sousa-gecko/third_party/libwebrtc/moz-patch-stack/s0141.patch

From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 4 May 2026 19:30:00 +0000
Subject: Bug 2034935 - Undelete deprecation of local_ssrc for recv streams.
 r=webrtc-reviewers,dbaker

This avoids regressing Teams.

Differential Revision: https://phabricator.services.mozilla.com/D297529
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/bc8a96604aaa6f1fbcd58261bf4bd6692cca3d3f
---
 audio/audio_receive_stream.cc       |  2 +-
 audio/channel_receive.cc            | 20 ++++++++++----------
 audio/channel_receive.h             |  3 ++-
 call/flexfec_receive_stream_impl.cc |  4 ++++
 call/receive_stream.h               |  4 +++-
 video/rtp_video_stream_receiver2.cc | 15 +++++----------
 6 files changed, 25 insertions(+), 23 deletions(-)

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 248a3e14bb..43c0f1ad26 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -95,7 +95,7 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
       config.jitter_buffer_min_delay_ms, config.enable_non_sender_rtt,
       config.decoder_factory, std::move(config.frame_decryptor),
       config.crypto_options, std::move(config.frame_transformer),
-      config.rtp.rtcp_event_observer, packet_router);
+      config.rtp.rtcp_event_observer, packet_router, config.rtp.local_ssrc);
 }
 }  // namespace
 
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 953532c2a5..95315a3709 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -123,7 +123,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
     bool enable_non_sender_rtt,
     uint32_t remote_ssrc,
     RtcpEventObserver* rtcp_event_observer,
-    PacketRouter* absl_nonnull packet_router) {
+    uint32_t local_ssrc) {
   RtpRtcpInterface::Configuration configuration = {
       .audio = true,
       .receiver_only = true,
@@ -138,10 +138,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
 
   std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp =
       ModuleRtpRtcpImpl2::CreateReceiveModule(
-          env, configuration, [packet_router]() {
-            return packet_router->SsrcOfFirstSender().value_or(
-                kFallbackRtcpSsrcForAudio);
-          });
+          env, configuration, [local_ssrc] { return local_ssrc; });
   return rtp_rtcp;
 }
 
@@ -162,7 +159,8 @@ class ChannelReceive : public ChannelReceiveInterface,
                  const CryptoOptions& crypto_options,
                  scoped_refptr<FrameTransformerInterface> frame_transformer,
                  RtcpEventObserver* rtcp_event_observer,
-                 PacketRouter* absl_nonnull packet_router);
+                 PacketRouter* absl_nonnull packet_router,
+                 uint32_t local_ssrc);
   ~ChannelReceive() override;
 
   void SetSink(AudioSinkInterface* sink) override;
@@ -590,7 +588,8 @@ ChannelReceive::ChannelReceive(
     const CryptoOptions& crypto_options,
     scoped_refptr<FrameTransformerInterface> frame_transformer,
     RtcpEventObserver* rtcp_event_observer,
-    PacketRouter* absl_nonnull packet_router)
+    PacketRouter* absl_nonnull packet_router,
+    uint32_t local_ssrc)
     : env_(env),
       worker_thread_(TaskQueueBase::Current()),
       rtp_receive_statistics_(ReceiveStatistics::Create(&env_.clock())),
@@ -601,7 +600,7 @@ ChannelReceive::ChannelReceive(
                                     enable_non_sender_rtt,
                                     remote_ssrc,
                                     rtcp_event_observer,
-                                    packet_router)),
+                                    local_ssrc)),
       remote_ssrc_(remote_ssrc),
       source_tracker_(&env_.clock()),
       neteq_(CreateNetEq(neteq_factory,
@@ -1220,13 +1219,14 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
     const CryptoOptions& crypto_options,
     scoped_refptr<FrameTransformerInterface> frame_transformer,
     RtcpEventObserver* rtcp_event_observer,
-    PacketRouter* absl_nonnull packet_router) {
+    PacketRouter* absl_nonnull packet_router,
+    uint32_t local_ssrc) {
   return std::make_unique<ChannelReceive>(
       env, neteq_factory, audio_device_module, rtcp_send_transport, remote_ssrc,
       jitter_buffer_max_packets, jitter_buffer_fast_playout,
       jitter_buffer_min_delay_ms, enable_non_sender_rtt, decoder_factory,
       std::move(frame_decryptor), crypto_options, std::move(frame_transformer),
-      rtcp_event_observer, packet_router);
+      rtcp_event_observer, packet_router, local_ssrc);
 }
 
 }  // namespace voe
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index c71f669e68..0e27ddb618 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -183,7 +183,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
     const webrtc::CryptoOptions& crypto_options,
     scoped_refptr<FrameTransformerInterface> frame_transformer,
     RtcpEventObserver* rtcp_event_observer,
-    PacketRouter* absl_nonnull packet_router);
+    PacketRouter* absl_nonnull packet_router,
+    uint32_t local_ssrc);
 
 }  // namespace voe
 }  // namespace webrtc
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 63d1b5d589..445811a1de 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -124,6 +124,10 @@ FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
            .rtcp_mode = config.rtcp_mode,
            .remote_ssrc = config.remote_ssrc},
           [packet_router] {
+            // Mozilla: Unreachable as Flexfec is not implemented.
+            //          When implementing, wire up signaled recvonly a=ssrc into
+            //          this recv_ssrsc_callback.
+            RTC_CHECK_NOTREACHED();
             // Use the same logic as for the video receiver.
             if (packet_router != nullptr) {
               return packet_router->SsrcOfFirstSender().value_or(
diff --git a/call/receive_stream.h b/call/receive_stream.h
index 231f448056..4304685ddb 100644
--- a/call/receive_stream.h
+++ b/call/receive_stream.h
@@ -37,7 +37,9 @@ class ReceiveStreamInterface {
     // This member is no longer used by WebRTC, but retained in order to
     // allow downstream code to compile.
     // TODO: issues.webrtc.org/41480926 - Delete when downstream changed.
-    [[deprecated("No longer used")]] uint32_t local_ssrc = 0;
+    // Mozilla: Undeprecated to allow using the signaled send ssrc in RTCP for
+    //          recvonly m-sections to make Teams happy.
+    uint32_t local_ssrc = 0;
   };
 
  protected:
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index f2048a2f45..450e958fbc 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -129,7 +129,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
     PacketRouter* packet_router,
     bool non_sender_rtt_measurement,
     RtcpEventObserver* rtcp_event_observer,
-    std::optional<uint32_t> remote_ssrc) {
+    std::optional<uint32_t> remote_ssrc,
+    uint32_t local_ssrc) {
   RtpRtcpInterface::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
@@ -145,14 +146,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
   configuration.remote_ssrc = remote_ssrc;
 
   auto rtp_rtcp = ModuleRtpRtcpImpl2::CreateReceiveModule(
-      env, configuration, [packet_router]() {
-        if (packet_router) {
-          return packet_router->SsrcOfFirstSender().value_or(
-              kFallbackRtcpSsrcForVideo);
-        } else {  // This happens at least in some test configurations.
-          return kFallbackRtcpSsrcForVideo;
-        }
-      });
+      env, configuration, [local_ssrc] { return local_ssrc; });
 
   return rtp_rtcp;
 }
@@ -325,7 +319,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
           packet_router,
           config_.rtp.rtcp_xr.receiver_reference_time_report,
           config_.rtp.rtcp_event_observer,
-          config->rtp.remote_ssrc)),
+          config->rtp.remote_ssrc,
+          config_.rtp.local_ssrc)),
       nack_periodic_processor_(nack_periodic_processor),
       complete_frame_callback_(complete_frame_callback),
       keyframe_request_method_(config_.rtp.keyframe_method),