Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
index 8e1f77a3ff4801950f973ceb4b600f199e38e883..f683f14bd48c375c64bf8309f69e0a86483d20bf 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -97,8 +97,7 @@ class BitrateAggregator { |
uint32_t ssrc_; |
}; |
-RTPSender::RTPSender(int32_t id, |
- bool audio, |
+RTPSender::RTPSender(bool audio, |
Clock* clock, |
Transport* transport, |
RtpAudioFeedback* audio_feedback, |
@@ -115,10 +114,8 @@ RTPSender::RTPSender(int32_t id, |
TickTime::MillisecondTimestamp()), |
bitrates_(new BitrateAggregator(bitrate_callback)), |
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()), |
- id_(id), |
audio_configured_(audio), |
- audio_(audio ? new RTPSenderAudio(id, clock, this, audio_feedback) |
- : nullptr), |
+ audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr), |
video_(audio ? nullptr : new RTPSenderVideo(clock, this)), |
paced_sender_(paced_sender), |
packet_router_(packet_router), |
@@ -740,7 +737,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { |
bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) { |
int bytes_sent = -1; |
if (transport_) { |
- bytes_sent = transport_->SendPacket(id_, packet, size); |
+ bytes_sent = transport_->SendPacket(packet, size); |
} |
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), |
"RTPSender::SendPacketToNetwork", "size", size, "sent", |