Files
Michael Froman 88a3d29156 Bug 1952339 - Vendor libwebrtc from 68e7b00928
Upstream commit: https://webrtc.googlesource.com/src/+/68e7b00928f99a80c65f1d15c438f46bedd839e9
    Reland "Revert "Reland "Allow sending to separate payload types for each simulcast index."""

    This is a reland of commit Ife7d43471c85fdea9bd26cc982bce410c0d75527

    In the previous implementation, there was an issue where switching codecs
    with VideoEncoderSelector would cause a crash, so I have fixed it.
    Specifically, if there are no changes to params.send_codecs, I now remove
    all parameters related to mixed-codec simulcast.
    This should ensure that when an unintended codec switch occurs,
    the behavior remains the same as before.

    Additionally, I have added a test to reproduce this crash issue.
    I confirmed that the issue occurred in the previous implementation and
    that it does not occur in the current implementation.

    Original change's description:
    > Revert "Reland "Allow sending to separate payload types for each simulcast index.""
    >
    > This reverts commit 49ac6b758cc3c28be2fc13028a829f016b453d39.
    >
    > Reason for revert: Break codec switch in singlecast.
    >
    > Original change's description:
    > > Reland "Allow sending to separate payload types for each simulcast index."
    > >
    > > This is a reland of commit bcb19c00ba8ab1788ba3c08f28ee1b23e0cc77b9
    > >
    > > Original change's description:
    > > > Allow sending to separate payload types for each simulcast index.
    > > >
    > > > This change is for mixed-codec simulcast.
    > > >
    > > > By obtaining the payload type via RtpConfig::GetStreamConfig(),
    > > > the correct payload type can be retrieved regardless of whether
    > > > RtpConfig::stream_configs is initialized or not.
    > > >
    > > > Bug: webrtc:362277533
    > > > Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
    > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
    > > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
    > > > Commit-Queue: Florent Castelli <orphis@webrtc.org>
    > > > Reviewed-by: Florent Castelli <orphis@webrtc.org>
    > > > Cr-Commit-Position: refs/heads/main@{#43197}
    > >
    > > Bug: webrtc:362277533
    > > Change-Id: Ia82c3390cceb9f68315c2fd9ba5114693669af32
    > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374780
    > > Commit-Queue: Henrik Boström <hbos@webrtc.org>
    > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
    > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
    > > Cr-Commit-Position: refs/heads/main@{#43787}
    >
    > Bug: webrtc:362277533
    > Change-Id: Ife7d43471c85fdea9bd26cc982bce410c0d75527
    > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376040
    > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
    > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
    > Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
    > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
    > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
    > Reviewed-by: Per Kjellander <perkj@webrtc.org>
    > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
    > Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
    > Cr-Commit-Position: refs/heads/main@{#43830}

    Bug: webrtc:362277533
    Change-Id: I1772fa478a4fd4d5096d9bf727ed4de045861c4e
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378100
    Reviewed-by: Erik Språng <sprang@webrtc.org>
    Commit-Queue: Henrik Boström <hbos@webrtc.org>
    Reviewed-by: Henrik Boström <hbos@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#44007}

Differential Revision: https://phabricator.services.mozilla.com/D244017
2025-03-07 17:07:22 -06:00

108 lines
4.3 KiB
Diff

From: Michael Froman <mfroman@mozilla.com>
Date: Mon, 18 Dec 2023 15:00:00 +0000
Subject: Bug 1867099 - revert libwebrtc 8602f604e0. r=bwc
Upstream 8602f604e0 removed code sending BYEs which breaks some of
our wpt. They've opened a bug for a real fix here:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15664
I've opened Bug 1870643 to track the real fix and upstream bug.
Differential Revision: https://phabricator.services.mozilla.com/D196729
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d92a578327f524ec3e1c144c82492a4c76b8266f
---
call/rtp_video_sender.cc | 1 +
modules/rtp_rtcp/source/rtcp_sender.cc | 19 +++++++++++++++++--
.../rtp_rtcp/source/rtcp_sender_unittest.cc | 5 +++--
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 1 +
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 2 +-
5 files changed, 23 insertions(+), 5 deletions(-)
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index b51ac82d97..8331122823 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -536,6 +536,7 @@ void RtpVideoSender::SetModuleIsActive(bool sending,
return;
}
+ // Sends a kRtcpByeCode when going from true to false.
rtp_module.SetSendingStatus(sending);
rtp_module.SetSendingMediaStatus(sending);
if (sending) {
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 5fa32af69b..d3f80bdf9d 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -222,8 +222,23 @@ bool RTCPSender::Sending() const {
void RTCPSender::SetSendingStatus(const FeedbackState& /* feedback_state */,
bool sending) {
- MutexLock lock(&mutex_rtcp_sender_);
- sending_ = sending;
+ bool sendRTCPBye = false;
+ {
+ MutexLock lock(&mutex_rtcp_sender_);
+
+ if (method_ != RtcpMode::kOff) {
+ if (sending == false && sending_ == true) {
+ // Trigger RTCP bye
+ sendRTCPBye = true;
+ }
+ }
+ sending_ = sending;
+ }
+ if (sendRTCPBye) {
+ if (SendRTCP(feedback_state, kRtcpBye) != 0) {
+ RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE";
+ }
+ }
}
void RTCPSender::SetNonSenderRttMeasurement(bool enabled) {
diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 9cca326ac5..ede02ea636 100644
--- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -346,12 +346,13 @@ TEST_F(RtcpSenderTest, SendBye) {
EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
-TEST_F(RtcpSenderTest, StopSendingDoesNotTriggersBye) {
+TEST_F(RtcpSenderTest, StopSendingTriggersBye) {
auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
rtcp_sender->SetSendingStatus(feedback_state(), true);
rtcp_sender->SetSendingStatus(feedback_state(), false);
- EXPECT_EQ(0, parser()->bye()->num_packets());
+ EXPECT_EQ(1, parser()->bye()->num_packets());
+ EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
}
TEST_F(RtcpSenderTest, SendFir) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index b4c944ccf8..9b5898b2e7 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -313,6 +313,7 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
if (rtcp_sender_.Sending() != sending) {
+ // Sends RTCP BYE when going from true to false
rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
}
return 0;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 02828619a6..e3dc8d9248 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -278,7 +278,7 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
// Returns the FlexFEC SSRC, if there is one.
virtual std::optional<uint32_t> FlexfecSsrc() const = 0;
- // Sets sending status.
+ // Sets sending status. Sends kRtcpByeCode when going from true to false.
// Returns -1 on failure else 0.
virtual int32_t SetSendingStatus(bool sending) = 0;