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 |
21 +++++++++------------
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,
24 insertions(+),
25 deletions(-)
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index
3ecfc8e67b..
78274314c8
100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -
93,
7 +
93,
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);
+ config.rtp.rtcp_event_observer, config.rtp.local_ssrc);
}
} // namespace
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index
6b4d0e30a2..b913acd88e
100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -
130,
7 +
130,
8 @@ class ChannelReceive : public ChannelReceiveInterface,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer,
- RtcpEventObserver* rtcp_event_observer);
+ RtcpEventObserver* rtcp_event_observer,
+ uint32_t local_ssrc);
~ChannelReceive() override;
void SetSink(AudioSinkInterface* sink) override;
@@ -
559,
7 +
560,
8 @@ ChannelReceive::ChannelReceive(
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer,
- RtcpEventObserver* rtcp_event_observer)
+ RtcpEventObserver* rtcp_event_observer,
+ uint32_t local_ssrc)
: env_(env),
worker_thread_(TaskQueueBase::Current()),
rtp_receive_statistics_(ReceiveStatistics::Create(&env_.clock())),
@@ -
595,
14 +
597,
8 @@ ChannelReceive::ChannelReceive(
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
- rtp_rtcp_ =
- ModuleRtpRtcpImpl2::CreateReceiveModule(env_, configuration, [this] {
- if (packet_router_ == nullptr) {
- return kFallbackRtcpSsrcForAudio;
- }
- return packet_router_->SsrcOfFirstSender().value_or(
- kFallbackRtcpSsrcForAudio);
- });
+ rtp_rtcp_ = ModuleRtpRtcpImpl2::CreateReceiveModule(
+ env_, configuration, [local_ssrc] { return local_ssrc; });
rtp_rtcp_->SetRemoteSSRC(remote_ssrc_);
// Ensure that RTCP is enabled for the created channel.
@@ -
1207,
13 +
1203,
14 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer,
- RtcpEventObserver* rtcp_event_observer) {
+ RtcpEventObserver* rtcp_event_observer,
+ 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);
+ rtcp_event_observer, local_ssrc);
}
} // namespace voe
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index
952784c168..baa3dffb4a
100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -
183,
7 +
183,
8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer,
- RtcpEventObserver* rtcp_event_observer);
+ RtcpEventObserver* rtcp_event_observer,
+ 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 9e4431bdb1..7744fe1e09 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -122,6 +122,10 @@ FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
.outgoing_transport = config.rtcp_send_transport,
.rtt_stats = rtt_stats},
[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 9bbff9a156..a5e74e1a5b 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -129,7 +129,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
RtcpCnameCallback* rtcp_cname_callback,
PacketRouter* packet_router,
bool non_sender_rtt_measurement,
- RtcpEventObserver* rtcp_event_observer) {
+ RtcpEventObserver* rtcp_event_observer,
+ uint32_t local_ssrc) {
RtpRtcpInterface::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = true;
@@ -143,14 +144,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
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; });
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
@@ -323,7 +317,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
rtcp_cname_callback,
packet_router,
config_.rtp.rtcp_xr.receiver_reference_time_report,
- config_.rtp.rtcp_event_observer)),
+ config_.rtp.rtcp_event_observer,
+ config_.rtp.local_ssrc)),
nack_periodic_processor_(nack_periodic_processor),
complete_frame_callback_(complete_frame_callback),
keyframe_request_method_(config_.rtp.keyframe_method),
