Move RtpPacketSender and merge it with RtpPacketPacer.
This interface is intended to only handle packet-sending parts of the
paced sender.
See https://webrtc-review.googlesource.com/c/src/+/145212 for context
Bug: webrtc:10809
Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28662}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 552b987..8eeacda 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -332,7 +332,7 @@
if (allocation_settings_.IncludeAudioInAllocationOnStart(
config_.min_bitrate_bps, config_.max_bitrate_bps, config_.has_dscp,
TransportSeqNumId(config_))) {
- rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
+ rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
rtc::Event thread_sync_event;
worker_queue_->PostTask([&] {
@@ -796,7 +796,7 @@
if (stream->allocation_settings_.IncludeAudioInAllocationOnReconfigure(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.has_dscp, TransportSeqNumId(new_config))) {
- stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
+ stream->rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtc::Event thread_sync_event;
stream->worker_queue_->PostTask([&] {
RTC_DCHECK_RUN_ON(stream->worker_queue_);
@@ -813,7 +813,7 @@
thread_sync_event.Wait(rtc::Event::kForever);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
- stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
+ stream->rtp_transport_->AccountForAudioPacketsInPacedSender(false);
stream->RemoveBitrateObserver();
stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 72eacb3..f00e0dcd 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -365,11 +365,11 @@
TransportSequenceNumberAllocator* seq_num_allocator_ RTC_GUARDED_BY(&crit_);
};
-class RtpPacketSenderProxy : public RtpPacketPacer {
+class RtpPacketSenderProxy : public RtpPacketSender {
public:
RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
- void SetPacketPacer(RtpPacketPacer* rtp_packet_pacer) {
+ void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
RTC_DCHECK(thread_checker_.IsCurrent());
rtc::CritScope lock(&crit_);
rtp_packet_pacer_ = rtp_packet_pacer;
@@ -394,14 +394,10 @@
}
}
- void SetAccountForAudioPackets(bool account_for_audio) override {
- RTC_NOTREACHED();
- }
-
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
- RtpPacketPacer* rtp_packet_pacer_ RTC_GUARDED_BY(&crit_);
+ RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&crit_);
};
class VoERtcpObserver : public RtcpBandwidthObserver {
@@ -1005,7 +1001,7 @@
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- RtpPacketPacer* rtp_packet_pacer = transport->packet_sender();
+ RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
TransportFeedbackObserver* transport_feedback_observer =
transport->transport_feedback_observer();
PacketRouter* packet_router = transport->packet_router();