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 103 matching lines @@
   bool SetLocalMonitor(bool enable);
 
   const std::vector<AudioCodec>& codecs();
   bool FindCodec(const AudioCodec& codec);
   bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
 
   const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
 
   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);
 
   // Method from webrtc::VoEMediaProcess
   void Process(int channel,
                webrtc::ProcessingTypes type,
                int16_t audio10ms[],
                size_t length,
                int sampling_freq,
                bool is_stereo) override;
@@ skipping 18 matching lines @@
   bool StartAecDump(rtc::PlatformFile file);
 
   // Check whether the supplied trace should be ignored.
   bool ShouldIgnoreTrace(const std::string& trace);
 
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
 
  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();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Applies either options or overrides.  Every option that is "set"
   // will be applied.  Every option not "set" will be ignored.  This
   // allows us to selectively turn on and off different options easily
@@ skipping 18 matching lines @@
 
   // webrtc::VoiceEngineObserver:
   void CallbackOnError(int channel, int errCode) override;
 
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
       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);
   bool PauseLocalMonitor();
   bool ResumeLocalMonitor();
 
   bool UnregisterProcessorChannel(MediaProcessorDirection channel_direction,
-                                  uint32 ssrc,
+                                  uint32_t ssrc,
                                   VoiceProcessor* voice_processor,
                                   MediaProcessorDirection processor_direction);
 
   void StartAecDump(const std::string& filename);
   void StopAecDump();
   int CreateVoiceChannel(VoEWrapper* voe);
 
   // When a voice processor registers with the engine, it is connected
   // to either the Rx or Tx signals, based on the direction parameter.
   // SignalXXMediaFrame will be invoked for every audio packet.
@@ skipping 30 matching lines @@
   // options_ is stored so that when option_overrides_ is cleared, we
   // can restore the options_ without the option_overrides.
   AudioOptions options_;
   AudioOptions option_overrides_;
 
   // When the media processor registers with the engine, the ssrc is cached
   // here so that a look up need not be made when the callback is invoked.
   // 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_;
 
   // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
   // values, and apply them in case they are missing in the audio options. We
   // need to do this because SetExtraOptions() will revert to defaults for
   // options which are not provided.
   Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
@@ skipping 14 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 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;
   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 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;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   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
   int SendPacket(const void* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendPacket(&packet) ? static_cast<int>(len) : -1;
   }
 
   int SendRTCPPacket(const void* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     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);
   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 15 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_