Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(219)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index ea7931fadcd729c1ed149fe05a1f2f483443b00c..8b7bbb3d023cf66dfc3ac925d6cb560390e6481e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -132,13 +132,11 @@ class RTCPSender::PacketBuiltCallback
};
RTCPSender::RTCPSender(
- int32_t id,
bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics,
RtcpPacketTypeCounterObserver* packet_type_counter_observer)
- : id_(id),
- audio_(audio),
+ : audio_(audio),
clock_(clock),
method_(kRtcpOff),
critical_section_transport_(
@@ -1119,7 +1117,7 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
CriticalSectionScoped lock(critical_section_transport_.get());
if (cbTransport_) {
- if (cbTransport_->SendRTCPPacket(id_, dataBuffer, length) > 0)
+ if (cbTransport_->SendRTCPPacket(dataBuffer, length) > 0)
return 0;
}
return -1;
@@ -1218,18 +1216,17 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
public:
- Sender(Transport* transport, int32_t id)
- : transport_(transport), id_(id), send_failure_(false) {}
+ Sender(Transport* transport)
+ : transport_(transport), send_failure_(false) {}
void OnPacketReady(uint8_t* data, size_t length) override {
- if (transport_->SendRTCPPacket(id_, data, length) <= 0)
+ if (transport_->SendRTCPPacket(data, length) <= 0)
send_failure_ = true;
}
Transport* const transport_;
- int32_t id_;
bool send_failure_;
- } sender(cbTransport_, id_);
+ } sender(cbTransport_);
uint8_t buffer[IP_PACKET_SIZE];
return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender.h ('k') | webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698