view libao2/ao_dxr2.c @ 24590:2c238fa777ff

ao_alsa: Fix get_space() return values larger than buffersize After a buffer underrun the ALSA get_space() function sometimes returned values larger than the ao had set in ao_data.buffersize. Fix this by replacing the old check against MAX_OUTBURST by one against ao_data.buffersize. There should be no need for the MAX_OUTBURST check; the current MPlayer side should no longer have any constant limit on the amount of data an ao can buffer or request at once. The get_space() values larger than ao_data.buffersize triggered errors in audio decoding causing the current attempt to fill audio buffers to be aborted. I'm not sure how often that caused behavior noticeably worse then an underrun already is.

#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <inttypes.h>
#include <dxr2ioctl.h>
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "libavutil/common.h"
#include "mpbswap.h"

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


static ao_info_t info =
{
	"DXR2 audio output",
	"dxr2",
	"Tobias Diedrich <[e-mail]>",
	""
};

LIBAO_EXTERN(dxr2)

static int volume=19;
static int last_freq_id = -1;
extern int dxr2_fd;

// to set/get/query special features/parameters
static int control(int cmd,void *arg){
  switch(cmd){
  case AOCONTROL_GET_VOLUME:
    if(dxr2_fd > 0) {
      ao_control_vol_t* vol = (ao_control_vol_t*)arg;
      vol->left = vol->right = volume * 19.0 / 100.0;
      return CONTROL_OK;
    }
    return CONTROL_ERROR;
  case AOCONTROL_SET_VOLUME:
    if(dxr2_fd > 0) {
      dxr2_oneArg_t v;
      float diff;
      ao_control_vol_t* vol = (ao_control_vol_t*)arg;
      // We need this trick because the volume stepping is often too small
      diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
      v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff)); 
      if(v.arg > 19) v.arg = 19;
      if(v.arg < 0) v.arg = 0;
      if(v.arg != volume) {
	volume = v.arg;
	if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume);
	  return CONTROL_ERROR;
	}
      }
      return CONTROL_OK;
    }
    return CONTROL_ERROR;
  }
  return CONTROL_UNKNOWN;
}

static int freq=0;
static int freq_id=0;

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

	if(dxr2_fd <= 0)
	  return 0;

        last_freq_id = -1;
        
	ao_data.outburst=2048;
	ao_data.samplerate=rate;
	ao_data.channels=channels;
	ao_data.buffersize=2048;
	ao_data.bps=rate*4;
	ao_data.format=format;
	freq=rate;

	switch(rate){
	case 48000:
		freq_id=DXR2_AUDIO_FREQ_48;
		break;
	case 96000:
		freq_id=DXR2_AUDIO_FREQ_96;
		break;
	case 44100:
		freq_id=DXR2_AUDIO_FREQ_441;
		break;
	case 32000:
		freq_id=DXR2_AUDIO_FREQ_32;
		break;
	case 22050:
		freq_id=DXR2_AUDIO_FREQ_2205;
		break;
#ifdef DXR2_AUDIO_FREQ_24
	// This is not yet in the dxr2 driver CVS
	// you can get the patch at
	// http://www.tdiedrich.de/~ranma/patches/dxr2.pcm1723.20020513
	case 24000:
		freq_id=DXR2_AUDIO_FREQ_24;
		break;
	case 64000:
		freq_id=DXR2_AUDIO_FREQ_64;
		break;
	case 88200:
		freq_id=DXR2_AUDIO_FREQ_882;
		break;
#endif
	default:
		mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate);
		return 0;
	}

	return 1;
}

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

}

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

}

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    // for now, just call reset();
    reset();
}

// resume playing, after audio_pause()
static void audio_resume(void)
{
}

extern int vo_pts;
// return: how many bytes can be played without blocking
static int get_space(void){
    float x=(float)(vo_pts-ao_data.pts)/90000.0;
    int y;
    if(x<=0) return 0;
    y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
    if(y>32768) y=32768;
    return y;
}

static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id)
{
  extern int write_dxr2(unsigned char *data, int len);
  
  if(dxr2_fd < 0) {
    mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n");
    return;
  }    

  if(last_freq_id != freq_id) {
    ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id);
    last_freq_id = freq_id;
  }

  send_mpeg_lpcm_packet (data, len, id, timestamp, freq_id, write_dxr2);
}

// 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){
  extern int write_dxr2(unsigned char *data, int len);

  // MPEG and AC3 don't work :-(
    if(ao_data.format==AF_FORMAT_MPEG2)
      send_mpeg_ps_packet (data, len, 0xC0, ao_data.pts, 2, write_dxr2);
    else if(ao_data.format==AF_FORMAT_AC3)
      send_mpeg_ps_packet (data, len, 0x80, ao_data.pts, 2, write_dxr2);
    else {
	int i;
	//unsigned short *s=data;
	uint16_t *s=data;
#ifndef WORDS_BIGENDIAN
	for(i=0;i<len/2;i++) s[i] = bswap_16(s[i]);
#endif
	dxr2_send_lpcm_packet(data,len,0xA0,ao_data.pts-10000,freq_id);
    }
    return len;
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(void){

    return 0.0;
}