Bug 1828517 - Vendor libwebrtc from 73e0cc8969

Upstream commit: https://webrtc.googlesource.com/src/+/73e0cc8969510b224f17cf8afa3abcd874143640
    Delete unused Audio Bwe integration test.

    Bug: none
    Change-Id: Id8eb4ad4630820441d18e8d1449f4a8d87da9a59
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291335
    Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
    Commit-Queue: Per Kjellander <perkj@webrtc.org>
    Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#39202}

third_party/libwebrtc/BUILD.gn

@@ -740,7 +740,6 @@ if (rtc_include_tests && !build_with_chromium) {
   rtc_test("webrtc_perf_tests") {
     testonly = true
     deps = [
-      "audio:audio_perf_tests",
       "call:call_perf_tests",
       "modules/audio_coding:audio_coding_perf_tests",
       "modules/audio_processing:audio_processing_perf_tests",

third_party/libwebrtc/README.moz-ff-commit

@@ -21222,3 +21222,6 @@ e15b9ff408
 # MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 cfbb247f6d
+# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+73e0cc8969

third_party/libwebrtc/README.mozilla

@@ -14170,3 +14170,5 @@ libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-l
 libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-24T22:45:23.860631.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-24T22:46:47.931784.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-24T22:47:43.571151.

third_party/libwebrtc/audio/BUILD.gn

@@ -331,32 +331,4 @@ if (rtc_include_tests) {
       }
     }
   }
-
-  if (!build_with_chromium) {
-    rtc_library("audio_perf_tests") {
-      testonly = true
-
-      sources = [
-        "test/audio_bwe_integration_test.cc",
-        "test/audio_bwe_integration_test.h",
-      ]
-      deps = [
-        "../api:simulated_network_api",
-        "../api/task_queue",
-        "../call:fake_network",
-        "../call:simulated_network",
-        "../common_audio",
-        "../rtc_base:task_queue_for_test",
-        "../system_wrappers",
-        "../test:field_trial",
-        "../test:fileutils",
-        "../test:test_common",
-        "../test:test_main",
-        "../test:test_support",
-        "//testing/gtest",
-      ]
-      absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
-      data = [ "//resources/voice_engine/audio_dtx16.wav" ]
-    }
-  }
 }

third_party/libwebrtc/audio/test/audio_bwe_integration_test.cc

@@ -1,140 +0,0 @@
-/*
- *  Copyright (c) 2017 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.
- */
-
-#include "audio/test/audio_bwe_integration_test.h"
-
-#include <memory>
-
-#include "absl/functional/any_invocable.h"
-#include "api/task_queue/task_queue_base.h"
-#include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
-#include "common_audio/wav_file.h"
-#include "rtc_base/task_queue_for_test.h"
-#include "system_wrappers/include/sleep.h"
-#include "test/field_trial.h"
-#include "test/gtest.h"
-#include "test/testsupport/file_utils.h"
-
-namespace webrtc {
-namespace test {
-
-namespace {
-enum : int {  // The first valid value is 1.
-  kTransportSequenceNumberExtensionId = 1,
-};
-
-// Wait a second between stopping sending and stopping receiving audio.
-constexpr int kExtraProcessTimeMs = 1000;
-}  // namespace
-
-AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeout) {}
-
-size_t AudioBweTest::GetNumVideoStreams() const {
-  return 0;
-}
-size_t AudioBweTest::GetNumAudioStreams() const {
-  return 1;
-}
-size_t AudioBweTest::GetNumFlexfecStreams() const {
-  return 0;
-}
-
-std::unique_ptr<TestAudioDeviceModule::Capturer>
-AudioBweTest::CreateCapturer() {
-  return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
-}
-
-void AudioBweTest::OnFakeAudioDevicesCreated(
-    TestAudioDeviceModule* send_audio_device,
-    TestAudioDeviceModule* recv_audio_device) {
-  send_audio_device_ = send_audio_device;
-}
-
-void AudioBweTest::PerformTest() {
-  send_audio_device_->WaitForRecordingEnd();
-  SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraProcessTimeMs);
-}
-
-absl::AnyInvocable<void() &&> StatsPollTask(Call* sender_call) {
-  RTC_CHECK(sender_call);
-  return [sender_call] {
-    Call::Stats call_stats = sender_call->GetStats();
-    EXPECT_GT(call_stats.send_bandwidth_bps, 25000);
-    TaskQueueBase::Current()->PostDelayedTask(StatsPollTask(sender_call),
-                                              TimeDelta::Millis(100));
-  };
-}
-
-class NoBandwidthDropAfterDtx : public AudioBweTest {
- public:
-  NoBandwidthDropAfterDtx()
-      : sender_call_(nullptr), stats_poller_("stats poller task queue") {}
-
-  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
-                          std::vector<AudioReceiveStreamInterface::Config>*
-                              receive_configs) override {
-    send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
-        test::CallTest::kAudioSendPayloadType,
-        {"OPUS",
-         48000,
-         2,
-         {{"ptime", "60"}, {"usedtx", "1"}, {"stereo", "1"}}});
-
-    send_config->min_bitrate_bps = 6000;
-    send_config->max_bitrate_bps = 100000;
-    send_config->rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                     kTransportSequenceNumberExtensionId));
-    for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) {
-      recv_config.rtp.extensions = send_config->rtp.extensions;
-      recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
-    }
-  }
-
-  std::string AudioInputFile() override {
-    return test::ResourcePath("voice_engine/audio_dtx16", "wav");
-  }
-
-  BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() override {
-    BuiltInNetworkBehaviorConfig pipe_config;
-    pipe_config.link_capacity_kbps = 50;
-    pipe_config.queue_length_packets = 1500;
-    pipe_config.queue_delay_ms = 300;
-    return pipe_config;
-  }
-
-  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
-    sender_call_ = sender_call;
-  }
-
-  void PerformTest() override {
-    stats_poller_.PostDelayedTask(StatsPollTask(sender_call_),
-                                  TimeDelta::Millis(100));
-    sender_call_->OnAudioTransportOverheadChanged(0);
-    AudioBweTest::PerformTest();
-  }
-
- private:
-  Call* sender_call_;
-  TaskQueueForTest stats_poller_;
-};
-
-using AudioBweIntegrationTest = CallTest;
-
-// TODO(tschumim): This test is flaky when run on android and mac. Re-enable the
-// test for when the issue is fixed.
-TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) {
-  NoBandwidthDropAfterDtx test;
-  RunBaseTest(&test);
-}
-
-}  // namespace test
-}  // namespace webrtc

third_party/libwebrtc/audio/test/audio_bwe_integration_test.h

@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2017 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 AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
-#define AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
-
-#include <memory>
-#include <string>
-
-#include "api/task_queue/task_queue_base.h"
-#include "api/test/simulated_network.h"
-#include "test/call_test.h"
-
-namespace webrtc {
-namespace test {
-
-class AudioBweTest : public test::EndToEndTest {
- public:
-  AudioBweTest();
-
- protected:
-  virtual std::string AudioInputFile() = 0;
-
-  virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() = 0;
-
-  size_t GetNumVideoStreams() const override;
-  size_t GetNumAudioStreams() const override;
-  size_t GetNumFlexfecStreams() const override;
-
-  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
-
-  void OnFakeAudioDevicesCreated(
-      TestAudioDeviceModule* send_audio_device,
-      TestAudioDeviceModule* recv_audio_device) override;
-
-  void PerformTest() override;
-
- private:
-  TestAudioDeviceModule* send_audio_device_;
-};
-
-}  // namespace test
-}  // namespace webrtc
-
-#endif  // AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_

third_party/libwebrtc/moz-patch-stack/0036.patch

@@ -44,7 +44,7 @@ index c9824916ad..d35ea79ced 100644
    # TODO(https://bugs.webrtc.org/14437):  Remove this section if general
    # Chromium fix resolves the problem.
 diff --git a/BUILD.gn b/BUILD.gn
-index c41059f288..02e9d1d4a0 100644
+index 5817d22227..8b2648a306 100644
 --- a/BUILD.gn
 +++ b/BUILD.gn
 @@ -33,7 +33,7 @@ if (is_android) {

third_party/libwebrtc/moz-patch-stack/0048.patch

@@ -20,7 +20,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/58f47eacaf10d12e2
  11 files changed, 27 insertions(+), 27 deletions(-)
 
 diff --git a/BUILD.gn b/BUILD.gn
-index 02e9d1d4a0..4e2a1f1e70 100644
+index 8b2648a306..f32e632ff6 100644
 --- a/BUILD.gn
 +++ b/BUILD.gn
 @@ -407,12 +407,12 @@ config("common_config") {

third_party/libwebrtc/moz-patch-stack/0060.patch

@@ -18,7 +18,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0300b32b7de70fb89
  5 files changed, 10 insertions(+), 3 deletions(-)
 
 diff --git a/BUILD.gn b/BUILD.gn
-index 4e2a1f1e70..03880dbb09 100644
+index f32e632ff6..0272f6a8fe 100644
 --- a/BUILD.gn
 +++ b/BUILD.gn
 @@ -215,6 +215,9 @@ config("common_inherited_config") {

third_party/libwebrtc/moz-patch-stack/0075.patch

@@ -11,7 +11,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b0658888969395dca
  2 files changed, 10 insertions(+)
 
 diff --git a/BUILD.gn b/BUILD.gn
-index 03880dbb09..873aec3850 100644
+index 0272f6a8fe..6515866c2d 100644
 --- a/BUILD.gn
 +++ b/BUILD.gn
 @@ -559,6 +559,10 @@ if (!build_with_chromium) {

third_party/libwebrtc/moz-patch-stack/0084.patch

@@ -14,7 +14,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d
  6 files changed, 38 insertions(+), 3 deletions(-)
 
 diff --git a/BUILD.gn b/BUILD.gn
-index 873aec3850..75e536e046 100644
+index 6515866c2d..465c4d9bfd 100644
 --- a/BUILD.gn
 +++ b/BUILD.gn
 @@ -549,6 +549,15 @@ if (!build_with_chromium) {