commit 1e51a388bcf7b644bf4f1a2fb09363f866c67fcf
author Erik Språng <sprang@webrtc.org> Wed Dec 11 16:47:09 2019 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Dec 11 16:32:14 2019 +0000
tree 40e8a382cc2208933040be1db2c41d8be870f076
parent 184da528a739e270baf3ca866409f8e0c7d77a6f

Makes padding prefer video SSRCs instead of audio.

Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.

This CL works around that by preferring to send padding on video SSRCs.

Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4f851ba..987ae0e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -334,6 +334,11 @@
   return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
 }
 
+bool ModuleRtpRtcpImpl::IsAudioConfigured() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
+                     : false;
+}
+
 void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
   RTC_CHECK(rtp_sender_);
   rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(

modules/rtp_rtcp/source/rtp_rtcp_impl.h

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index d50b925..976653a 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -125,6 +125,8 @@
 
   bool SendingMedia() const override;
 
+  bool IsAudioConfigured() const override;
+
   void SetAsPartOfAllocation(bool part_of_allocation) override;
 
   bool OnSendingRtpFrame(uint32_t timestamp,

modules/rtp_rtcp/source/rtp_sender.cc

diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 6010d03..c993e47 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -545,6 +545,10 @@
   return sending_media_;
 }
 
+bool RTPSender::IsAudioConfigured() const {
+  return audio_configured_;
+}
+
 void RTPSender::SetTimestampOffset(uint32_t timestamp) {
   rtc::CritScope lock(&send_critsect_);
   timestamp_offset_ = timestamp;

modules/rtp_rtcp/source/rtp_sender.h

diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index cb59bb2..8915e39 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -54,6 +54,7 @@
 
   void SetSendingMediaStatus(bool enabled);
   bool SendingMedia() const;
+  bool IsAudioConfigured() const;
 
   uint32_t TimestampOffset() const;
   void SetTimestampOffset(uint32_t timestamp);