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. |