tinkering with audio speed

1 files changed, 10 insertions, 2 deletions

diff --git a/rotord/libavexporter.h b/rotord/libavexporter.h
index fd2da93..24b656c 100644
--- a/rotord/libavexporter.h
+++ b/rotord/libavexporter.h
@@ -29,6 +29,11 @@
  * @example doc/examples/muxing.c
 */
 
+
+ //
+ //http://stackoverflow.com/questions/14871916/can-i-concurrently-read-an-earlier-section-of-an-mp4-file-while-streaming-writin
+ //
+
 #include <stdlib.h>
 #include <stdio.h>
 #include <string>
@@ -208,12 +213,15 @@ namespace libav {
             sampleptr=new uint8_t[bufsize];
             memset(sampleptr,0,bufsize);
         }
+        
         avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                  sampleptr,
                                  audio_input_frame_size *
                                  av_get_bytes_per_sample(c->sample_fmt) *
-                                 c->channels, 1);
-
+                                 c->channels, 0); //;
+        //frame->sample_rate=44100; //hard coded input rate- nope, this doesn't help
+        //frame->format=AV_SAMPLE_FMT_S16P;
+        //?? why is ffmpeg reporting fltp as the sample format??? doesn't seem to have an effect to change this though
         ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
         if (!samples) {
             free(sampleptr);