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28 | 28 |
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29 | 29 | class ADTSAudioFileServerMediaSubsession_BC: public FileServerMediaSubsession_BC {
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30 | 30 | public:
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31 |
| - static ADTSAudioFileServerMediaSubsession_BC* |
32 |
| - createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource, |
33 |
| - int sampleRate, int numChannels); |
| 31 | + static ADTSAudioFileServerMediaSubsession_BC* |
| 32 | + createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource, |
| 33 | + int sampleRate, int numChannels, Boolean enableSpeaker); |
34 | 34 |
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35 | 35 | protected:
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36 |
| - ADTSAudioFileServerMediaSubsession_BC(UsageEnvironment& env, |
37 |
| - char const* fileName, Boolean reuseFirstSource, |
38 |
| - int sampleRate, int numChannels); |
39 |
| - // called only by createNew(); |
40 |
| - virtual ~ADTSAudioFileServerMediaSubsession_BC(); |
| 36 | + ADTSAudioFileServerMediaSubsession_BC(UsageEnvironment& env, |
| 37 | + char const* fileName, Boolean reuseFirstSource, |
| 38 | + int sampleRate, int numChannels, Boolean enableSpeaker); |
| 39 | + // called only by createNew(); |
| 40 | + virtual ~ADTSAudioFileServerMediaSubsession_BC(); |
41 | 41 |
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42 | 42 | protected: // redefined virtual functions
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43 |
| - virtual FramedSource* createNewStreamSource(unsigned clientSessionId, |
44 |
| - unsigned& estBitrate); |
45 |
| - virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock, |
46 |
| - unsigned char rtpPayloadTypeIfDynamic, |
47 |
| - FramedSource* inputSource); |
| 43 | + virtual FramedSource* createNewStreamSource(unsigned clientSessionId, |
| 44 | + unsigned& estBitrate); |
| 45 | + virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock, |
| 46 | + unsigned char rtpPayloadTypeIfDynamic, |
| 47 | + FramedSource* inputSource); |
48 | 48 |
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49 |
| - virtual MediaSink* createNewStreamDestination(unsigned clientSessionId, |
50 |
| - unsigned& estBitrate); |
51 |
| - // "estBitrate" is the stream's estimated bitrate, in kbps |
52 |
| - virtual RTPSource* createNewRTPSource(Groupsock* rtpGroupsock, |
53 |
| - unsigned char rtpPayloadTypeIfDynamic, |
54 |
| - MediaSink* outputSink); |
55 |
| - virtual char const* getAuxSDPLineForBackChannel(MediaSink* mediaSink, RTPSource* rtpSource); |
| 49 | + virtual MediaSink* createNewStreamDestination(unsigned clientSessionId, |
| 50 | + unsigned& estBitrate); |
| 51 | + // "estBitrate" is the stream's estimated bitrate, in kbps |
| 52 | + virtual RTPSource* createNewRTPSource(Groupsock* rtpGroupsock, |
| 53 | + unsigned char rtpPayloadTypeIfDynamic, |
| 54 | + MediaSink* outputSink); |
| 55 | + virtual char const* getAuxSDPLineForBackChannel(MediaSink* mediaSink, RTPSource* rtpSource); |
56 | 56 |
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57 | 57 | private:
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58 |
| - int fSampleRate; |
59 |
| - int fNumChannels; |
60 |
| - char* fAuxSDPLine; |
61 |
| - unsigned char fRTPPayloadFormat; |
62 |
| - unsigned fRTPTimestampFrequency; |
| 58 | + int fSampleRate; |
| 59 | + int fNumChannels; |
| 60 | + Boolean fEnableSpeaker; |
| 61 | + char* fAuxSDPLine; |
| 62 | + unsigned char fRTPPayloadFormat; |
| 63 | + unsigned fRTPTimestampFrequency; |
63 | 64 | };
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64 | 65 |
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65 | 66 | #endif
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