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RTC: Cleanup code, remove RTC from SrsSource

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This commit is contained in:
winlin 2020-05-12 13:43:09 +08:00
parent 25496b734b
commit d434dc951d
3 changed files with 1 additions and 90 deletions
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22
trunk/src/app/srs_app_source.hpp
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@ -62,9 +62,6 @@ class SrsBuffer;
#ifdef SRS_HDS
class SrsHds;
#endif
#ifdef SRS_RTC
class SrsRtcPublisher;
#endif
// The time jitter algorithm:
// 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
@ -358,10 +355,6 @@ private:
private:
// The format, codec information.
SrsRtmpFormat* format;
#ifdef SRS_RTC
// rtc handler
SrsRtc* rtc;
#endif
// hls handler.
SrsHls* hls;
// The DASH encoder.
@ -560,10 +553,6 @@ private:
// The last die time, when all consumers quit and no publisher,
// We will remove the source when source die.
srs_utime_t die_at;
#ifdef SRS_RTC
private:
SrsRtcPublisher* rtc_publisher_;
#endif
public:
SrsSource();
virtual ~SrsSource();
@ -630,17 +619,6 @@ public:
virtual void on_edge_proxy_unpublish();
public:
virtual std::string get_curr_origin();
#ifdef SRS_RTC
public:
// For RTC, we need to package SPS/PPS(in cached meta) before each IDR.
SrsMetaCache* cached_meta();
// Get and set the publisher, passed to consumer to process requests such as PLI.
SrsRtcPublisher* rtc_publisher();
void set_rtc_publisher(SrsRtcPublisher* v);
// When got RTC audio message, which is encoded in opus.
// TODO: FIXME: Merge with on_audio.
srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
#endif
};
#endif