4 - Implement a better way to calculate current audio pts and use it for

audio decoded with ad_libvorbis, ad_ffmpeg and ad_faad.

Patch by Uoti Urpala


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18243 b3059339-0415-0410-9bf9-f77b7e298cf2

3 files changed, 21 insertions, 3 deletions

diff --git a/libmpcodecs/ad_faad.c b/libmpcodecs/ad_faad.c
index b681a9c7a3..80470d4a49 100644
--- a/libmpcodecs/ad_faad.c
+++ b/libmpcodecs/ad_faad.c
@@ -253,8 +253,13 @@ static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
   } else {
    // packetized (.mp4) aac stream:
     unsigned char* bufptr=NULL;
-    int buflen=ds_get_packet(sh->ds, &bufptr);
+    double pts;
+    int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
     if(buflen<=0) break;
+    if (pts != MP_NOPTS_VALUE) {
+	sh->pts = pts;
+	sh->pts_bytes = 0;
+    }
     faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
   }
   //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
@@ -271,6 +276,7 @@ static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
       memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
       last_dec_len = sh->samplesize*faac_finfo.samples;
       len += last_dec_len;
+      sh->pts_bytes += last_dec_len;
     //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
     }
   }
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
index 22cb72b8a5..f9fb0455a6 100644
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -149,8 +149,13 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
     int y,len=-1;
     while(len<minlen){
 	int len2=0;
-	int x=ds_get_packet(sh_audio->ds,&start);
+	double pts;
+	int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
 	if(x<=0) break; // error
+	if (pts != MP_NOPTS_VALUE) {
+	    sh_audio->pts = pts;
+	    sh_audio->pts_bytes = 0;
+	}
 	y=avcodec_decode_audio(sh_audio->context,(int16_t*)buf,&len2,start,x);
 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
 	if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
@@ -159,6 +164,7 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
 	  //len=len2;break;
 	  if(len<0) len=len2; else len+=len2;
 	  buf+=len2;
+	  sh_audio->pts_bytes += len2;
 	}
         mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
     }
diff --git a/libmpcodecs/ad_libvorbis.c b/libmpcodecs/ad_libvorbis.c
index 3c765b6acd..2d94ac4431 100644
--- a/libmpcodecs/ad_libvorbis.c
+++ b/libmpcodecs/ad_libvorbis.c
@@ -229,9 +229,14 @@ static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
 	while(len < minlen) {
 	  while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))<=0){
 	    ogg_packet op;
+	    double pts;
 	    memset(&op,0,sizeof(op)); //op.b_o_s = op.e_o_s = 0;
-	    op.bytes = ds_get_packet(sh->ds,&op.packet);
+	    op.bytes = ds_get_packet_pts(sh->ds,&op.packet, &pts);
 	    if(op.bytes<=0) break;
+	    if (pts != MP_NOPTS_VALUE) {
+		sh->pts = pts;
+		sh->pts_bytes = 0;
+	    }
 	    if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */
 	      vorbis_synthesis_blockin(&ov->vd,&ov->vb);
 	  }
@@ -303,6 +308,7 @@ static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
 	    if(clipflag)
 	      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence));
 	    len+=2*ov->vi.channels*bout;
+	    sh->pts_bytes += 2*ov->vi.channels*bout;
 	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
 	    samples-=bout;
 	    vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how