annotate stream/ai_alsa.c @ 30857:77ce62f13d42

Use the high-level QuickTime decoding APIs (DecompressSequenceFrameS and friends) instead of the unsupported, internal ones (ImageCodecBeginBand etc.). This is a prerequisite for, among others, Apple ProRes 4:2:2 support, and simplifies the file by quite a bit. Tested on Linux with all existing QuickTime codecs I could get to work in the first place; qt261, qtavui, qtsvq3 have no change. qtcvid appears to not give bit-exact the same output as before, but it looks just the same in playback to me. qt3ivx stops crashing on exit (so works better than before). With some extra patches and a codecs.conf entry, ProRes 4:2:2 also works, including on Linux. Since codec initialization is now actually done on decoder init instead of on first frame, fallback should also work a bit better (although usually, qtvideo is last in the chain). Also made the decoder complain explicitly if the demuxer data is not there (ie., the user tried to run without -demuxer mov). This patch is a cleaned up version of what Andrew Wason (rectalogic A rectalogic D com) posted to mplayer-dev-eng in June.

/*
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

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

#include "config.h"

#include <alsa/asoundlib.h>
#include "audio_in.h"
#include "mp_msg.h"
#include "help_mp.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, MSGTR_MPDEMUX_AIALSA_PcmBrokenConfig);
	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, MSGTR_MPDEMUX_AIALSA_UnavailableAccessType);
	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, MSGTR_MPDEMUX_AIALSA_UnavailableSampleFmt);
	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, MSGTR_MPDEMUX_AIALSA_UnavailableChanCount,
	       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, MSGTR_MPDEMUX_AIALSA_CannotInstallHWParams);
	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, MSGTR_MPDEMUX_AIALSA_PeriodEqualsBufferSize, 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, MSGTR_MPDEMUX_AIALSA_CannotInstallSWParams);
	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, MSGTR_MPDEMUX_AIALSA_ErrorOpeningAudio, 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, MSGTR_MPDEMUX_AIALSA_AlsaStatusError, 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, MSGTR_MPDEMUX_AIALSA_AlsaXRUN,
	       diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
	if (mp_msg_test(MSGT_TV, MSGL_V)) {
	    mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaStatus);
	    snd_pcm_status_dump(status, ai->alsa.log);
	}
	if ((res = snd_pcm_prepare(ai->alsa.handle))<0) {
	    mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaXRUNPrepareError, snd_strerror(res));
	    return -1;
	}
	return 0;		/* ok, data should be accepted again */
    }
    mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaReadWriteError);
    return -1;
}