Bug 1918268 - Vendor libwebrtc from 5b9d4adfc8

Upstream commit: https://webrtc.googlesource.com/src/+/5b9d4adfc88b7df0aea6f2f4d4b884d2eabcd657
    Move rtp_packet_sender.h to api/

    Old copy of the header and some previous usage is kept around
    for compatibility with downstream projects for now.

    Bug: chromium:345101934
    Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
    Commit-Queue: Florent Castelli <orphis@webrtc.org>
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#42681}

third_party/libwebrtc/README.moz-ff-commit

@@ -31827,3 +31827,6 @@ d74d085dc5
 # MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 9b81d2c954
+# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+5b9d4adfc8

third_party/libwebrtc/README.mozilla

@@ -21242,3 +21242,5 @@ libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc
 libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-09-17T15:26:30.625586.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-09-17T15:42:29.369232.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-09-17T15:43:54.124853.

third_party/libwebrtc/api/BUILD.gn

@@ -587,6 +587,11 @@ rtc_source_set("audio_quality_analyzer_api") {
   ]
 }
 
+rtc_library("rtp_packet_sender") {
+  visibility = [ "*" ]
+  sources = [ "rtp_packet_sender.h" ]
+}
+
 rtc_source_set("stats_observer_interface") {
   visibility = [ "*" ]
   testonly = true

third_party/libwebrtc/api/rtp_packet_sender.h

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_RTP_PACKET_SENDER_H_
+#define API_RTP_PACKET_SENDER_H_
+
+#include <memory>
+#include <vector>
+
+namespace webrtc {
+
+class RtpPacketToSend;
+
+class RtpPacketSender {
+ public:
+  virtual ~RtpPacketSender() = default;
+
+  // Insert a set of packets into queue, for eventual transmission. Based on the
+  // type of packets, they will be prioritized and scheduled relative to other
+  // packets and the current target send rate.
+  virtual void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
+
+  // Clear any pending packets with the given SSRC from the queue.
+  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
+  // updated.
+  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_PACKET_SENDER_H_

third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn

@@ -21,7 +21,8 @@ rtc_library("rtp_rtcp_format") {
     "include/rtcp_statistics.h",
     "include/rtp_cvo.h",
     "include/rtp_header_extension_map.h",
-    "include/rtp_packet_sender.h",
+    "include/rtp_packet_sender.h",  # Kept for compatibility with downstream
+                                    # projects
     "include/rtp_rtcp_defines.h",
     "source/byte_io.h",
     "source/rtcp_packet.h",
@@ -114,6 +115,7 @@ rtc_library("rtp_rtcp_format") {
     "../../api:function_view",
     "../../api:refcountedbase",
     "../../api:rtp_headers",
+    "../../api:rtp_packet_sender",  # For compatibility with downstream projects
     "../../api:rtp_parameters",
     "../../api:scoped_refptr",
     "../../api/audio_codecs:audio_codecs_api",
@@ -281,6 +283,7 @@ rtc_library("rtp_rtcp") {
     "../../api:function_view",
     "../../api:rtp_headers",
     "../../api:rtp_packet_info",
+    "../../api:rtp_packet_sender",
     "../../api:rtp_parameters",
     "../../api:scoped_refptr",
     "../../api:sequence_checker",
@@ -671,6 +674,7 @@ if (rtc_include_tests) {
       "../../api:mock_transformable_video_frame",
       "../../api:rtp_headers",
       "../../api:rtp_packet_info",
+      "../../api:rtp_packet_sender",
       "../../api:rtp_parameters",
       "../../api:scoped_refptr",
       "../../api:time_controller",

third_party/libwebrtc/modules/rtp_rtcp/include/rtp_packet_sender.h

@@ -17,24 +17,7 @@
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
-namespace webrtc {
-
-class RtpPacketSender {
- public:
-  virtual ~RtpPacketSender() = default;
-
-  // Insert a set of packets into queue, for eventual transmission. Based on the
-  // type of packets, they will be prioritized and scheduled relative to other
-  // packets and the current target send rate.
-  virtual void EnqueuePackets(
-      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
-
-  // Clear any pending packets with the given SSRC from the queue.
-  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
-  // updated.
-  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
-};
-
-}  // namespace webrtc
+// Kept for compatibility with the newer location of the header
+#include "api/rtp_packet_sender.h"
 
 #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_

third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -15,6 +15,7 @@
 
 #include "absl/strings/string_view.h"
 #include "api/rtc_event_log/rtc_event.h"
+#include "api/rtp_packet_sender.h"
 #include "api/units/frequency.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -23,7 +24,6 @@
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/packet_sequencer.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"

third_party/libwebrtc/moz-patch-stack/0032.patch

@@ -206,7 +206,7 @@ index 3c78142a25..dbc03c9f8b 100644
      } else {
        deps += [
 diff --git a/api/BUILD.gn b/api/BUILD.gn
-index e537068ab5..2710b2476e 100644
+index 074c5a398a..d4b3cd64e8 100644
 --- a/api/BUILD.gn
 +++ b/api/BUILD.gn
 @@ -44,6 +44,9 @@ rtc_source_set("enable_media") {
@@ -298,7 +298,7 @@ index e537068ab5..2710b2476e 100644
  
  rtc_source_set("frame_transformer_interface") {
    visibility = [ "*" ]
-@@ -580,6 +609,7 @@ rtc_source_set("peer_network_dependencies") {
+@@ -585,6 +614,7 @@ rtc_source_set("peer_network_dependencies") {
  }
  
  rtc_source_set("peer_connection_quality_test_fixture_api") {
@@ -306,7 +306,7 @@ index e537068ab5..2710b2476e 100644
    visibility = [ "*" ]
    testonly = true
    sources = [ "test/peerconnection_quality_test_fixture.h" ]
-@@ -626,6 +656,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
+@@ -631,6 +661,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
      "//third_party/abseil-cpp/absl/types:optional",
    ]
  }
@@ -314,7 +314,7 @@ index e537068ab5..2710b2476e 100644
  
  rtc_source_set("frame_generator_api") {
    visibility = [ "*" ]
-@@ -749,6 +780,7 @@ rtc_library("create_frame_generator") {
+@@ -754,6 +785,7 @@ rtc_library("create_frame_generator") {
    ]
  }
  
@@ -322,7 +322,7 @@ index e537068ab5..2710b2476e 100644
  rtc_library("create_peer_connection_quality_test_frame_generator") {
    visibility = [ "*" ]
    testonly = true
-@@ -765,6 +797,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
+@@ -770,6 +802,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
      "//third_party/abseil-cpp/absl/types:optional",
    ]
  }
@@ -330,7 +330,7 @@ index e537068ab5..2710b2476e 100644
  
  rtc_source_set("libjingle_logging_api") {
    visibility = [ "*" ]
-@@ -945,6 +978,7 @@ rtc_source_set("refcountedbase") {
+@@ -950,6 +983,7 @@ rtc_source_set("refcountedbase") {
    ]
  }
  
@@ -338,7 +338,7 @@ index e537068ab5..2710b2476e 100644
  rtc_library("ice_transport_factory") {
    visibility = [ "*" ]
    sources = [
-@@ -970,6 +1004,7 @@ rtc_library("ice_transport_factory") {
+@@ -975,6 +1009,7 @@ rtc_library("ice_transport_factory") {
      "rtc_event_log:rtc_event_log",
    ]
  }

third_party/libwebrtc/test/peer_scenario/tests/bwe_ramp_up_test.cc

@@ -16,6 +16,7 @@
 #include "api/units/time_delta.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "pc/media_session.h"
 #include "pc/test/mock_peer_connection_observers.h"

third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc

@@ -10,11 +10,9 @@
 
 #include "media/base/stream_params.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
-#include "pc/media_session.h"
 #include "pc/session_description.h"
-#include "test/field_trial.h"
-#include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/peer_scenario/peer_scenario.h"