Moved congestion controller to task queue.
The goal of this work is to make it easier to experiment with the bandwidth estimation implementation. For this reason network control functionality is moved from SendSideCongestionController(SSCC), PacedSender and BitrateController to the newly created GoogCcNetworkController which implements the newly created NetworkControllerInterface. This allows the implementation to be replaced at runtime in the future. This is the first part of a split of a larger CL, see: https://webrtc-review.googlesource.com/c/src/+/39788/8 For further explanations. Bug: webrtc:8415 Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 Reviewed-on: https://webrtc-review.googlesource.com/43840 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21868}
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@ -98,7 +98,6 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
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int64_t capture_timestamp,
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bool retransmission,
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const PacedPacketInfo& pacing_info) {
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RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
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rtc::CritScope cs(&modules_crit_);
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for (auto* rtp_module : rtp_send_modules_) {
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if (!rtp_module->SendingMedia())
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@ -114,7 +113,6 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
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size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
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const PacedPacketInfo& pacing_info) {
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RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
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size_t total_bytes_sent = 0;
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rtc::CritScope cs(&modules_crit_);
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// Rtp modules are ordered by which stream can most benefit from padding.
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@ -223,7 +221,6 @@ bool PacketRouter::SendRemb(int64_t bitrate_bps,
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}
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bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
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RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
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rtc::CritScope cs(&modules_crit_);
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// Prefer send modules.
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for (auto* rtp_module : rtp_send_modules_) {
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