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Unified Diff: talk/app/webrtc/peerconnection.cc

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase + revert basictypes.h (to be landed separately just in case of a revert due to unexpected us… Created 5 years, 2 months ago
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Index: talk/app/webrtc/peerconnection.cc
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index bf9a80d9b7199ccab41c5503e0ae26be9d000d65..86902b03591277b91d4a6106b38e7caf8e04d495 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -862,13 +862,13 @@ void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) {
void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
- uint32 ssrc) {
+ uint32_t ssrc) {
receivers_.push_back(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
}
void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream,
VideoTrackInterface* video_track,
- uint32 ssrc) {
+ uint32_t ssrc) {
receivers_.push_back(new VideoRtpReceiver(video_track, ssrc, session_.get()));
}
@@ -902,14 +902,14 @@ void PeerConnection::OnRemoveRemoteVideoTrack(
void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
- uint32 ssrc) {
+ uint32_t ssrc) {
senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
stats_->AddLocalAudioTrack(audio_track, ssrc);
}
void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
VideoTrackInterface* video_track,
- uint32 ssrc) {
+ uint32_t ssrc) {
senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
}
@@ -917,7 +917,7 @@ void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
// description.
void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
- uint32 ssrc) {
+ uint32_t ssrc) {
auto it = FindSenderForTrack(audio_track);
if (it == senders_.end()) {
LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
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