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Unified Diff: webrtc/audio/audio_receive_stream.cc

Issue 2886993005: Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface. (Closed)
Patch Set: Created 3 years, 7 months ago
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Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index cb90a68a0f72e6898fe11726d31d2b99c9770c98..330682bc9093c7186095aa6ac18a24f7f01d1368 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -20,6 +20,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/call/rtp_transport_controller_receive_interface.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -62,12 +63,12 @@ std::string AudioReceiveStream::Config::ToString() const {
namespace internal {
AudioReceiveStream::AudioReceiveStream(
+ RtpTransportControllerReceiveInterface* transport,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log)
- : config_(config),
- audio_state_(audio_state) {
+ : config_(config), audio_state_(audio_state) {
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
@@ -107,6 +108,10 @@ AudioReceiveStream::AudioReceiveStream(
}
// Configure bandwidth estimation.
channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
+
+ // Register with transport.
+ rtp_transport_receiver_ =
+ transport->CreateReceiver(config_.rtp.remote_ssrc, this);
pthatcher1 2017/05/17 23:24:53 So CreateReceiver really means "add sink with a co
nisse-webrtc 2017/05/18 08:45:42 Yes. The plan is that it should get additional res
}
AudioReceiveStream::~AudioReceiveStream() {
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