commit 51ba6835592cf8ac76843ac7e9ebfc897d91e7f1 parent 763222619b9385ade5b94b883b7bb08413b212f1 Author: Michael Froman <mfroman@mozilla.com> Date: Wed, 8 Oct 2025 16:05:37 -0500 Bug 1993083 - Vendor libwebrtc from 9b3d4e8e23 Upstream commit: https://webrtc.googlesource.com/src/+/9b3d4e8e2371c25f9718a344619098d23f738cdd IWYU audio/ Done by grepping for any <st*\.h> and removing them using find audio/ -name "*.h" -o -name "*.cc" | xargs sed -i '/<stdint.h>/d' et al followed by find audio/ -name "*.h" -o -name "*.cc" | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Bug: webrtc:42226242 Change-Id: I0aff75801622d9f3f8106a3bc15191072f73faff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396780 Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45009} Diffstat:
43 files changed, 300 insertions(+), 69 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 /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 2025-10-08T21:04:23.406426+00:00. +libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-08T21:05:27.160993+00:00. # base of lastest vendoring -4a5b9e9078 +9b3d4e8e23 diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn @@ -139,7 +139,9 @@ if (rtc_include_tests) { ":audio", "../api:simulated_network_api", "../api/audio:audio_device", + "../api/audio_codecs:audio_codecs_api", "../api/task_queue", + "../call:call_interfaces", "../call:fake_network", "../modules/audio_device:test_audio_device_module", "../system_wrappers", @@ -177,6 +179,7 @@ if (rtc_include_tests) { "../api:frame_transformer_factory", "../api:frame_transformer_interface", "../api:function_view", + "../api:location", "../api:make_ref_counted", "../api:mock_audio_mixer", "../api:mock_frame_decryptor", @@ -190,6 +193,7 @@ if (rtc_include_tests) { "../api:simulated_network_api", "../api:transport_api", "../api/audio:audio_frame_api", + "../api/audio:audio_frame_processor", "../api/audio:audio_mixer_api", "../api/audio:audio_processing_statistics", "../api/audio_codecs:audio_codecs_api", @@ -217,6 +221,7 @@ if (rtc_include_tests) { "../call:rtp_receiver", "../call:rtp_sender", "../common_audio", + "../modules/async_audio_processing", "../modules/audio_coding:audio_coding_module_typedefs", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", @@ -226,6 +231,7 @@ if (rtc_include_tests) { "../modules/rtp_rtcp", "../modules/rtp_rtcp:mock_rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:logging", "../rtc_base:safe_compare", "../rtc_base:task_queue_for_test", "../rtc_base:threading", @@ -243,6 +249,7 @@ if (rtc_include_tests) { "../test:wait_until", "../test/time_controller", "utility:utility_tests", + "//third_party/abseil-cpp/absl/functional:any_invocable", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings:string_view", ] diff --git a/third_party/libwebrtc/audio/audio_level.cc b/third_party/libwebrtc/audio/audio_level.cc @@ -10,8 +10,11 @@ #include "audio/audio_level.h" +#include <cstdint> + #include "api/audio/audio_frame.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "rtc_base/synchronization/mutex.h" namespace webrtc { namespace voe { diff --git a/third_party/libwebrtc/audio/audio_level.h b/third_party/libwebrtc/audio/audio_level.h @@ -11,6 +11,8 @@ #ifndef AUDIO_AUDIO_LEVEL_H_ #define AUDIO_AUDIO_LEVEL_H_ +#include <cstdint> + #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/audio/audio_receive_stream.cc b/third_party/libwebrtc/audio/audio_receive_stream.cc @@ -25,6 +25,7 @@ #include "api/audio/audio_mixer.h" #include "api/audio_codecs/audio_format.h" #include "api/call/audio_sink.h" +#include "api/crypto/frame_decryptor_interface.h" #include "api/environment/environment.h" #include "api/frame_transformer_interface.h" #include "api/neteq/neteq_factory.h" diff --git a/third_party/libwebrtc/audio/audio_send_stream.h b/third_party/libwebrtc/audio/audio_send_stream.h @@ -11,24 +11,33 @@ #ifndef AUDIO_AUDIO_SEND_STREAM_H_ #define AUDIO_AUDIO_SEND_STREAM_H_ +#include <cstddef> +#include <cstdint> #include <memory> +#include <optional> #include <utility> #include <vector> -#include "absl/functional/any_invocable.h" +#include "api/call/bitrate_allocation.h" #include "api/environment/environment.h" #include "api/field_trials_view.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" -#include "api/task_queue/task_queue_base.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" #include "audio/audio_level.h" #include "audio/channel_send.h" #include "call/audio_send_stream.h" #include "call/audio_state.h" #include "call/bitrate_allocator.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "rtc_base/experiments/struct_parameters_parser.h" #include "rtc_base/race_checker.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { class RtcpRttStats; diff --git a/third_party/libwebrtc/audio/audio_send_stream_tests.cc b/third_party/libwebrtc/audio/audio_send_stream_tests.cc @@ -8,17 +8,24 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <cstddef> +#include <cstdint> #include <string> -#include <utility> #include <vector> +#include "api/array_view.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "rtc_base/logging.h" #include "test/call_test.h" -#include "test/field_trial.h" #include "test/gtest.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/audio_state.cc b/third_party/libwebrtc/audio/audio_state.cc @@ -11,18 +11,26 @@ #include "audio/audio_state.h" #include <algorithm> -#include <memory> +#include <cstddef> +#include <cstdint> #include <utility> #include <vector> #include "api/audio/audio_device.h" +#include "api/audio/audio_device_defines.h" +#include "api/audio/audio_processing.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" -#include "audio/audio_receive_stream.h" #include "audio/audio_send_stream.h" +#include "call/audio_receive_stream.h" +#include "call/audio_sender.h" +#include "call/audio_state.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/task_utils/repeating_task.h" namespace webrtc { namespace internal { diff --git a/third_party/libwebrtc/audio/audio_state.h b/third_party/libwebrtc/audio/audio_state.h @@ -11,14 +11,16 @@ #ifndef AUDIO_AUDIO_STATE_H_ #define AUDIO_AUDIO_STATE_H_ +#include <cstddef> #include <map> -#include <memory> +#include "api/audio/audio_device.h" +#include "api/audio/audio_processing.h" #include "api/sequence_checker.h" #include "audio/audio_transport_impl.h" #include "call/audio_state.h" +#include "rtc_base/checks.h" #include "rtc_base/containers/flat_set.h" -#include "rtc_base/ref_count.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/audio/audio_state_unittest.cc @@ -10,18 +10,35 @@ #include "audio/audio_state.h" +#include <cmath> +#include <cstddef> +#include <cstdint> +#include <cstdlib> #include <memory> #include <numbers> #include <utility> #include <vector> +#include "absl/functional/any_invocable.h" +#include "absl/strings/string_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_frame_processor.h" +#include "api/audio/audio_mixer.h" +#include "api/location.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" +#include "api/task_queue/task_queue_factory.h" #include "api/task_queue/test/mock_task_queue_base.h" +#include "call/audio_state.h" #include "call/test/mock_audio_receive_stream.h" #include "call/test/mock_audio_send_stream.h" +#include "modules/async_audio_processing/async_audio_processing.h" #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "rtc_base/thread.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/audio_transport_impl.h b/third_party/libwebrtc/audio/audio_transport_impl.h @@ -11,10 +11,13 @@ #ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_ #define AUDIO_AUDIO_TRANSPORT_IMPL_H_ +#include <cstddef> +#include <cstdint> #include <memory> +#include <optional> #include <vector> -#include "api/audio/audio_device.h" +#include "api/audio/audio_device_defines.h" #include "api/audio/audio_mixer.h" #include "api/audio/audio_processing.h" #include "api/scoped_refptr.h" diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc @@ -26,7 +26,7 @@ #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/buffer.h" -#include "rtc_base/string_encode.h" +#include "rtc_base/checks.h" #include "system_wrappers/include/ntp_time.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h @@ -11,16 +11,20 @@ #ifndef AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_ #define AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_ +#include <cstdint> +#include <functional> #include <memory> #include <string> +#include "api/array_view.h" #include "api/frame_transformer_interface.h" #include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/units/timestamp.h" #include "rtc_base/system/no_unique_address.h" -#include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate_unittest.cc @@ -22,6 +22,7 @@ #include "api/scoped_refptr.h" #include "api/test/mock_frame_transformer.h" #include "api/test/mock_transformable_audio_frame.h" +#include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/thread.h" #include "system_wrappers/include/ntp_time.h" diff --git a/third_party/libwebrtc/audio/conversion.h b/third_party/libwebrtc/audio/conversion.h @@ -11,8 +11,8 @@ #ifndef AUDIO_CONVERSION_H_ #define AUDIO_CONVERSION_H_ -#include <stddef.h> -#include <stdint.h> +#include <cstddef> +#include <cstdint> namespace webrtc { diff --git a/third_party/libwebrtc/audio/remix_resample.h b/third_party/libwebrtc/audio/remix_resample.h @@ -11,6 +11,8 @@ #ifndef AUDIO_REMIX_RESAMPLE_H_ #define AUDIO_REMIX_RESAMPLE_H_ +#include <cstdint> + #include "api/audio/audio_frame.h" #include "api/audio/audio_view.h" #include "common_audio/resampler/include/push_resampler.h" diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc @@ -10,12 +10,16 @@ #include "audio/test/audio_end_to_end_test.h" -#include <algorithm> +#include <cstddef> #include <memory> +#include <vector> -#include "api/task_queue/task_queue_base.h" -#include "call/fake_network_pipe.h" +#include "api/audio/audio_device.h" +#include "api/audio_codecs/audio_format.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "modules/audio_device/include/test_audio_device.h" +#include "test/call_test.h" #include "test/gtest.h" #include "test/video_test_constants.h" diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.h b/third_party/libwebrtc/audio/test/audio_end_to_end_test.h @@ -10,13 +10,13 @@ #ifndef AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ #define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ +#include <cstddef> #include <memory> -#include <string> #include <vector> #include "api/audio/audio_device.h" -#include "api/task_queue/task_queue_base.h" -#include "api/test/simulated_network.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "modules/audio_device/include/test_audio_device.h" #include "test/call_test.h" diff --git a/third_party/libwebrtc/audio/test/audio_stats_test.cc b/third_party/libwebrtc/audio/test/audio_stats_test.cc @@ -8,9 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <cstdint> +#include <cstdlib> + +#include "api/test/simulated_network.h" #include "audio/test/audio_end_to_end_test.h" -#include "rtc_base/numerics/safe_compare.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "rtc_base/thread.h" +#include "test/call_test.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/test/nack_test.cc b/third_party/libwebrtc/audio/test/nack_test.cc @@ -8,8 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <cstdint> +#include <vector> + +#include "api/test/simulated_network.h" #include "audio/test/audio_end_to_end_test.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "rtc_base/thread.h" +#include "test/call_test.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations.cc @@ -10,12 +10,13 @@ #include "audio/utility/audio_frame_operations.h" -#include <string.h> - #include <algorithm> #include <cstdint> +#include <cstring> #include <utility> +#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "common_audio/include/audio_util.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.h b/third_party/libwebrtc/audio/utility/audio_frame_operations.h @@ -11,12 +11,11 @@ #ifndef AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_ #define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_ -#include <stddef.h> -#include <stdint.h> +#include <cstddef> +#include <cstdint> -#include "absl/base/attributes.h" -#include "api/array_view.h" #include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc @@ -10,6 +10,11 @@ #include "audio/utility/audio_frame_operations.h" +#include <cstddef> +#include <cstdint> + +#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "rtc_base/checks.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.cc b/third_party/libwebrtc/audio/utility/channel_mixer.cc @@ -10,9 +10,14 @@ #include "audio/utility/channel_mixer.h" +#include <cstddef> +#include <cstdint> +#include <cstring> + +#include "api/audio/audio_frame.h" +#include "api/audio/channel_layout.h" #include "audio/utility/channel_mixing_matrix.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.h b/third_party/libwebrtc/audio/utility/channel_mixer.h @@ -11,9 +11,8 @@ #ifndef AUDIO_UTILITY_CHANNEL_MIXER_H_ #define AUDIO_UTILITY_CHANNEL_MIXER_H_ -#include <stddef.h> -#include <stdint.h> - +#include <cstddef> +#include <cstdint> #include <memory> #include <vector> diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc @@ -10,13 +10,13 @@ #include "audio/utility/channel_mixing_matrix.h" -#include <stddef.h> - #include <algorithm> +#include <cstddef> +#include <vector> +#include "api/audio/channel_layout.h" #include "audio/utility/channel_mixer.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc @@ -10,8 +10,7 @@ #include "audio/utility/channel_mixing_matrix.h" -#include <stddef.h> - +#include <cstddef> #include <vector> #include "api/audio/channel_layout.h" diff --git a/third_party/libwebrtc/audio/voip/BUILD.gn b/third_party/libwebrtc/audio/voip/BUILD.gn @@ -16,16 +16,24 @@ rtc_library("voip_core") { deps = [ ":audio_channel", "..:audio", + "../../api:array_view", + "../../api:make_ref_counted", "../../api:scoped_refptr", + "../../api:transport_api", "../../api/audio:audio_device", + "../../api/audio:audio_mixer_api", "../../api/audio:audio_processing", "../../api/audio_codecs:audio_codecs_api", "../../api/environment", "../../api/task_queue", "../../api/voip:voip_api", + "../../call:audio_sender_interface", "../../modules/audio_mixer:audio_mixer_impl", "../../rtc_base:criticalsection", "../../rtc_base:logging", + "../../rtc_base:macromagic", + "../../rtc_base:random", + "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", ] } @@ -38,12 +46,21 @@ rtc_library("audio_channel") { deps = [ ":audio_egress", ":audio_ingress", + "../../api:array_view", + "../../api:ref_count", + "../../api:rtp_headers", + "../../api:scoped_refptr", "../../api:transport_api", + "../../api/audio:audio_mixer_api", "../../api/audio_codecs:audio_codecs_api", + "../../api/environment", "../../api/task_queue", "../../api/voip:voip_api", + "../../call:audio_sender_interface", + "../../modules/audio_coding:audio_coding_module_typedefs", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:rtp_rtcp_format", + "../../rtc_base:checks", "../../rtc_base:criticalsection", "../../rtc_base:logging", "../../rtc_base:refcount", @@ -75,7 +92,9 @@ rtc_library("audio_ingress") { "../../rtc_base:checks", "../../rtc_base:criticalsection", "../../rtc_base:logging", + "../../rtc_base:macromagic", "../../rtc_base:rtc_numerics", + "../../rtc_base:safe_conversions", "../../rtc_base:safe_minmax", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", @@ -90,15 +109,21 @@ rtc_library("audio_egress") { ] deps = [ "..:audio", + "../../api:array_view", "../../api:sequence_checker", + "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", "../../api/environment", "../../api/task_queue", "../../call:audio_sender_interface", "../../modules/audio_coding", + "../../modules/audio_coding:audio_coding_module_typedefs", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:rtp_rtcp_format", + "../../rtc_base:checks", "../../rtc_base:logging", + "../../rtc_base:macromagic", + "../../rtc_base:safe_conversions", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:no_unique_address", diff --git a/third_party/libwebrtc/audio/voip/audio_channel.cc b/third_party/libwebrtc/audio/voip/audio_channel.cc @@ -10,14 +10,25 @@ #include "audio/voip/audio_channel.h" +#include <cstdint> +#include <memory> #include <utility> -#include <vector> -#include "api/audio_codecs/audio_format.h" -#include "api/task_queue/task_queue_factory.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/call/transport.h" +#include "api/environment/environment.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" +#include "api/voip/voip_statistics.h" +#include "audio/voip/audio_egress.h" +#include "audio/voip/audio_ingress.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/rtp_rtcp/include/receive_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" -#include "rtc_base/logging.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/audio_channel.h b/third_party/libwebrtc/audio/voip/audio_channel.h @@ -11,18 +11,29 @@ #ifndef AUDIO_VOIP_AUDIO_CHANNEL_H_ #define AUDIO_VOIP_AUDIO_CHANNEL_H_ +#include <cstdint> #include <map> #include <memory> -#include <queue> +#include <optional> #include <utility> -#include "api/task_queue/task_queue_factory.h" +#include "api/array_view.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/environment/environment.h" +#include "api/ref_count.h" +#include "api/scoped_refptr.h" #include "api/voip/voip_base.h" #include "api/voip/voip_statistics.h" #include "audio/voip/audio_egress.h" #include "audio/voip/audio_ingress.h" +#include "call/audio_sender.h" +#include "modules/rtp_rtcp/include/receive_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" -#include "rtc_base/ref_count.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/audio_egress.cc b/third_party/libwebrtc/audio/voip/audio_egress.cc @@ -10,11 +10,25 @@ #include "audio/voip/audio_egress.h" +#include <cstddef> +#include <cstdint> +#include <memory> #include <utility> -#include <vector> +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/environment/environment.h" #include "api/sequence_checker.h" +#include "api/task_queue/task_queue_factory.h" +#include "audio/utility/audio_frame_operations.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/audio_egress.h b/third_party/libwebrtc/audio/voip/audio_egress.h @@ -11,24 +11,24 @@ #ifndef AUDIO_VOIP_AUDIO_EGRESS_H_ #define AUDIO_VOIP_AUDIO_EGRESS_H_ +#include <cstddef> +#include <cstdint> #include <memory> -#include <string> +#include <optional> #include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" -#include "api/task_queue/task_queue_factory.h" #include "audio/audio_level.h" -#include "audio/utility/audio_frame_operations.h" #include "call/audio_sender.h" #include "modules/audio_coding/include/audio_coding_module.h" -#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "modules/rtp_rtcp/source/rtp_sender_audio.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/no_unique_address.h" -#include "rtc_base/time_utils.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/audio_ingress.h b/third_party/libwebrtc/audio/voip/audio_ingress.h @@ -13,28 +13,30 @@ #include <algorithm> #include <atomic> +#include <cstdint> #include <map> #include <memory> #include <optional> -#include <utility> #include "api/array_view.h" #include "api/audio/audio_mixer.h" #include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/neteq/neteq.h" -#include "api/rtp_headers.h" #include "api/scoped_refptr.h" #include "api/voip/voip_statistics.h" #include "audio/audio_level.h" #include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" -#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/test/BUILD.gn b/third_party/libwebrtc/audio/voip/test/BUILD.gn @@ -17,6 +17,7 @@ if (rtc_include_tests) { "../../../api/task_queue:task_queue", "../../../api/task_queue/test:mock_task_queue_base", "../../../test:test_support", + "//third_party/abseil-cpp/absl/strings:string_view", ] } @@ -26,10 +27,15 @@ if (rtc_include_tests) { sources = [ "voip_core_unittest.cc" ] deps = [ "..:voip_core", + "../../../api:make_ref_counted", + "../../../api:scoped_refptr", + "../../../api/audio:audio_processing", + "../../../api/audio_codecs:audio_codecs_api", "../../../api/audio_codecs:builtin_audio_decoder_factory", "../../../api/audio_codecs:builtin_audio_encoder_factory", "../../../api/environment:environment_factory", "../../../api/task_queue:default_task_queue_factory", + "../../../api/voip:voip_api", "../../../modules/audio_device:mock_audio_device", "../../../modules/audio_processing:mocks", "../../../test:audio_codec_mocks", @@ -77,7 +83,13 @@ if (rtc_include_tests) { deps = [ "..:audio_egress", "..:audio_ingress", + "../../../api:array_view", + "../../../api:rtp_headers", + "../../../api:scoped_refptr", "../../../api:transport_api", + "../../../api/audio:audio_frame_api", + "../../../api/audio:audio_mixer_api", + "../../../api/audio_codecs:audio_codecs_api", "../../../api/audio_codecs:builtin_audio_decoder_factory", "../../../api/audio_codecs:builtin_audio_encoder_factory", "../../../api/environment", @@ -101,7 +113,12 @@ if (rtc_include_tests) { sources = [ "audio_egress_unittest.cc" ] deps = [ "..:audio_egress", + "../../../api:array_view", + "../../../api:rtp_headers", + "../../../api:scoped_refptr", "../../../api:transport_api", + "../../../api/audio:audio_frame_api", + "../../../api/audio_codecs:audio_codecs_api", "../../../api/audio_codecs:builtin_audio_encoder_factory", "../../../api/environment", "../../../api/environment:environment_factory", diff --git a/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc @@ -10,21 +10,31 @@ #include "audio/voip/audio_egress.h" +#include <cstddef> +#include <cstdint> +#include <memory> +#include <optional> + +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/call/transport.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "modules/audio_mixer/sine_wave_generator.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "rtc_base/event.h" -#include "rtc_base/logging.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" -#include "test/run_loop.h" #include "test/time_controller/simulated_time_controller.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc @@ -10,22 +10,33 @@ #include "audio/voip/audio_ingress.h" +#include <cstddef> +#include <cstdint> +#include <memory> + +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" -#include "api/call/transport.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" -#include "api/task_queue/default_task_queue_factory.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "audio/voip/audio_egress.h" #include "modules/audio_mixer/sine_wave_generator.h" +#include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "rtc_base/event.h" -#include "rtc_base/logging.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" -#include "test/run_loop.h" #include "test/time_controller/simulated_time_controller.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/test/mock_task_queue.h b/third_party/libwebrtc/audio/voip/test/mock_task_queue.h @@ -13,9 +13,10 @@ #include <memory> +#include "absl/strings/string_view.h" +#include "api/task_queue/task_queue_base.h" #include "api/task_queue/task_queue_factory.h" #include "api/task_queue/test/mock_task_queue_base.h" -#include "test/gmock.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc b/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc @@ -10,11 +10,21 @@ #include "audio/voip/voip_core.h" +#include <memory> +#include <utility> + +#include "api/audio/audio_processing.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/voip/voip_base.h" +#include "api/voip/voip_dtmf.h" #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_processing/include/mock_audio_processing.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" #include "test/run_loop.h" diff --git a/third_party/libwebrtc/audio/voip/voip_core.cc b/third_party/libwebrtc/audio/voip/voip_core.cc @@ -11,11 +11,36 @@ #include "audio/voip/voip_core.h" #include <algorithm> +#include <cstddef> +#include <cstdint> +#include <map> #include <memory> +#include <optional> #include <utility> +#include <vector> +#include "api/array_view.h" +#include "api/audio/audio_device.h" +#include "api/audio/audio_processing.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_format.h" +#include "api/call/transport.h" +#include "api/environment/environment.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/voip/voip_base.h" +#include "api/voip/voip_dtmf.h" +#include "api/voip/voip_statistics.h" +#include "api/voip/voip_volume_control.h" +#include "audio/audio_transport_impl.h" +#include "audio/voip/audio_channel.h" +#include "call/audio_sender.h" +#include "modules/audio_mixer/audio_mixer_impl.h" #include "rtc_base/logging.h" +#include "rtc_base/random.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/time_utils.h" namespace webrtc { diff --git a/third_party/libwebrtc/audio/voip/voip_core.h b/third_party/libwebrtc/audio/voip/voip_core.h @@ -11,16 +11,19 @@ #ifndef AUDIO_VOIP_VOIP_CORE_H_ #define AUDIO_VOIP_VOIP_CORE_H_ +#include <cstdint> #include <map> #include <memory> -#include <queue> +#include <optional> #include <unordered_map> -#include <vector> +#include "api/array_view.h" #include "api/audio/audio_device.h" +#include "api/audio/audio_mixer.h" #include "api/audio/audio_processing.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/scoped_refptr.h" #include "api/voip/voip_base.h" @@ -32,8 +35,8 @@ #include "api/voip/voip_volume_control.h" #include "audio/audio_transport_impl.h" #include "audio/voip/audio_channel.h" -#include "modules/audio_mixer/audio_mixer_impl.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/moz-patch-stack/s0035.patch b/third_party/libwebrtc/moz-patch-stack/s0035.patch @@ -20,10 +20,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0b311007c033e838 11 files changed, 55 insertions(+), 10 deletions(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc -index 1c4d634729..a748c4f398 100644 +index f703ae0041..3a266c98fa 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc -@@ -57,6 +57,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const { +@@ -58,6 +58,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const { << (rtcp_mode == RtcpMode::kCompound ? "compound" : (rtcp_mode == RtcpMode::kReducedSize ? "reducedSize" : "off")); @@ -32,7 +32,7 @@ index 1c4d634729..a748c4f398 100644 ss << '}'; return ss.str(); } -@@ -90,7 +92,7 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive( +@@ -91,7 +93,7 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive( config.jitter_buffer_min_delay_ms, config.enable_non_sender_rtt, config.decoder_factory, config.codec_pair_id, std::move(config.frame_decryptor), config.crypto_options, diff --git a/third_party/libwebrtc/moz-patch-stack/s0039.patch b/third_party/libwebrtc/moz-patch-stack/s0039.patch @@ -16,10 +16,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c186df8a088e46285 1 file changed, 1 deletion(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc -index a748c4f398..1f2808b922 100644 +index 3a266c98fa..5e9e3ccc2b 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc -@@ -379,7 +379,6 @@ int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const { +@@ -380,7 +380,6 @@ int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const { } std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const { diff --git a/third_party/libwebrtc/moz-patch-stack/s0103.patch b/third_party/libwebrtc/moz-patch-stack/s0103.patch @@ -276,7 +276,7 @@ index 802f8cc436..eb4fbcad65 100644 rtc_library("scalability_mode") { diff --git a/audio/BUILD.gn b/audio/BUILD.gn -index aa82f7a149..f1e014b310 100644 +index 9b287604dc..8f169039f3 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -8,8 +8,8 @@