view libmpdemux/ai_alsa.c @ 7974:db1f16543379

enable volume filter and fix nonsense default volume (still not usable because mixer.c has no mechanism to pass volume commands to libaf)
author rfelker
date Wed, 30 Oct 2002 04:11:26 +0000
parents d12421dd1265
children 31f12f99118b
line wrap: on
line source

#include <stdio.h>
#include <stdlib.h>
#include <sys/time.h>

#include "config.h"

#if defined(USE_TV) && defined(HAVE_TV_V4L) && defined(HAVE_ALSA9)

#include <alsa/asoundlib.h>
#include "audio_in.h"
#include "mp_msg.h"

int ai_alsa_setup(audio_in_t *ai)
{
    snd_pcm_hw_params_t *params;
    snd_pcm_sw_params_t *swparams;
    int buffer_size;
    int err;
    unsigned int rate;

    snd_pcm_hw_params_alloca(&params);
    snd_pcm_sw_params_alloca(&swparams);

    err = snd_pcm_hw_params_any(ai->alsa.handle, params);
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Broken configuration for this PCM: no configurations available\n");
	return -1;
    }
    err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
				       SND_PCM_ACCESS_RW_INTERLEAVED);
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Access type not available\n");
	return -1;
    }
    err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Sample format not available\n");
	return -1;
    }
    err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
    if (err < 0) {
	ai->channels = snd_pcm_hw_params_get_channels(params);
	mp_msg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n",
	       ai->channels);
    } else {
	ai->channels = ai->req_channels;
    }

    err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0);
    assert(err >= 0);
    rate = err;
    ai->samplerate = rate;

    ai->alsa.buffer_time = 1000000;
    ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
							       ai->alsa.buffer_time, 0);
    assert(ai->alsa.buffer_time >= 0);
    ai->alsa.period_time = ai->alsa.buffer_time / 4;
    ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
							       ai->alsa.period_time, 0);
    assert(ai->alsa.period_time >= 0);
    err = snd_pcm_hw_params(ai->alsa.handle, params);
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Unable to install hw params:");
	snd_pcm_hw_params_dump(params, ai->alsa.log);
	return -1;
    }
    ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0);
    buffer_size = snd_pcm_hw_params_get_buffer_size(params);
    if (ai->alsa.chunk_size == buffer_size) {
	mp_msg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size);
	return -1;
    }
    snd_pcm_sw_params_current(ai->alsa.handle, swparams);
    err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0);
    assert(err >= 0);
    err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
    assert(err >= 0);

    err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
    assert(err >= 0);
    err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
    assert(err >= 0);

    assert(err >= 0);
    if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "unable to install sw params:\n");
	snd_pcm_sw_params_dump(swparams, ai->alsa.log);
	return -1;
    }

    if (mp_msg_test(MSGT_TV, MSGL_V)) {
	snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
    }

    ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
    ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
    ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
    ai->samplesize = ai->alsa.bits_per_sample;
    ai->bytes_per_sample = ai->alsa.bits_per_sample/8;

    return 0;
}

int ai_alsa_init(audio_in_t *ai)
{
    int err;
    
    err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0);
    if (err < 0) {
	mp_msg(MSGT_TV, MSGL_ERR, "Error opening audio: %s\n", snd_strerror(err));
	return -1;
    }
    
    err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0);
    
    if (err < 0) {
	return -1;
    }
    
    err = ai_alsa_setup(ai);

    return err;
}

#ifndef timersub
#define	timersub(a, b, result) \
do { \
	(result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
	(result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
	if ((result)->tv_usec < 0) { \
		--(result)->tv_sec; \
		(result)->tv_usec += 1000000; \
	} \
} while (0)
#endif

int ai_alsa_xrun(audio_in_t *ai)
{
    snd_pcm_status_t *status;
    int res;
	
    snd_pcm_status_alloca(&status);
    if ((res = snd_pcm_status(ai->alsa.handle, status))<0) {
	mp_msg(MSGT_TV, MSGL_ERR, "ALSA status error: %s", snd_strerror(res));
	return -1;
    }
    if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) {
	struct timeval now, diff, tstamp;
	gettimeofday(&now, 0);
	snd_pcm_status_get_trigger_tstamp(status, &tstamp);
	timersub(&now, &tstamp, &diff);
	mp_msg(MSGT_TV, MSGL_ERR, "ALSA xrun!!! (at least %.3f ms long)\n",
	       diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
	if (mp_msg_test(MSGT_TV, MSGL_V)) {
	    mp_msg(MSGT_TV, MSGL_ERR, "ALSA Status:\n");
	    snd_pcm_status_dump(status, ai->alsa.log);
	}
	if ((res = snd_pcm_prepare(ai->alsa.handle))<0) {
	    mp_msg(MSGT_TV, MSGL_ERR, "ALSA xrun: prepare error: %s", snd_strerror(res));
	    return -1;
	}
	return 0;		/* ok, data should be accepted again */
    }
    mp_msg(MSGT_TV, MSGL_ERR, "ALSA read/write error");
    return -1;
}

#endif /* HAVE_ALSA9 */