Use suffixed {uint,int}{8,16,32,64}_t types.

talk/media/webrtc/webrtcvoiceengine.h

Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 5bf18998e1ea79e4f428d032e3b39151a8f2f6af..33f5ed716ccaa5292a63237763b20110e0cee794 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -121,10 +121,10 @@ class WebRtcVoiceEngine
 
   void SetLogging(int min_sev, const char* filter);
 
-  bool RegisterProcessor(uint32 ssrc,
+  bool RegisterProcessor(uint32_t ssrc,
                          VoiceProcessor* voice_processor,
                          MediaProcessorDirection direction);
-  bool UnregisterProcessor(uint32 ssrc,
+  bool UnregisterProcessor(uint32_t ssrc,
                            VoiceProcessor* voice_processor,
                            MediaProcessorDirection direction);
 
@@ -163,8 +163,8 @@ class WebRtcVoiceEngine
 
  private:
   typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList;
-  typedef sigslot::
-      signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
+  typedef sigslot::signal3<uint32_t, MediaProcessorDirection, AudioFrame*>
+      FrameSignal;
 
   void Construct();
   void ConstructCodecs();
@@ -203,8 +203,8 @@ class WebRtcVoiceEngine
       bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
   bool FindChannelAndSsrc(int channel_num,
                           WebRtcVoiceMediaChannel** channel,
-                          uint32* ssrc) const;
-  bool FindChannelNumFromSsrc(uint32 ssrc,
+                          uint32_t* ssrc) const;
+  bool FindChannelNumFromSsrc(uint32_t ssrc,
                               MediaProcessorDirection direction,
                               int* channel_num);
   bool ChangeLocalMonitor(bool enable);
@@ -212,7 +212,7 @@ class WebRtcVoiceEngine
   bool ResumeLocalMonitor();
 
   bool UnregisterProcessorChannel(MediaProcessorDirection channel_direction,
-                                  uint32 ssrc,
+                                  uint32_t ssrc,
                                   VoiceProcessor* voice_processor,
                                   MediaProcessorDirection processor_direction);
 
@@ -263,8 +263,8 @@ class WebRtcVoiceEngine
   // This is necessary because the lookup results in mux_channels_cs lock being
   // held and if a remote participant leaves the hangout at the same time
   // we hit a deadlock.
-  uint32 tx_processor_ssrc_;
-  uint32 rx_processor_ssrc_;
+  uint32_t tx_processor_ssrc_;
+  uint32_t rx_processor_ssrc_;
 
   rtc::CriticalSection signal_media_critical_;
 
@@ -299,13 +299,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
-  bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+  bool SetAudioSend(uint32_t ssrc,
+                    bool mute,
+                    const AudioOptions* options,
                     AudioRenderer* renderer) override;
   bool AddSendStream(const StreamParams& sp) override;
-  bool RemoveSendStream(uint32 ssrc) override;
+  bool RemoveSendStream(uint32_t ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
-  bool RemoveRecvStream(uint32 ssrc) override;
-  bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
+  bool RemoveRecvStream(uint32_t ssrc) override;
+  bool SetRemoteRenderer(uint32_t ssrc, AudioRenderer* renderer) override;
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
   int GetTimeSinceLastTyping() override;
@@ -314,12 +316,12 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
                                     int reporting_threshold,
                                     int penalty_decay,
                                     int type_event_delay) override;
-  bool SetOutputScaling(uint32 ssrc, double left, double right) override;
+  bool SetOutputScaling(uint32_t ssrc, double left, double right) override;
 
   bool SetRingbackTone(const char* buf, int len) override;
-  bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) override;
+  bool PlayRingbackTone(uint32_t ssrc, bool play, bool loop) override;
   bool CanInsertDtmf() override;
-  bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
+  bool InsertDtmf(uint32_t ssrc, int event, int duration, int flags) override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
@@ -329,7 +331,7 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool GetStats(VoiceMediaInfo* info) override;
   // Gets last reported error from WebRtc voice engine.  This should be only
   // called in response a failure.
-  void GetLastMediaError(uint32* ssrc,
+  void GetLastMediaError(uint32_t* ssrc,
                          VoiceMediaChannel::Error* error) override;
 
   // implements Transport interface
@@ -345,12 +347,12 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
     return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
   }
 
-  bool FindSsrc(int channel_num, uint32* ssrc);
-  void OnError(uint32 ssrc, int error);
+  bool FindSsrc(int channel_num, uint32_t* ssrc);
+  void OnError(uint32_t ssrc, int error);
 
   bool sending() const { return send_ != SEND_NOTHING; }
-  int GetReceiveChannelNum(uint32 ssrc) const;
-  int GetSendChannelNum(uint32 ssrc) const;
+  int GetReceiveChannelNum(uint32_t ssrc) const;
+  int GetSendChannelNum(uint32_t ssrc) const;
 
  private:
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
@@ -361,8 +363,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
-  bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
-  bool MuteStream(uint32 ssrc, bool mute);
+  bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
+  bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
@@ -373,14 +375,14 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool EnableRtcp(int channel);
   bool ResetRecvCodecs(int channel);
   bool SetPlayout(int channel, bool playout);
-  static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
+  static uint32_t ParseSsrc(const void* data, size_t len, bool rtcp);
   static Error WebRtcErrorToChannelError(int err_code);
 
   class WebRtcVoiceChannelRenderer;
   // Map of ssrc to WebRtcVoiceChannelRenderer object.  A new object of
   // WebRtcVoiceChannelRenderer will be created for every new stream and
   // will be destroyed when the stream goes away.
-  typedef std::map<uint32, WebRtcVoiceChannelRenderer*> ChannelMap;
+  typedef std::map<uint32_t, WebRtcVoiceChannelRenderer*> ChannelMap;
   typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
       unsigned char);
 
@@ -406,8 +408,8 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
                           const RtpHeaderExtension* extension);
   void RecreateAudioReceiveStreams();
-  void AddAudioReceiveStream(uint32 ssrc);
-  void RemoveAudioReceiveStream(uint32 ssrc);
+  void AddAudioReceiveStream(uint32_t ssrc);
+  void RemoveAudioReceiveStream(uint32_t ssrc);
   bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
 
   bool SetChannelRecvRtpHeaderExtensions(
@@ -443,13 +445,13 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
   // contained in send_channels_, otherwise not.
   ChannelMap send_channels_;
   std::vector<RtpHeaderExtension> send_extensions_;
-  uint32 default_receive_ssrc_;
+  uint32_t default_receive_ssrc_;
   // Note the default channel (voe_channel()) can reside in both
   // receive_channels_ and send_channels_ in non-conference mode and in that
   // case it will only be there if a non-zero default_receive_ssrc_ is set.
   ChannelMap receive_channels_;  // for multiple sources
-  std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_;
-  std::map<uint32, StreamParams> receive_stream_params_;
+  std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
+  std::map<uint32_t, StreamParams> receive_stream_params_;
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.