Index: talk/app/webrtc/webrtcsession.cc |
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc |
index 15ddc28c7f55d9c17f4fb1531421bf128c14f64e..2ab9a1e6969599983724f9ffdce7aebb91b5d418 100644 |
--- a/talk/app/webrtc/webrtcsession.cc |
+++ b/talk/app/webrtc/webrtcsession.cc |
@@ -265,9 +265,9 @@ static void UpdateSessionDescriptionSecurePolicy(cricket::CryptoType type, |
} |
} |
-static bool GetAudioSsrcByTrackId( |
- const SessionDescription* session_description, |
- const std::string& track_id, uint32 *ssrc) { |
+static bool GetAudioSsrcByTrackId(const SessionDescription* session_description, |
+ const std::string& track_id, |
+ uint32_t* ssrc) { |
const cricket::ContentInfo* audio_info = |
cricket::GetFirstAudioContent(session_description); |
if (!audio_info) { |
@@ -289,7 +289,8 @@ static bool GetAudioSsrcByTrackId( |
} |
static bool GetTrackIdBySsrc(const SessionDescription* session_description, |
- uint32 ssrc, std::string* track_id) { |
+ uint32_t ssrc, |
+ std::string* track_id) { |
ASSERT(track_id != NULL); |
const cricket::ContentInfo* audio_info = |
@@ -461,7 +462,7 @@ static void SetOptionFromOptionalConstraint( |
} |
} |
-uint32 ConvertIceTransportTypeToCandidateFilter( |
+uint32_t ConvertIceTransportTypeToCandidateFilter( |
PeerConnectionInterface::IceTransportsType type) { |
switch (type) { |
case PeerConnectionInterface::kNone: |
@@ -1212,13 +1213,15 @@ bool WebRtcSession::SetIceTransports( |
ConvertIceTransportTypeToCandidateFilter(type)); |
} |
-bool WebRtcSession::GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id) { |
+bool WebRtcSession::GetLocalTrackIdBySsrc(uint32_t ssrc, |
+ std::string* track_id) { |
if (!base_local_description()) |
return false; |
return webrtc::GetTrackIdBySsrc(base_local_description(), ssrc, track_id); |
} |
-bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id) { |
+bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32_t ssrc, |
+ std::string* track_id) { |
if (!base_remote_description()) |
return false; |
return webrtc::GetTrackIdBySsrc(base_remote_description(), ssrc, track_id); |
@@ -1230,7 +1233,8 @@ std::string WebRtcSession::BadStateErrMsg(State state) { |
return desc.str(); |
} |
-void WebRtcSession::SetAudioPlayout(uint32 ssrc, bool enable, |
+void WebRtcSession::SetAudioPlayout(uint32_t ssrc, |
+ bool enable, |
cricket::AudioRenderer* renderer) { |
ASSERT(signaling_thread()->IsCurrent()); |
if (!voice_channel_) { |
@@ -1250,7 +1254,8 @@ void WebRtcSession::SetAudioPlayout(uint32 ssrc, bool enable, |
} |
} |
-void WebRtcSession::SetAudioSend(uint32 ssrc, bool enable, |
+void WebRtcSession::SetAudioSend(uint32_t ssrc, |
+ bool enable, |
const cricket::AudioOptions& options, |
cricket::AudioRenderer* renderer) { |
ASSERT(signaling_thread()->IsCurrent()); |
@@ -1263,7 +1268,7 @@ void WebRtcSession::SetAudioSend(uint32 ssrc, bool enable, |
} |
} |
-void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) { |
+void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) { |
ASSERT(signaling_thread()->IsCurrent()); |
ASSERT(volume >= 0 && volume <= 10); |
if (!voice_channel_) { |
@@ -1276,7 +1281,7 @@ void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) { |
} |
} |
-bool WebRtcSession::SetCaptureDevice(uint32 ssrc, |
+bool WebRtcSession::SetCaptureDevice(uint32_t ssrc, |
cricket::VideoCapturer* camera) { |
ASSERT(signaling_thread()->IsCurrent()); |
@@ -1296,7 +1301,7 @@ bool WebRtcSession::SetCaptureDevice(uint32 ssrc, |
return true; |
} |
-void WebRtcSession::SetVideoPlayout(uint32 ssrc, |
+void WebRtcSession::SetVideoPlayout(uint32_t ssrc, |
bool enable, |
cricket::VideoRenderer* renderer) { |
ASSERT(signaling_thread()->IsCurrent()); |
@@ -1312,7 +1317,8 @@ void WebRtcSession::SetVideoPlayout(uint32 ssrc, |
} |
} |
-void WebRtcSession::SetVideoSend(uint32 ssrc, bool enable, |
+void WebRtcSession::SetVideoSend(uint32_t ssrc, |
+ bool enable, |
const cricket::VideoOptions* options) { |
ASSERT(signaling_thread()->IsCurrent()); |
if (!video_channel_) { |
@@ -1333,7 +1339,7 @@ bool WebRtcSession::CanInsertDtmf(const std::string& track_id) { |
LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists."; |
return false; |
} |
- uint32 send_ssrc = 0; |
+ uint32_t send_ssrc = 0; |
// The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc |
// exists. |
if (!GetAudioSsrcByTrackId(base_local_description(), track_id, |
@@ -1351,7 +1357,7 @@ bool WebRtcSession::InsertDtmf(const std::string& track_id, |
LOG(LS_ERROR) << "InsertDtmf: No audio channel exists."; |
return false; |
} |
- uint32 send_ssrc = 0; |
+ uint32_t send_ssrc = 0; |
if (!VERIFY(GetAudioSsrcByTrackId(base_local_description(), |
track_id, &send_ssrc))) { |
LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id; |