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Remove usage of old copy of rtp_packet_sender.h
Bug: chromium:345101934 Change-Id: I9123dbd39f5d1e34dd1874b840ab6f34f34849a7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357863 Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42725}
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parent
90c430cda4
commit
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10 changed files with 8 additions and 31 deletions
call
modules
pacing
rtp_rtcp
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@ -111,6 +111,7 @@ rtc_library("rtp_interfaces") {
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"../api:frame_transformer_interface",
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"../api:network_state_predictor_api",
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"../api:rtp_headers",
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"../api:rtp_packet_sender",
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"../api:rtp_parameters",
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"../api/crypto:options",
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"../api/environment",
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@ -23,6 +23,7 @@
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#include "api/crypto/crypto_options.h"
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#include "api/fec_controller.h"
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#include "api/frame_transformer_interface.h"
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#include "api/rtp_packet_sender.h"
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#include "api/transport/bandwidth_estimation_settings.h"
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#include "api/transport/bitrate_settings.h"
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#include "api/transport/network_control.h"
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@ -31,7 +32,6 @@
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#include "common_video/frame_counts.h"
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#include "modules/rtp_rtcp/include/report_block_data.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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@ -33,6 +33,7 @@ rtc_library("pacing") {
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"../../api:field_trials_view",
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"../../api:field_trials_view",
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"../../api:function_view",
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"../../api:rtp_packet_sender",
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"../../api:sequence_checker",
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"../../api/rtc_event_log",
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"../../api/task_queue:pending_task_safety_flag",
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@ -157,7 +157,7 @@ Several methods are used to gather statistics in pacer state:
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* `ExpectedQueueTime()` total bytes in the queue divided by the send rate.
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[RTPSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h;drc=77ee8542dd35d5143b5788ddf47fb7cdb96eb08e
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[RtpPacketSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/include/rtp_packet_sender.h;drc=ea55b0872f14faab23a4e5dbcb6956369c8ed5dc
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[RtpPacketSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/rtp_packet_sender.h;drc=ea55b0872f14faab23a4e5dbcb6956369c8ed5dc
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[RtpPacketPacer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/rtp_packet_pacer.h;drc=e7bc3a347760023dd4840cf6ebdd1e6c8592f4d7
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[PacketRouter]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/packet_router.h;drc=3d2210876e31d0bb5c7de88b27fd02ceb1f4e03e
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[TaskQueuePacedSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/task_queue_paced_sender.h;drc=5051693ada61bc7b78855c6fb3fa87a0394fa813
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@ -22,6 +22,7 @@
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#include "absl/types/optional.h"
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#include "api/field_trials_view.h"
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#include "api/function_view.h"
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#include "api/rtp_packet_sender.h"
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#include "api/transport/field_trial_based_config.h"
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#include "api/transport/network_types.h"
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#include "api/units/data_size.h"
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@ -30,7 +31,6 @@
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#include "modules/pacing/interval_budget.h"
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#include "modules/pacing/prioritized_packet_queue.h"
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#include "modules/pacing/rtp_packet_pacer.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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@ -16,11 +16,11 @@
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/transport/network_types.h"
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#include "api/units/data_rate.h"
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#include "api/units/data_size.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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namespace webrtc {
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@ -21,8 +21,6 @@ rtc_library("rtp_rtcp_format") {
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"include/rtcp_statistics.h",
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"include/rtp_cvo.h",
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"include/rtp_header_extension_map.h",
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"include/rtp_packet_sender.h", # Kept for compatibility with downstream
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# projects
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"include/rtp_rtcp_defines.h",
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"source/byte_io.h",
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"source/rtcp_packet.h",
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@ -1,23 +0,0 @@
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/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
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#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
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#include <memory>
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#include <vector>
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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// Kept for compatibility with the newer location of the header
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#include "api/rtp_packet_sender.h"
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#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
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@ -19,12 +19,12 @@
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#include "absl/types/optional.h"
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#include "api/field_trials_view.h"
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#include "api/frame_transformer_interface.h"
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#include "api/rtp_packet_sender.h"
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#include "api/scoped_refptr.h"
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#include "api/units/time_delta.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/report_block_data.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
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@ -22,9 +22,9 @@
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#include "api/array_view.h"
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#include "api/call/transport.h"
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#include "api/field_trials_view.h"
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#include "api/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/flexfec_sender.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_history.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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