зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1766646 - Vendor libwebrtc from 7336422fe3f
Upstream commit: https://webrtc.googlesource.com/src/+/7336422fe3fcbaaefde4ab0c4e370ab399bf6785 Delete some unneeded references to ProcessThread. Bug: None Change-Id: I77528df2a8bd2d461440cf59ada8229e732a1e00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242370 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35613}
This commit is contained in:
Родитель
1228b22e6d
Коммит
8273daa852
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@ -10377,3 +10377,6 @@ c3795ff2161
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# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
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# base of lastest vendoring
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2d319df955f
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# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
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# base of lastest vendoring
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7336422fe3f
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@ -6926,3 +6926,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
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libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-21T18:52:16.648766.
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# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
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libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-21T18:53:59.860941.
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# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
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libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-21T18:54:38.740183.
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@ -83,7 +83,6 @@ rtc_library("audio") {
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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@ -29,9 +29,7 @@
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namespace webrtc {
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class PacketRouter;
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class ProcessThread;
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class RtcEventLog;
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class RtpPacketReceived;
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class RtpStreamReceiverControllerInterface;
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class RtpStreamReceiverInterface;
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@ -39,7 +39,6 @@
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/format_macros.h"
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#include "rtc_base/location.h"
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@ -31,7 +31,6 @@
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#include "modules/audio_processing/rms_level.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/event.h"
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#include "rtc_base/format_macros.h"
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@ -53,9 +53,6 @@ class VoipCore : public VoipEngine,
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public VoipVolumeControl {
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public:
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// Construct VoipCore with provided arguments.
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// ProcessThread implementation can be injected by `process_thread`
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// (mainly for testing purpose) and when set to nullptr, default
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// implementation will be used.
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VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
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std::unique_ptr<TaskQueueFactory> task_queue_factory,
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@ -35,7 +35,6 @@
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#include "call/simulated_network.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/task_queue.h"
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@ -18,7 +18,6 @@
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#include <vector>
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#include "api/media_types.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/clock.h"
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@ -23,7 +23,6 @@
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#include "modules/rtp_rtcp/include/flexfec_receiver.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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@ -18,7 +18,6 @@
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#include "api/transport/bitrate_settings.h"
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#include "api/transport/network_control.h"
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#include "api/transport/webrtc_key_value_config.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/task_queue.h"
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namespace webrtc {
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@ -25,7 +25,6 @@
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/include/video_codec_interface.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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@ -18,7 +18,6 @@
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#include "absl/strings/match.h"
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#include "modules/pacing/bitrate_prober.h"
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#include "modules/pacing/interval_budget.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/logging.h"
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@ -133,7 +133,6 @@ if (!build_with_chromium) {
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"../../test:test_main",
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"../../test:test_support",
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"../../test:video_test_common",
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"../utility",
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"//testing/gtest",
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"//third_party/abseil-cpp/absl/memory",
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]
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@ -21,7 +21,6 @@
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#include "api/video/i420_buffer.h"
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#include "api/video/video_frame.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_capture/video_capture_factory.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/time_utils.h"
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@ -19,7 +19,6 @@
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#include "modules/include/module_common_types.h"
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#include "modules/include/module_common_types_public.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/decoding_state.h"
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#include "modules/video_coding/event_wrapper.h"
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#include "modules/video_coding/include/video_coding.h"
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@ -93,7 +93,6 @@ rtc_library("video") {
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/rtp_rtcp:rtp_video_header",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:codec_globals_headers",
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"../modules/video_coding:nack_requester",
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@ -15,7 +15,6 @@
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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