND_LIB_MAJOR < 1)
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err = snd_pcm_close(handle);
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#else
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err = snd_pcm_close(handle);
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#endif
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free(device);
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return err;
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}
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/*
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* Play some audio and pass a status message upstream, if applicable.
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* Returns 0 on success.
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*/
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int
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alsa_play(struct cdda_block *blk)
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{
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signed short *ptr;
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int err = 0, frames;
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ptr = (signed short *)blk->buf;
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frames = blk->buflen / (channels * 2);
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DEBUGLOG("play %i frames, %i bytes\n", frames, blk->buflen);
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while (frames > 0) {
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err = snd_pcm_writei(handle, ptr, frames);
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if (err == -EAGAIN)
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continue;
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if(err == -EPIPE) {
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err = snd_pcm_prepare(handle);
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continue;
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} else if (err < 0)
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break;
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ptr += err * channels;
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frames -= err;
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DEBUGLOG("played %i, rest %i\n", err / channels, frames);
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}
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if (err < 0) {
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ERRORLOG("alsa_write failed: %s\n", snd_strerror(err));
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err = snd_pcm_prepare(handle);
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if (err < 0) {
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ERRORLOG("Unable to snd_pcm_prepare pcm stream: %s\n", snd_strerror(err));
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}
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blk->status = WM_CDM_CDDAERROR;
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return err;
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}
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return 0;
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}
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/*
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* Stop the audio immediately.
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*/
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int
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alsa_stop( void )
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{
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int err;
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DEBUGLOG("alsa_stop\n");
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err = snd_pcm_drop(handle);
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if (err < 0) {
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ERRORLOG("Unable to drop pcm stream: %s\n", snd_strerror(err));
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}
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err = snd_pcm_prepare(handle);
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if (err < 0) {
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ERRORLOG("Unable to snd_pcm_prepare pcm stream: %s\n", snd_strerror(err));
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}
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return err;
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}
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/*
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* Get the current audio state.
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*/
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int
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alsa_state(struct cdda_block *blk)
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{
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return -1; /* not implemented yet for ALSA */
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}
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static struct audio_oops alsa_oops = {
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.wmaudio_open = alsa_open,
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.wmaudio_close = alsa_close,
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.wmaudio_play = alsa_play,
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.wmaudio_stop = alsa_stop,
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.wmaudio_state = alsa_state,
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.wmaudio_balance = NULL,
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.wmaudio_volume = NULL
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};
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struct audio_oops*
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setup_alsa(const char *dev, const char *ctl)
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{
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static int init_complete = 0;
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if(dev && strlen(dev) > 0) {
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device = strdup(dev);
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} else {
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device = strdup("plughw:0,0"); /* playback device */
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}
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if(init_complete) {
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ERRORLOG("already initialized\n");
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return NULL;
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}
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if(!alsa_open())
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init_complete = 1;
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else
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return NULL;
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return &alsa_oops;
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}
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#endif /* ALSA */
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