comparison mplayer.c @ 22488:b6aa3eedf49b

Move the sh_audio->delay field to mpctx->delay. The value is related to overall a/v sync and is not used by audio demuxers or decoders.

22487:1c5ea79749ea

        (int)(100.0*vout_time_usage*playback_speed/(double)sh_video->timer),
        (100.0*audio_time_usage*playback_speed/(double)sh_video->timer));
    else
      saddf(line, &pos, width, "??%% ??%% ??,?%% ");
  } else if (mpctx->sh_audio) {
    if (mpctx->sh_audio->delay > 0.5)
      saddf(line, &pos, width, "%4.1f%% ",
        100.0*audio_time_usage/(double)mpctx->sh_audio->delay);
    else
      saddf(line, &pos, width, "??,?%% ");
  }

  // VO stats

	     * trying to measure.  This keeps the two from competing, but still
	     * allows the code to correct for PTS drift *only*.  (Using a delay
	     * value here, even a "corrected" one, would be incompatible with
	     * autosync mode.)
	     */
	    a_pts = written_audio_pts(mpctx->sh_audio, mpctx->d_audio) - mpctx->sh_audio->delay;
	else
	    a_pts = playing_audio_pts(mpctx->sh_audio, mpctx->d_audio, mpctx->audio_out);

	v_pts = mpctx->sh_video->pts;

	    if (default_max_pts_correction >= 0)
		max_pts_correction = default_max_pts_correction;
	    else
		max_pts_correction = mpctx->sh_video->frametime*0.10; // +-10% of time
	    if (!between_frames) {
		mpctx->sh_audio->delay+=x;
		c_total+=x;
	    }
	    if(!quiet)
		print_status(a_pts - audio_delay, AV_delay, c_total);
	}

    while (1) {
	// all the current uses of ao_data.pts seem to be in aos that handle
	// sync completely wrong; there should be no need to use ao_data.pts
	// in get_space()
	ao_data.pts = ((mpctx->sh_video?mpctx->sh_video->timer:0)+sh_audio->delay)*90000.0;
	bytes_to_write = mpctx->audio_out->get_space();
	if (mpctx->sh_video || bytes_to_write >= ao_data.outburst)
	    break;

	// handle audio-only case:

	current_module="play_audio";

	// Is this pts value actually useful for the aos that access it?
	// They're obviously badly broken in the way they handle av sync;
	// would not having access to this make them more broken?
	ao_data.pts = ((mpctx->sh_video?mpctx->sh_video->timer:0)+sh_audio->delay)*90000.0;
	playsize = mpctx->audio_out->play(sh_audio->a_out_buffer, playsize, playflags);

	if (playsize > 0) {
	    sh_audio->a_out_buffer_len -= playsize;
	    memmove(sh_audio->a_out_buffer, &sh_audio->a_out_buffer[playsize],
		    sh_audio->a_out_buffer_len);
	    sh_audio->delay += playback_speed*playsize/(double)ao_data.bps;
	}
	else if (audio_eof && mpctx->audio_out->get_delay() < .04) {
	    // Sanity check to avoid hanging in case current ao doesn't output
	    // partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
	    mp_msg(MSGT_CPLAYER, MSGL_WARN, "Audio output truncated at end.\n");

	     * closer to the delay value gets to that which "-nosound"
	     * would have used, and the longer it will take for A/V
	     * sync to settle at the right value (but it eventually will.)
	     * This settling time is very short for values below 100.
	     */
	    float predicted = mpctx->sh_audio->delay / playback_speed + *time_frame;
	    float difference = delay - predicted;
	    delay = predicted + difference / (float)autosync;
	}

	*time_frame = delay - mpctx->sh_audio->delay / playback_speed;

	// delay = amount of audio buffered in soundcard/driver
	if (delay > 0.25) delay=0.25; else
	if (delay < 0.10) delay=0.10;
	if (*time_frame > delay*0.6) {

	    return -1;
	if (in_size > max_framesize)
	    max_framesize = in_size; // stats
	sh_video->timer += frame_time;
	if (mpctx->sh_audio)
	    mpctx->sh_audio->delay -= frame_time;
	// video_read_frame can change fps (e.g. for ASF video)
	vo_fps = sh_video->fps;
	// check for frame-drop:
	current_module = "check_framedrop";
	if (mpctx->sh_audio && !mpctx->d_audio->eof) {
	    static int dropped_frames;
	    float delay = playback_speed*mpctx->audio_out->get_delay();
	    float d = delay-mpctx->sh_audio->delay;
	    // we should avoid dropping too many frames in sequence unless we
	    // are too late. and we allow 100ms A-V delay here:
	    if (d < -dropped_frames*frame_time-0.100 &&
				mpctx->osd_function != OSD_PAUSE) {
		drop_frame = frame_dropping;

	}
	frame_time = sh_video->pts - sh_video->last_pts;
	sh_video->last_pts = sh_video->pts;
	sh_video->timer += frame_time;
	if(mpctx->sh_audio)
	    mpctx->sh_audio->delay -= frame_time;
	*blit_frame = 1;
    }
    return frame_time;
}

	if (vo_config_count)
	    mpctx->video_out->control(VOCTRL_RESET, NULL);
	mpctx->sh_video->num_buffered_pts = 0;
	mpctx->sh_video->last_pts = MP_NOPTS_VALUE;
	mpctx->num_buffered_frames = 0;
	// Not all demuxers set d_video->pts during seek, so this value
	// (which is used by at least vobsub and edl code below) may
	// be completely wrong (probably 0).
	mpctx->sh_video->pts = mpctx->d_video->pts;
	update_subtitles(mpctx->sh_video, mpctx->d_sub, 1);

    audio_delay += mpctx->sh_video->stream_delay;
}
if(mpctx->sh_audio){
  if (! ignore_start)
    audio_delay -= mpctx->sh_audio->stream_delay;
  mpctx->sh_audio->delay=-audio_delay;
}

if(!mpctx->sh_audio){
  mp_msg(MSGT_CPLAYER,MSGL_INFO,MSGTR_NoSound);
  mp_msg(MSGT_CPLAYER,MSGL_V,"Freeing %d unused audio chunks.\n",mpctx->d_audio->packs);

	 off_t len = ( mpctx->demuxer->movi_end - mpctx->demuxer->movi_start );
	 off_t pos = ( mpctx->demuxer->file_format == DEMUXER_TYPE_AUDIO?mpctx->stream->pos:mpctx->demuxer->filepos );
	 guiIntfStruct.Position=(len <= 0? 0.0f : ( pos - mpctx->demuxer->movi_start ) * 100.0f / len );
	}
	if ( mpctx->sh_video ) guiIntfStruct.TimeSec=mpctx->sh_video->pts;
	  else if ( mpctx->sh_audio ) guiIntfStruct.TimeSec=mpctx->sh_audio->delay;
	guiIntfStruct.LengthInSec=demuxer_get_time_length(mpctx->demuxer);
	guiGetEvent( guiReDraw,NULL );
	guiGetEvent( guiSetVolume,NULL );
	if(guiIntfStruct.Playing==0) break; // STOP
	if(guiIntfStruct.Playing==2) mpctx->osd_function=OSD_PAUSE;

22488:b6aa3eedf49b

        (int)(100.0*vout_time_usage*playback_speed/(double)sh_video->timer),
        (100.0*audio_time_usage*playback_speed/(double)sh_video->timer));
    else
      saddf(line, &pos, width, "??%% ??%% ??,?%% ");
  } else if (mpctx->sh_audio) {
    if (mpctx->delay > 0.5)
      saddf(line, &pos, width, "%4.1f%% ",
        100.0*audio_time_usage/(double)mpctx->delay);
    else
      saddf(line, &pos, width, "??,?%% ");
  }

  // VO stats

	     * trying to measure.  This keeps the two from competing, but still
	     * allows the code to correct for PTS drift *only*.  (Using a delay
	     * value here, even a "corrected" one, would be incompatible with
	     * autosync mode.)
	     */
	    a_pts = written_audio_pts(mpctx->sh_audio, mpctx->d_audio) - mpctx->delay;
	else
	    a_pts = playing_audio_pts(mpctx->sh_audio, mpctx->d_audio, mpctx->audio_out);

	v_pts = mpctx->sh_video->pts;

	    if (default_max_pts_correction >= 0)
		max_pts_correction = default_max_pts_correction;
	    else
		max_pts_correction = mpctx->sh_video->frametime*0.10; // +-10% of time
	    if (!between_frames) {
		mpctx->delay+=x;
		c_total+=x;
	    }
	    if(!quiet)
		print_status(a_pts - audio_delay, AV_delay, c_total);
	}

    while (1) {
	// all the current uses of ao_data.pts seem to be in aos that handle
	// sync completely wrong; there should be no need to use ao_data.pts
	// in get_space()
	ao_data.pts = ((mpctx->sh_video?mpctx->sh_video->timer:0)+mpctx->delay)*90000.0;
	bytes_to_write = mpctx->audio_out->get_space();
	if (mpctx->sh_video || bytes_to_write >= ao_data.outburst)
	    break;

	// handle audio-only case:

	current_module="play_audio";

	// Is this pts value actually useful for the aos that access it?
	// They're obviously badly broken in the way they handle av sync;
	// would not having access to this make them more broken?
	ao_data.pts = ((mpctx->sh_video?mpctx->sh_video->timer:0)+mpctx->delay)*90000.0;
	playsize = mpctx->audio_out->play(sh_audio->a_out_buffer, playsize, playflags);

	if (playsize > 0) {
	    sh_audio->a_out_buffer_len -= playsize;
	    memmove(sh_audio->a_out_buffer, &sh_audio->a_out_buffer[playsize],
		    sh_audio->a_out_buffer_len);
	    mpctx->delay += playback_speed*playsize/(double)ao_data.bps;
	}
	else if (audio_eof && mpctx->audio_out->get_delay() < .04) {
	    // Sanity check to avoid hanging in case current ao doesn't output
	    // partial chunks and doesn't check for AOPLAY_FINAL_CHUNK
	    mp_msg(MSGT_CPLAYER, MSGL_WARN, "Audio output truncated at end.\n");

	     * closer to the delay value gets to that which "-nosound"
	     * would have used, and the longer it will take for A/V
	     * sync to settle at the right value (but it eventually will.)
	     * This settling time is very short for values below 100.
	     */
	    float predicted = mpctx->delay / playback_speed + *time_frame;
	    float difference = delay - predicted;
	    delay = predicted + difference / (float)autosync;
	}

	*time_frame = delay - mpctx->delay / playback_speed;

	// delay = amount of audio buffered in soundcard/driver
	if (delay > 0.25) delay=0.25; else
	if (delay < 0.10) delay=0.10;
	if (*time_frame > delay*0.6) {

	    return -1;
	if (in_size > max_framesize)
	    max_framesize = in_size; // stats
	sh_video->timer += frame_time;
	if (mpctx->sh_audio)
	    mpctx->delay -= frame_time;
	// video_read_frame can change fps (e.g. for ASF video)
	vo_fps = sh_video->fps;
	// check for frame-drop:
	current_module = "check_framedrop";
	if (mpctx->sh_audio && !mpctx->d_audio->eof) {
	    static int dropped_frames;
	    float delay = playback_speed*mpctx->audio_out->get_delay();
	    float d = delay-mpctx->delay;
	    // we should avoid dropping too many frames in sequence unless we
	    // are too late. and we allow 100ms A-V delay here:
	    if (d < -dropped_frames*frame_time-0.100 &&
				mpctx->osd_function != OSD_PAUSE) {
		drop_frame = frame_dropping;

	}
	frame_time = sh_video->pts - sh_video->last_pts;
	sh_video->last_pts = sh_video->pts;
	sh_video->timer += frame_time;
	if(mpctx->sh_audio)
	    mpctx->delay -= frame_time;
	*blit_frame = 1;
    }
    return frame_time;
}

	if (vo_config_count)
	    mpctx->video_out->control(VOCTRL_RESET, NULL);
	mpctx->sh_video->num_buffered_pts = 0;
	mpctx->sh_video->last_pts = MP_NOPTS_VALUE;
	mpctx->num_buffered_frames = 0;
	mpctx->delay = 0;
	// Not all demuxers set d_video->pts during seek, so this value
	// (which is used by at least vobsub and edl code below) may
	// be completely wrong (probably 0).
	mpctx->sh_video->pts = mpctx->d_video->pts;
	update_subtitles(mpctx->sh_video, mpctx->d_sub, 1);

    audio_delay += mpctx->sh_video->stream_delay;
}
if(mpctx->sh_audio){
  if (! ignore_start)
    audio_delay -= mpctx->sh_audio->stream_delay;
  mpctx->delay=-audio_delay;
}

if(!mpctx->sh_audio){
  mp_msg(MSGT_CPLAYER,MSGL_INFO,MSGTR_NoSound);
  mp_msg(MSGT_CPLAYER,MSGL_V,"Freeing %d unused audio chunks.\n",mpctx->d_audio->packs);

	 off_t len = ( mpctx->demuxer->movi_end - mpctx->demuxer->movi_start );
	 off_t pos = ( mpctx->demuxer->file_format == DEMUXER_TYPE_AUDIO?mpctx->stream->pos:mpctx->demuxer->filepos );
	 guiIntfStruct.Position=(len <= 0? 0.0f : ( pos - mpctx->demuxer->movi_start ) * 100.0f / len );
	}
	if ( mpctx->sh_video ) guiIntfStruct.TimeSec=mpctx->sh_video->pts;
	  else if ( mpctx->sh_audio ) guiIntfStruct.TimeSec=mpctx->delay;
	guiIntfStruct.LengthInSec=demuxer_get_time_length(mpctx->demuxer);
	guiGetEvent( guiReDraw,NULL );
	guiGetEvent( guiSetVolume,NULL );
	if(guiIntfStruct.Playing==0) break; // STOP
	if(guiIntfStruct.Playing==2) mpctx->osd_function=OSD_PAUSE;