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Update talk to 50918584.

Browse Source 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2013-08-15 23:38:54 +00:00
parent dde7d4c6ed
commit 822fbd8b68
108 changed files with 2926 additions and 4301 deletions
Download Patch File Download Diff File Expand all files Collapse all files

35
webrtc/voice_engine/channel.h
View File

@ -35,16 +35,21 @@
namespace webrtc
{
class CriticalSectionWrapper;
class ProcessThread;
class AudioDeviceModule;
class RtpRtcp;
class CriticalSectionWrapper;
class FileWrapper;
class ProcessThread;
class ReceiveStatistics;
class RtpDump;
class VoiceEngineObserver;
class RTPPayloadRegistry;
class RtpReceiver;
class RTPReceiverAudio;
class RtpRtcp;
class TelephoneEventHandler;
class VoEMediaProcess;
class VoERTPObserver;
class VoERTCPObserver;
class VoERTPObserver;
class VoiceEngineObserver;
struct CallStatistics;
struct ReportBlock;
@ -133,12 +138,6 @@ public:
int32_t DeRegisterExternalTransport();
int32_t ReceivedRTPPacket(const int8_t* data, int32_t length);
int32_t ReceivedRTCPPacket(const int8_t* data, int32_t length);
int32_t SetPacketTimeoutNotification(bool enable, int timeoutSeconds);
int32_t GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds);
int32_t RegisterDeadOrAliveObserver(VoEConnectionObserver& observer);
int32_t DeRegisterDeadOrAliveObserver();
int32_t SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds);
int32_t GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds);
// VoEFile
int StartPlayingFileLocally(const char* fileName, bool loop,
@ -215,7 +214,7 @@ public:
int SetInitSequenceNumber(short sequenceNumber);
// VoEVideoSyncExtended
int GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const;
int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
// VoEEncryption
int RegisterExternalEncryption(Encryption& encryption);
@ -307,6 +306,11 @@ public:
uint16_t payloadSize,
const WebRtcRTPHeader* rtpHeader);
bool OnRecoveredPacket(const uint8_t* packet, int packet_length) {
// Generic FEC not supported for audio.
return true;
}
public:
// From RtpFeedback in the RTP/RTCP module
int32_t OnInitializeDecoder(
@ -330,6 +334,8 @@ public:
void OnIncomingCSRCChanged(int32_t id,
uint32_t CSRC, bool added);
void ResetStatistics();
public:
// From RtcpFeedback in the RTP/RTCP module
void OnApplicationDataReceived(int32_t id,
@ -433,6 +439,7 @@ public:
uint32_t EncodeAndSend();
private:
bool IsPacketRetransmitted(const RTPHeader& header) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
int InsertInbandDtmfTone();
int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
@ -453,6 +460,10 @@ private:
private:
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
scoped_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
scoped_ptr<RtpRtcp> _rtpRtcpModule;
AudioCodingModule& _audioCodingModule;
RtpDump& _rtpDumpIn;