| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 69e88e44c472a0d49f9f6088ba3f581242d39328..b384b9f774d1ccd8c9f40cac6d02a8c8cea68645 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -135,7 +135,7 @@ 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;
|
| + uint32_t* ssrc) const;
|
|
|
| void StartAecDump(const std::string& filename);
|
| void StopAecDump();
|
| @@ -195,13 +195,15 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| 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;
|
| @@ -210,10 +212,10 @@ 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 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;
|
| @@ -237,11 +239,11 @@ class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
|
| return VoiceMediaChannel::SendRtcp(&packet);
|
| }
|
|
|
| - 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);
|
|
|
| - 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);
|
| @@ -252,8 +254,8 @@ 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(); }
|
| @@ -264,14 +266,14 @@ 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);
|
|
|
| @@ -297,8 +299,8 @@ 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(
|
| @@ -332,13 +334,13 @@ 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.
|
|
|