comparison libao2/ao_sun.c @ 29263:0f1b5b68af32

whitespace cosmetics: Remove all trailing whitespace.

29262:7d545a6b8aff

#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"

static const ao_info_t info = 
{
    "Sun audio output",
    "sun",
    "Juergen Keil",
    ""

			     * "write" below.
			     */
    silence = calloc(1, len);
    if (silence == NULL)
	goto error;
    
    if ((fd = open(dev, O_WRONLY)) < 0)
	goto error;

    AUDIO_INITINFO(&info);
    info.play.sample_rate = 44100;

    if (ioctl(fd, AUDIO_SETINFO, &info)) {
	if ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
	goto error;
    }
    
    if (write(fd, silence, len) != len) {
	if ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
	goto error;
    }

     * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
     * chunks (== 4096 samples) to the audio device.  If we see a minimum
     * sample counter increment from the soundcard driver of less than
     * 2000 samples,  we assume that the driver provides a useable realtime
     * sample counter in the AUDIO_INFO play.samples field.  Timing based
     * on sample counts should be much more accurate than counting whole 
     * 16kbyte chunks.
     */
    if (min_increment < 2000)
	rtsc_ok = RTSC_ENABLED;

    if ( mp_msg_test(MSGT_AO,MSGL_V) )
	mp_msg(MSGT_AO,MSGL_V,"ao_sun: minimum sample counter increment per 10msec interval: %d\n"
	       "\t%susing sample counter based timing code\n",
	       min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
    

error:
    if (silence != NULL) free(silence);
    if (fd >= 0) {
	// remove the 0 bytes from the above measurement from the

	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between 
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	free(sr);
	return 0;
    } else {

    };

    for (i = 0; audiocs_rates[i]; i++) {
	err = abs(audiocs_rates[i] - sample_rate);
	if (err == 0) {
	    /* 
	     * exact supported sample rate match, no need to
	     * retry something elise
	     */
	    return 0;
	}

	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between 
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	max_rate = sr->samp_rates[1];
    } else {
	/* HW supports fixed sample rates only */

							/ AUDIO_MID_BALANCE;
		vol->right = volume;
	    }
	    close( fd );
	    return CONTROL_OK;
	}	
	return CONTROL_ERROR;
    }
    case AOCONTROL_SET_VOLUME:
    {
	ao_control_vol_t *vol = (ao_control_vol_t *)arg;

	    info.output_muted = (volume == 0);
#endif
	    ioctl( fd,AUDIO_SETINFO,&info );
	    close( fd );
	    return CONTROL_OK;
	}	
	return CONTROL_ERROR;
    }
    }
    return CONTROL_UNKNOWN;
}

	convert_u8_s8 = 0;

	if (pass & 1) {
	    /*
	     * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is 
	     * not supported, but 8-bit signed encoding is.
	     *
	     * Try S8, and if it works, use our own U8->S8 conversion before
	     * sending the samples to the sound driver.
	     */

	     *
	     * In case the requested sample rate is very close to one of the
	     * supported rates,  use the fixed supported rate instead.
	     */
	    if (!(info.play.sample_rate =
		  find_close_samplerate_match(audio_fd, rate))) 
	      continue;

	    /*
	     * I'm not returning the correct sample rate in
	     * |ao_data.samplerate|, to avoid software resampling.

29263:0f1b5b68af32

#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"

static const ao_info_t info =
{
    "Sun audio output",
    "sun",
    "Juergen Keil",
    ""

			     * "write" below.
			     */
    silence = calloc(1, len);
    if (silence == NULL)
	goto error;

    if ((fd = open(dev, O_WRONLY)) < 0)
	goto error;

    AUDIO_INITINFO(&info);
    info.play.sample_rate = 44100;

    if (ioctl(fd, AUDIO_SETINFO, &info)) {
	if ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
	goto error;
    }

    if (write(fd, silence, len) != len) {
	if ( mp_msg_test(MSGT_AO,MSGL_V) )
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
	goto error;
    }

     * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
     * chunks (== 4096 samples) to the audio device.  If we see a minimum
     * sample counter increment from the soundcard driver of less than
     * 2000 samples,  we assume that the driver provides a useable realtime
     * sample counter in the AUDIO_INFO play.samples field.  Timing based
     * on sample counts should be much more accurate than counting whole
     * 16kbyte chunks.
     */
    if (min_increment < 2000)
	rtsc_ok = RTSC_ENABLED;

    if ( mp_msg_test(MSGT_AO,MSGL_V) )
	mp_msg(MSGT_AO,MSGL_V,"ao_sun: minimum sample counter increment per 10msec interval: %d\n"
	       "\t%susing sample counter based timing code\n",
	       min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");


error:
    if (silence != NULL) free(silence);
    if (fd >= 0) {
	// remove the 0 bytes from the above measurement from the

	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	free(sr);
	return 0;
    } else {

    };

    for (i = 0; audiocs_rates[i]; i++) {
	err = abs(audiocs_rates[i] - sample_rate);
	if (err == 0) {
	    /*
	     * exact supported sample rate match, no need to
	     * retry something elise
	     */
	    return 0;
	}

	free(sr);
    }

    if (sr->flags & MIXER_SR_LIMITS) {
	/*
	 * HW can playback any rate between
	 * sr->samp_rates[0] .. sr->samp_rates[1]
	 */
	max_rate = sr->samp_rates[1];
    } else {
	/* HW supports fixed sample rates only */

							/ AUDIO_MID_BALANCE;
		vol->right = volume;
	    }
	    close( fd );
	    return CONTROL_OK;
	}
	return CONTROL_ERROR;
    }
    case AOCONTROL_SET_VOLUME:
    {
	ao_control_vol_t *vol = (ao_control_vol_t *)arg;

	    info.output_muted = (volume == 0);
#endif
	    ioctl( fd,AUDIO_SETINFO,&info );
	    close( fd );
	    return CONTROL_OK;
	}
	return CONTROL_ERROR;
    }
    }
    return CONTROL_UNKNOWN;
}

	convert_u8_s8 = 0;

	if (pass & 1) {
	    /*
	     * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is
	     * not supported, but 8-bit signed encoding is.
	     *
	     * Try S8, and if it works, use our own U8->S8 conversion before
	     * sending the samples to the sound driver.
	     */

	     *
	     * In case the requested sample rate is very close to one of the
	     * supported rates,  use the fixed supported rate instead.
	     */
	    if (!(info.play.sample_rate =
		  find_close_samplerate_match(audio_fd, rate)))
	      continue;

	    /*
	     * I'm not returning the correct sample rate in
	     * |ao_data.samplerate|, to avoid software resampling.