ao_sun.c

自己移植的linux下的流媒体播放器原代码,支持mms协议,支持ftp和http协议.

	if ((sun_mixer_device = mixer_device) == NULL || !sun_mixer_device[0]) {
	    sun_mixer_device = malloc(strlen(audio_dev) + 4);
	    strcpy(sun_mixer_device, audio_dev);
	    strcat(sun_mixer_device, "ctl");
	}
    }

    if (ao_subdevice) audio_dev = ao_subdevice;
}

// to set/get/query special features/parameters
static int control(int cmd,void *arg){
    switch(cmd){
    case AOCONTROL_SET_DEVICE:
	audio_dev=(char*)arg;
	return CONTROL_OK;
    case AOCONTROL_QUERY_FORMAT:
	return CONTROL_TRUE;
    case AOCONTROL_GET_VOLUME:
    {
        int fd;

	if ( !sun_mixer_device )    /* control function is used before init? */
	    setup_device_paths();

	fd=open( sun_mixer_device,O_RDONLY );
	if ( fd != -1 )
	{
	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
	    float volume;
	    struct audio_info info;
	    ioctl( fd,AUDIO_GETINFO,&info);
	    volume = info.play.gain * 100. / AUDIO_MAX_GAIN;
	    if ( info.play.balance == AUDIO_MID_BALANCE ) {
		vol->right = vol->left = volume;
	    } else if ( info.play.balance < AUDIO_MID_BALANCE ) {
		vol->left  = volume;
		vol->right = volume * info.play.balance / AUDIO_MID_BALANCE;
	    } else {
		vol->left  = volume * (AUDIO_RIGHT_BALANCE-info.play.balance)
							/ 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;
        int fd;

	if ( !sun_mixer_device )    /* control function is used before init? */
	    setup_device_paths();

	fd=open( sun_mixer_device,O_RDONLY );
	if ( fd != -1 )
	{
	    struct audio_info info;
	    float volume;
	    AUDIO_INITINFO(&info);
	    volume = vol->right > vol->left ? vol->right : vol->left;
	    if ( volume != 0 ) {
		info.play.gain = volume * AUDIO_MAX_GAIN / 100;
		if ( vol->right == vol->left )
		    info.play.balance = AUDIO_MID_BALANCE;
		else
		    info.play.balance = (vol->right - vol->left + volume) * AUDIO_RIGHT_BALANCE / (2*volume);
	    }
#if !defined (__OpenBSD__) && !defined (__NetBSD__)
	    info.output_muted = (volume == 0);
#endif
	    ioctl( fd,AUDIO_SETINFO,&info );
	    close( fd );
	    return CONTROL_OK;
	}	
	return CONTROL_ERROR;
    }
    }
    return CONTROL_UNKNOWN;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){

    audio_info_t info;
    int pass;
    int ok;
    int convert_u8_s8;

    setup_device_paths();

    if (enable_sample_timing == RTSC_UNKNOWN
	&& !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
	enable_sample_timing = realtime_samplecounter_available(audio_dev);
    }

    printf("ao2: %d Hz  %d chans  %s [0x%X]\n",
	   rate,channels,af_fmt2str_short(format),format);

    audio_fd=open(audio_dev, O_WRONLY);
    if(audio_fd<0){
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno));
	return 0;
    }

    ioctl(audio_fd, AUDIO_DRAIN, 0);

    if (af2sunfmt(format) == AUDIO_ENCODING_NONE)
      format = AF_FORMAT_S16_NE;

    for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */

	AUDIO_INITINFO(&info);
	info.play.encoding = af2sunfmt(ao_data.format = format);
	info.play.precision =
	    (format==AF_FORMAT_S16_NE
	     ? AUDIO_PRECISION_16
	     : AUDIO_PRECISION_8);
	info.play.channels = ao_data.channels = channels;
	info.play.sample_rate = ao_data.samplerate = rate;

	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.
	     */
#ifdef AUDIO_ENCODING_LINEAR8
	    if (info.play.encoding != AUDIO_ENCODING_LINEAR8)
#endif
		continue;
	    info.play.encoding = AUDIO_ENCODING_LINEAR;
	    convert_u8_s8 = 1;
	}

	if (pass & 2) {
	    /*
	     * on some sun audio drivers, only certain fixed sample rates are
	     * supported.
	     *
	     * 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.
	     *
	     * ao_data.samplerate = info.play.sample_rate;
	     */
	}

	if (pass & 4) {
	    /* like "pass & 2", but use the highest supported sample rate */
	    if (!(info.play.sample_rate
		  = ao_data.samplerate
		  = find_highest_samplerate(audio_fd)))
		continue;
	}

	ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
	if (ok) {
	    /* audio format accepted by audio driver */
	    break;
	}

	/*
	 * format not supported?
	 * retry with different encoding and/or sample rate
	 */
    }

    if (!ok) {
	char buf[128];
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
	       channels, af_fmt2str(format, buf, 128), rate);
	return 0;
    }

    if (convert_u8_s8)
      ao_data.format = AF_FORMAT_S8;

    bytes_per_sample = channels * info.play.precision / 8;
    ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate;
    ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;

#ifdef	__not_used__
    /*
     * hmm, ao_data.buffersize is currently not used in this driver, do there's
     * no need to measure it
     */
    if(ao_data.buffersize==-1){
	// Measuring buffer size:
	void* data;
	ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
	data = malloc(ao_data.outburst);
	memset(data, format==AF_FORMAT_U8 ? 0x80 : 0, ao_data.outburst);
	while(ao_data.buffersize<0x40000){
	    fd_set rfds;
	    struct timeval tv;
	    FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
	    tv.tv_sec=0; tv.tv_usec = 0;
	    if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
	    write(audio_fd,data,ao_data.outburst);
	    ao_data.buffersize+=ao_data.outburst;
	}
	free(data);
	if(ao_data.buffersize==0){
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantUseSelect);
	    return 0;
	}
#ifdef	__svr4__
	// remove the 0 bytes from the above ao_data.buffersize measurement from the
	// audio driver's STREAMS queue
	ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
	ioctl(audio_fd, AUDIO_DRAIN, 0);
#endif
    }
#endif	/* __not_used__ */

    AUDIO_INITINFO(&info);
    info.play.samples = 0;
    info.play.eof = 0;
    info.play.error = 0;
    ioctl (audio_fd, AUDIO_SETINFO, &info);

    queued_bursts = 0;
    queued_samples = 0;

    return 1;
}

// close audio device
static void uninit(int immed){
#ifdef	__svr4__
    // throw away buffered data in the audio driver's STREAMS queue
    if (immed)
	ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
    close(audio_fd);
}

// stop playing and empty buffers (for seeking/pause)
static void reset(){
    audio_info_t info;

    uninit(1);
    audio_fd=open(audio_dev, O_WRONLY);
    if(audio_fd<0){
	mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_SUN_CantReopenReset, strerror(errno));
	return;
    }

    ioctl(audio_fd, AUDIO_DRAIN, 0);

    AUDIO_INITINFO(&info);
    info.play.encoding = af2sunfmt(ao_data.format);
    info.play.precision =
	(ao_data.format==AF_FORMAT_S16_NE 
	 ? AUDIO_PRECISION_16
	 : AUDIO_PRECISION_8);
    info.play.channels = ao_data.channels;
    info.play.sample_rate = ao_data.samplerate;
    info.play.samples = 0;
    info.play.eof = 0;
    info.play.error = 0;
    ioctl (audio_fd, AUDIO_SETINFO, &info);
    queued_bursts = 0;
    queued_samples = 0;
}

// stop playing, keep buffers (for pause)
static void audio_pause()
{
    struct audio_info info;
    AUDIO_INITINFO(&info);
    info.play.pause = 1;
    ioctl(audio_fd, AUDIO_SETINFO, &info);
}

// resume playing, after audio_pause()
static void audio_resume()
{
    struct audio_info info;
    AUDIO_INITINFO(&info);
    info.play.pause = 0;
    ioctl(audio_fd, AUDIO_SETINFO, &info);
}


// return: how many bytes can be played without blocking
static int get_space(){
    audio_info_t info;

    // check buffer
#ifdef HAVE_AUDIO_SELECT
    {
	fd_set rfds;
	struct timeval tv;
	FD_ZERO(&rfds);
	FD_SET(audio_fd, &rfds);
	tv.tv_sec = 0;
	tv.tv_usec = 0;
	if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
    }
#endif

#if !defined (__OpenBSD__) && !defined(__NetBSD__)
    ioctl(audio_fd, AUDIO_GETINFO, &info);
    if (queued_bursts - info.play.eof > 2)
	return 0;
#endif

#if defined(__NetBSD__) || defined(__OpenBSD__)
    ioctl(audio_fd, AUDIO_GETINFO, &info);
    return info.hiwat * info.blocksize - info.play.seek;
#else
    return ao_data.outburst;
#endif

}

// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
    if (len < ao_data.outburst) return 0;
    len /= ao_data.outburst;
    len *= ao_data.outburst;

    len = write(audio_fd, data, len);
    if(len > 0) {
	queued_samples += len / bytes_per_sample;
	if (write(audio_fd,data,0) < 0)
	    perror("ao_sun: send EOF audio record");
	else
	    queued_bursts ++;
    }
    return len;
}


// return: delay in seconds between first and last sample in buffer
static float get_delay(){
    audio_info_t info;
    ioctl(audio_fd, AUDIO_GETINFO, &info);
#if defined (__OpenBSD__) || defined(__NetBSD__)
    return (float) info.play.seek/ (float)byte_per_sec ;
#else
    if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
	return (float)(queued_samples - info.play.samples) / (float)ao_data.samplerate;
    else
	return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
#endif
}