From: Mike Jerris <mike@jerris.com>
Date: Fri, 2 Oct 2015 17:52:51 +0000 (-0500)
Subject: Merge pull request #487 in FS/freeswitch from ~DRAGOS_OANCEA/freeswitch-dragos:opus... 
X-Git-Tag: v1.6.3~1^2~52
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=41431c0f09a3396e4476b2cba25e0a068923a500;p=thirdparty%2Ffreeswitch.git

Merge pull request #487 in FS/freeswitch from ~DRAGOS_OANCEA/freeswitch-dragos:opus-keep-fec-on-improvements-loss to master

* commit '6385fb863a7e800dbb2668fd0ed4e1203b1ba18e':
  FS-8161: mod_opus: Keep FEC enabled only if loss > 10 ( otherwise PLC is supposed to be better)
---

41431c0f09a3396e4476b2cba25e0a068923a500
diff --cc src/mod/codecs/mod_opus/mod_opus.c
index 52dbffc4aa,74f797512a..7829effd7a
--- a/src/mod/codecs/mod_opus/mod_opus.c
+++ b/src/mod/codecs/mod_opus/mod_opus.c
@@@ -308,17 -309,37 +308,44 @@@ static switch_bool_t switch_opus_has_fe
  	return  SWITCH_FALSE;
  }
  
+ /* this is only useful for fs = 8000 hz, the map is only used
+  * at the beginning of the call. */
+ static int switch_opus_get_fec_bitrate(int fs, int loss)
+ {
+ 	int threshold_bitrates[25] = {
+ 		15600,15200,15200,15200,14800,
+ 		14800,14800,14800,14400,14400,
+ 		14400,14000,14000,14000,13600,
+ 		13600,13600,13600,13200,13200,
+ 		13200,12800,12800,12800,12400
+ 	};
+ 
+ 	if (loss <= 0){
+ 		return SWITCH_STATUS_FALSE;
+ 	} 
+ 
+ 	if (fs == 8000) {
+ 		if (loss >=25) { 
+ 			return threshold_bitrates[24];
+ 		} else {
+ 			return threshold_bitrates[loss-1];
+ 		}
+ 	}
+ 
+ 	return SWITCH_STATUS_FALSE ; 
+ }
+ 
  static switch_status_t switch_opus_info(void * encoded_data, uint32_t len, uint32_t samples_per_second, char *print_text)
  {
 -	int nb_samples;
 -	int nb_frames;
 +	/* nb_silk_frames: number of silk-frames (10 or 20 ms) in an opus frame:  0, 1, 2 or 3 */
 +	/* computed from the 5 MSB (configuration) of the TOC byte (encoded_data[0]) */
 +	/* nb_opus_frames: number of opus frames in the packet */
 +	/* computed from the 2 LSB (p0p1) of the TOC byte */
 +	/* p0p1 = 0  => nb_opus_frames= 1 */
 +	/* p0p1 = 1 or 2  => nb_opus_frames= 2 */
 +	/* p0p1 = 3  =>  given by the 6 LSB of  encoded_data[1] */
 +
 +	int nb_samples, nb_silk_frames, nb_opus_frames, n, i; 
  	int audiobandwidth;
  	const char *audiobandwidth_str = "UNKNOWN";
  	opus_int16 frame_sizes[48];