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

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 174 matching lines @@
   const AudioOptions& options() const { return options_; }
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
-  bool SetAudioSend(uint32 ssrc, bool enable, const AudioOptions* options,
+  bool SetAudioSend(uint32_t ssrc,
+                    bool enable,
+                    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;
   void SetTypingDetectionParameters(int time_window,
                                     int cost_per_typing,
                                     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 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;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   bool GetStats(VoiceMediaInfo* info) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendPacket(&packet);
   }
 
   bool SendRtcp(const uint8_t* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendRtcp(&packet);
   }
 
   void OnError(int error);
 
-  int GetReceiveChannelId(uint32 ssrc) const;
-  int GetSendChannelId(uint32 ssrc) const;
+  int GetReceiveChannelId(uint32_t ssrc) const;
+  int GetSendChannelId(uint32_t ssrc) const;
 
  private:
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetOptions(const AudioOptions& options);
   bool SetMaxSendBandwidth(int bps);
   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(); }
   int GetOutputLevel(int channel);
   bool GetRedSendCodec(const AudioCodec& red_codec,
                        const std::vector<AudioCodec>& all_codecs,
                        webrtc::CodecInst* send_codec);
   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);
 
   void SetNack(int channel, bool nack_enabled);
   void SetNack(const ChannelMap& channels, bool nack_enabled);
   bool SetSendCodec(const webrtc::CodecInst& send_codec);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   void ConfigureSendChannel(int channel);
   bool ConfigureRecvChannel(int channel);
   bool DeleteChannel(int channel);
   bool InConferenceMode() const {
     return options_.conference_mode.GetWithDefaultIfUnset(false);
   }
   bool IsDefaultChannel(int channel_id) const {
     return channel_id == voe_channel();
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
 
   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(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
   bool SetChannelSendRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
 
   rtc::ThreadChecker thread_checker_;
@@ skipping 13 matching lines @@
   bool typing_noise_detected_;
   SendFlags desired_send_;
   SendFlags send_;
   webrtc::Call* const call_;
 
   // send_channels_ contains the channels which are being used for sending.
   // When the default channel (voe_channel) is used for sending, it is
   // 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.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_