commit d46a174f0c4d7bca52d1caeeb29ce59c987bb869
author Yura Yaroshevich <yura.yaroshevich@gmail.com> Wed Apr 14 13:26:14 2021
committer Commit Bot <commit-bot@chromium.org> Sun Apr 18 21:53:32 2021
tree 66c067f60fa3b93a02cd1a366a4e409add7316d2
parent 7fa8d4651629505fa9ae4f8477fec41a96ff7552

Expose adaptive_ptime from iOS SDK.

Bug: None
Change-Id: I48fd0937f51dc972b3eccd66f99ae80378e32fe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214968
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33766}

modules/audio_coding/codecs/opus/audio_encoder_opus.cc

diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 203cb5ae..7c62e98 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -704,6 +704,11 @@
 }
 
 void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
+  if (next_frame_length_ms_ != frame_length_ms) {
+    RTC_LOG(LS_VERBOSE) << "Update Opus frame length "
+                        << "from " << next_frame_length_ms_ << " ms "
+                        << "to " << frame_length_ms << " ms.";
+  }
   next_frame_length_ms_ = frame_length_ms;
 }
 

sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h

diff --git a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
index af6d583..07f6b7a 100644
--- a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
+++ b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
@@ -65,6 +65,10 @@
 /** The relative DiffServ Code Point priority. */
 @property(nonatomic, assign) RTCPriority networkPriority;
 
+/** Allow dynamic frame length changes for audio:
+ https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime */
+@property(nonatomic, assign) BOOL adaptiveAudioPacketTime;
+
 - (instancetype)init;
 
 @end

sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm

diff --git a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
index a42439f..d6087da 100644
--- a/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
@@ -24,6 +24,7 @@
 @synthesize ssrc = _ssrc;
 @synthesize bitratePriority = _bitratePriority;
 @synthesize networkPriority = _networkPriority;
+@synthesize adaptiveAudioPacketTime = _adaptiveAudioPacketTime;
 
 - (instancetype)init {
   webrtc::RtpEncodingParameters nativeParameters;
@@ -61,6 +62,7 @@
     _bitratePriority = nativeParameters.bitrate_priority;
     _networkPriority = [RTC_OBJC_TYPE(RTCRtpEncodingParameters)
         priorityFromNativePriority:nativeParameters.network_priority];
+    _adaptiveAudioPacketTime = nativeParameters.adaptive_ptime;
   }
   return self;
 }
@@ -93,6 +95,7 @@
   parameters.bitrate_priority = _bitratePriority;
   parameters.network_priority =
       [RTC_OBJC_TYPE(RTCRtpEncodingParameters) nativePriorityFromPriority:_networkPriority];
+  parameters.adaptive_ptime = _adaptiveAudioPacketTime;
   return parameters;
 }