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Unified Diff: talk/media/webrtc/webrtcvoiceengine.h

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: henrikg@ feedback Created 5 years, 2 months ago
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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.

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