view libao2/ao_alsa5.c @ 26146:20a126aaa756

ve_vfw.c: #include aviheader.h instead of wine avifmt.h Compilation was broken after libmpdemux/muxer.h started including libmpdemux/aviheader.h. ve_vfw.c included both muxer.h and loader/wine/avifmt.h, and the latter has definitions that conflict with aviheader.h ones. Fix by removing the avifmt.h include. I did not carefully check that changing the includes doesn't break any ve_vfw.c code. However it at least fixes compilation, and if the avifmt.h versions differ in some significant way then the code is fundamentally broken anyway: ve_vfw cannot use different versions of the avi struct definitions when it also uses shared muxer.h types (those must use the standard definitions to keep the type compatible with what's used in other files).

/*
  ao_alsa5 - ALSA-0.5.x output plugin for MPlayer

  (C) Alex Beregszaszi

  Thanks to Arpi for helping me ;)
*/

#include <errno.h>
#include <sys/asoundlib.h>

#include "config.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"

#include "mp_msg.h"
#include "help_mp.h"

static ao_info_t info = 
{
    "ALSA-0.5.x audio output",
    "alsa5",
    "Alex Beregszaszi",
    ""
};

LIBAO_EXTERN(alsa5)

static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static int alsa_rate = SND_PCM_RATE_CONTINUOUS;

/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
    return(CONTROL_UNKNOWN);
}

/*
    open & setup audio device
    return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
    int err;
    int cards = -1;
    snd_pcm_channel_params_t params;
    snd_pcm_channel_setup_t setup;
    snd_pcm_info_t info;
    snd_pcm_channel_info_t chninfo;

    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
	channels, af_fmt2str_short(format));

    alsa_handler = NULL;

    mp_msg(MSGT_AO, MSGL_V, "alsa-init: compiled for ALSA-%s (%d)\n", SND_LIB_VERSION_STR,
        SND_LIB_VERSION);

    if ((cards = snd_cards()) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_SoundCardNotFound);
	return(0);
    }

    ao_data.format = format;
    ao_data.channels = channels;
    ao_data.samplerate = rate_hz;
    ao_data.bps = ao_data.samplerate*ao_data.channels;
    ao_data.outburst = OUTBURST;
    ao_data.buffersize = 16384;

    memset(&alsa_format, 0, sizeof(alsa_format));
    switch (format)
    {
	case AF_FORMAT_S8:
	    alsa_format.format = SND_PCM_SFMT_S8;
	    break;
	case AF_FORMAT_U8:
	    alsa_format.format = SND_PCM_SFMT_U8;
	    break;
	case AF_FORMAT_U16_LE:
	    alsa_format.format = SND_PCM_SFMT_U16_LE;
	    break;
	case AF_FORMAT_U16_BE:
	    alsa_format.format = SND_PCM_SFMT_U16_BE;
	    break;
#ifndef WORDS_BIGENDIAN
	case AF_FORMAT_AC3:
#endif
	case AF_FORMAT_S16_LE:
	    alsa_format.format = SND_PCM_SFMT_S16_LE;
	    break;
#ifdef WORDS_BIGENDIAN
	case AF_FORMAT_AC3:
#endif
	case AF_FORMAT_S16_BE:
	    alsa_format.format = SND_PCM_SFMT_S16_BE;
	    break;
	default:
	    alsa_format.format = SND_PCM_SFMT_MPEG;
	    break;
    }
    
    switch(alsa_format.format)
    {
	case SND_PCM_SFMT_S16_LE:
	case SND_PCM_SFMT_U16_LE:
	    ao_data.bps *= 2;
	    break;
	case -1:
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(format));
	    return(0);
	default:
	    break;
    }

    switch(rate_hz)
    {
	case 8000:
	    alsa_rate = SND_PCM_RATE_8000;
	    break;
	case 11025:
	    alsa_rate = SND_PCM_RATE_11025;
	    break;
	case 16000:
	    alsa_rate = SND_PCM_RATE_16000;
	    break;
	case 22050:
	    alsa_rate = SND_PCM_RATE_22050;
	    break;
	case 32000:
	    alsa_rate = SND_PCM_RATE_32000;
	    break;
	case 44100:
	    alsa_rate = SND_PCM_RATE_44100;
	    break;
	case 48000:
	    alsa_rate = SND_PCM_RATE_48000;
	    break;
	case 88200:
	    alsa_rate = SND_PCM_RATE_88200;
	    break;
	case 96000:
	    alsa_rate = SND_PCM_RATE_96000;
	    break;
	case 176400:
	    alsa_rate = SND_PCM_RATE_176400;
	    break;
	case 192000:
	    alsa_rate = SND_PCM_RATE_192000;
	    break;
	default:
	    alsa_rate = SND_PCM_RATE_CONTINUOUS;
	    break;
    }

    alsa_format.rate = ao_data.samplerate;
    alsa_format.voices = ao_data.channels;
    alsa_format.interleave = 1;

    if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlayBackError, snd_strerror(err));
	return(0);
    }

    if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmInfoError, snd_strerror(err));
	return(0);
    }

    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_SoundcardsFound,
	cards, info.name);

    if (info.flags & SND_PCM_INFO_PLAYBACK)
    {
	memset(&chninfo, 0, sizeof(chninfo));
	chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
	if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
	{
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmChanInfoError, snd_strerror(err));
	    return(0);
	}

#ifndef __QNX__
	if (chninfo.buffer_size)
	    ao_data.buffersize = chninfo.buffer_size;
#endif

	mp_msg(MSGT_AO, MSGL_V, "alsa-init: setting preferred buffer size from driver: %d bytes\n",
	    ao_data.buffersize);
    }

    memset(&params, 0, sizeof(params));
    params.channel = SND_PCM_CHANNEL_PLAYBACK;
    params.mode = SND_PCM_MODE_STREAM;
    params.format = alsa_format;
    params.start_mode = SND_PCM_START_DATA;
    params.stop_mode = SND_PCM_STOP_ROLLOVER;
    params.buf.stream.queue_size = ao_data.buffersize;
    params.buf.stream.fill = SND_PCM_FILL_NONE;

    if ((err = snd_pcm_channel_params(alsa_handler, &params)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetParms, snd_strerror(err));
	return(0);
    }

    memset(&setup, 0, sizeof(setup));
    setup.channel = SND_PCM_CHANNEL_PLAYBACK;
    setup.mode = SND_PCM_MODE_STREAM;
    setup.format = alsa_format;
    setup.buf.stream.queue_size = ao_data.buffersize;
    setup.msbits_per_sample = ao_data.bps;
    
    if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err));
	return(0);
    }

    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ChanPrepareError, snd_strerror(err));
	return(0);
    }

    mp_msg(MSGT_AO, MSGL_INFO, "AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
	ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
	snd_pcm_get_format_name(alsa_format.format));
    return(1);
}

/* close audio device */
static void uninit(int immed)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_DrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_FlushError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_close(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmCloseError, snd_strerror(err));
	return;
    }
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetDrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetFlushError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetChanPrepareError, snd_strerror(err));
	return;
    }
}

/* stop playing, keep buffers (for pause) */
static void audio_pause(void)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseDrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseFlushError, snd_strerror(err));
	return;
    }
}

/* resume playing, after audio_pause() */
static void audio_resume(void)
{
    int err;
    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResumePrepareError, snd_strerror(err));
	return;
    }
}

/*
    plays 'len' bytes of 'data'
    returns: number of bytes played
*/
static int play(void* data, int len, int flags)
{
    int got_len;
    
    if (!len)
	return(0);
    
    if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
    {
	if (got_len == -EPIPE) /* underrun? */
	{
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_Underrun);
	    if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
	    {
		mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlaybackPrepareError, snd_strerror(got_len));
		return(0);
	    }
	    if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
	    {
		mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_WriteErrorAfterReset,
		    snd_strerror(got_len));
		return(0);
	    }
	    return(got_len); /* 2nd write was ok */
	}
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_OutPutError, snd_strerror(got_len));
	return(0);
    }
    return(got_len);
}

/* how many byes are free in the buffer */
static int get_space(void)
{
    snd_pcm_channel_status_t ch_stat;
    
    ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;

    if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
	return(0); /* error occurred */
    else
	return(ch_stat.free);
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
    snd_pcm_channel_status_t ch_stat;
    
    ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
    
    if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
	return((float)ao_data.buffersize/(float)ao_data.bps); /* error occurred */
    else
	return((float)ch_stat.count/(float)ao_data.bps);
}