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commit 61b89b55c6783b0d636ea2cc05e97025acb4dfbe
parent 951d0e29e428397ec33624c33359e5328a944c6f
Author: Dan Baker <dbaker@mozilla.com>
Date:   Mon, 27 Oct 2025 16:52:06 -0600

Bug 1995393 - Vendor libwebrtc from e30e9adc16

Upstream commit: https://webrtc.googlesource.com/src/+/e30e9adc16ccc5b41bdaf7c6eaf67db12230e4c3
    Delete deprecated variant of the AudioEncoder::EnableAudioNetworkAdaptor

    Bug: None
    Change-Id: I9e85f952ef094fb9d7fc17e5b86dfe9c9dfd2eeb
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406765
    Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
    Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
    Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#45505}

Diffstat:
Mthird_party/libwebrtc/README.mozilla.last-vendor | 4++--
Mthird_party/libwebrtc/api/audio_codecs/audio_encoder.cc | 6------
Mthird_party/libwebrtc/api/audio_codecs/audio_encoder.h | 6------
Mthird_party/libwebrtc/audio/audio_send_stream.cc | 16++--------------
Mthird_party/libwebrtc/audio/audio_send_stream_unittest.cc | 6+++---
Mthird_party/libwebrtc/moz-patch-stack/s0027.patch | 6+++---
Mthird_party/libwebrtc/moz-patch-stack/s0029.patch | 2+-
Mthird_party/libwebrtc/moz-patch-stack/s0099.patch | 2+-
Mthird_party/libwebrtc/moz-patch-stack/s0102.patch | 2+-
Mthird_party/libwebrtc/test/BUILD.gn | 1+
Mthird_party/libwebrtc/test/mock_audio_encoder.h | 4++--
11 files changed, 16 insertions(+), 39 deletions(-)

diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor @@ -1,4 +1,4 @@ # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc -libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T22:49:49.261284+00:00. +libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T22:51:54.552861+00:00. # base of lastest vendoring -24c9f61399 +e30e9adc16 diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc @@ -14,7 +14,6 @@ #include <cstdint> #include <memory> #include <optional> -#include <string> #include "absl/strings/string_view.h" #include "api/array_view.h" @@ -83,11 +82,6 @@ AudioEncoder::ReclaimContainedEncoders() { return nullptr; } -bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, - RtcEventLog* /* event_log */) { - return EnableAudioNetworkAdaptor(config_string); -} - bool AudioEncoder::EnableAudioNetworkAdaptor(absl::string_view /*config*/) { return false; } diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h @@ -16,7 +16,6 @@ #include <memory> #include <optional> -#include <string> #include <utility> #include <vector> @@ -31,8 +30,6 @@ namespace webrtc { -class RtcEventLog; - // Statistics related to Audio Network Adaptation. struct ANAStats { ANAStats(); @@ -202,9 +199,6 @@ class AudioEncoder { virtual ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders(); // Enables audio network adaptor. Returns true if successful. - [[deprecated]] - virtual bool EnableAudioNetworkAdaptor(const std::string& config_string, - RtcEventLog* event_log); virtual bool EnableAudioNetworkAdaptor(absl::string_view config); // Disables audio network adaptor. diff --git a/third_party/libwebrtc/audio/audio_send_stream.cc b/third_party/libwebrtc/audio/audio_send_stream.cc @@ -607,14 +607,8 @@ bool AudioSendStream::SetupSendCodec(const Config& new_config) { // Enable ANA if configured (currently only used by Opus). if (new_config.audio_network_adaptor_config) { -// TODO: bugs.webrtc.org/42223992 - call non-deprecated variant of the -// `EnableAudioNetworkAdaptor` when deprecated one is removed from the -// interface. -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" if (encoder->EnableAudioNetworkAdaptor( - *new_config.audio_network_adaptor_config, &env_.event_log())) { -#pragma clang diagnostic pop + *new_config.audio_network_adaptor_config)) { RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC " << new_config.rtp.ssrc; } else { @@ -714,14 +708,8 @@ void AudioSendStream::ReconfigureANA(const Config& new_config) { if (new_config.audio_network_adaptor_config) { channel_send_->CallEncoder([&](AudioEncoder* encoder) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); -// TODO: bugs.webrtc.org/42223992 - call non-deprecated variant of the -// `EnableAudioNetworkAdaptor` when deprecated one is removed from the -// interface. -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" if (encoder->EnableAudioNetworkAdaptor( - *new_config.audio_network_adaptor_config, &env_.event_log())) { -#pragma clang diagnostic pop + *new_config.audio_network_adaptor_config)) { RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC " << new_config.rtp.ssrc; if (overhead_per_packet_ > 0) { diff --git a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc @@ -548,10 +548,10 @@ TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) { const SdpAudioFormat& format) { auto mock_encoder = SetupAudioEncoderMock(format); EXPECT_CALL(*mock_encoder, - EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _)) + EnableAudioNetworkAdaptor(StrEq(kAnaConfigString))) .WillOnce(Return(true)); EXPECT_CALL(*mock_encoder, - EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _)) + EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString))) .WillOnce(Return(true)); return mock_encoder; })); @@ -580,7 +580,7 @@ TEST(AudioSendStreamTest, AudioNetworkAdaptorReceivesOverhead) { *mock_encoder, OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>()))); EXPECT_CALL(*mock_encoder, - EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _)) + EnableAudioNetworkAdaptor(StrEq(kAnaConfigString))) .WillOnce(Return(true)); // Note: Overhead is received AFTER ANA has been enabled. EXPECT_CALL( diff --git a/third_party/libwebrtc/moz-patch-stack/s0027.patch b/third_party/libwebrtc/moz-patch-stack/s0027.patch @@ -1199,7 +1199,7 @@ index c1181618e9..4a772795ed 100644 deps += [ "..:logging", diff --git a/test/BUILD.gn b/test/BUILD.gn -index d0a12dc0f7..9ce787a429 100644 +index 7651da3c90..5d39040dd8 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -279,6 +279,7 @@ rtc_library("audio_test_common") { @@ -1241,7 +1241,7 @@ index d0a12dc0f7..9ce787a429 100644 sources += [ "testsupport/jpeg_frame_writer.cc" ] } else { sources += [ "testsupport/jpeg_frame_writer_ios.cc" ] -@@ -1359,6 +1367,7 @@ if (!build_with_chromium) { +@@ -1360,6 +1368,7 @@ if (!build_with_chromium) { } } @@ -1249,7 +1249,7 @@ index d0a12dc0f7..9ce787a429 100644 if (!build_with_chromium && is_android) { rtc_android_library("native_test_java") { testonly = true -@@ -1401,6 +1410,7 @@ if (!build_with_chromium && is_android) { +@@ -1402,6 +1411,7 @@ if (!build_with_chromium && is_android) { sources = [ "android/org/webrtc/native_test/NativeTestWebrtc.java" ] } } diff --git a/third_party/libwebrtc/moz-patch-stack/s0029.patch b/third_party/libwebrtc/moz-patch-stack/s0029.patch @@ -15,7 +15,7 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc 4 files changed, 36 insertions(+) diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc -index ee85052333..d7028b7116 100644 +index 4710362715..a56b45d405 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -437,6 +437,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats( diff --git a/third_party/libwebrtc/moz-patch-stack/s0099.patch b/third_party/libwebrtc/moz-patch-stack/s0099.patch @@ -86,7 +86,7 @@ index ef363d15a6..90389a59fc 100644 if (!build_with_mozilla) { diff --git a/test/BUILD.gn b/test/BUILD.gn -index 9ce787a429..90a398fff3 100644 +index 5d39040dd8..9981ae4d80 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -491,6 +491,12 @@ rtc_source_set("test_support") { diff --git a/third_party/libwebrtc/moz-patch-stack/s0102.patch b/third_party/libwebrtc/moz-patch-stack/s0102.patch @@ -756,7 +756,7 @@ index 822b565610..f2ea548824 100644 import("../webrtc.gni") diff --git a/test/BUILD.gn b/test/BUILD.gn -index 90a398fff3..685e1ac082 100644 +index 9981ae4d80..9abf24e919 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -6,10 +6,10 @@ diff --git a/third_party/libwebrtc/test/BUILD.gn b/third_party/libwebrtc/test/BUILD.gn @@ -1244,6 +1244,7 @@ rtc_library("audio_codec_mocks") { "../api/units:time_delta", "../rtc_base:buffer", "../rtc_base:checks", + "//third_party/abseil-cpp/absl/strings:string_view", ] } diff --git a/third_party/libwebrtc/test/mock_audio_encoder.h b/third_party/libwebrtc/test/mock_audio_encoder.h @@ -14,9 +14,9 @@ #include <cstddef> #include <cstdint> #include <optional> -#include <string> #include <utility> +#include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/units/data_rate.h" @@ -66,7 +66,7 @@ class MockAudioEncoder : public AudioEncoder { MOCK_METHOD(bool, EnableAudioNetworkAdaptor, - (const std::string& config_string, RtcEventLog*), + (absl::string_view config_string), (override)); // Note, we explicitly chose not to create a mock for the Encode method.