view libmpdemux/ai_alsa.c @ 15205:19243f85e164

nico partially fixed the bug i reported; here's the rest of the fix. basically demux_audio was mixing data in its header buffer in a bogus manner, whereby it could sometimes "make up" valid mpeg headers where no such header actually occurred in the file. it should be correct now. btw these changes also fix the bug where mplayer reports huge initial cpu usage for sound when playing mp3 files.

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

#include "config.h"

#if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2)) && 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 */