Index: webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
index 54a99c93a1de5fc4dbb62a08536d529242ca9572..462458ddb1defc7aa5569fcfe0b343ec7004a869 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h |
@@ -11,6 +11,9 @@ |
#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ |
#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ |
+#include <vector> |
+ |
+#include "webrtc/api/rtpreceiverinterface.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
#include "webrtc/typedefs.h" |
@@ -20,6 +23,28 @@ struct CodecInst; |
class RTPPayloadRegistry; |
class VideoCodec; |
+class RtpContributingSource : public RtpContributingSourceInterface { |
+ public: |
+ RtpContributingSource() : timestamp_(0), source_(0) {} |
+ |
+ explicit RtpContributingSource(int64_t timestamp, uint32_t source) |
+ : timestamp_(timestamp), source_(source) {} |
+ |
+ int64_t timestamp() const override { return timestamp_; } |
+ |
+ uint32_t source() const override { return source_; } |
+ |
+ // Always return 0 in the current implementation. |
+ // TODO(zhihuang): Implement this method to return real audio level. |
+ int8_t audio_level() const override { return 0; } |
hbos
2017/03/30 09:51:54
Return a value that doesn't map to a meaningful va
Zhi Huang
2017/03/31 06:44:04
Done.
|
+ |
+ void set_timestamp(int64_t timestamp) { timestamp_ = timestamp; } |
+ |
+ private: |
+ int64_t timestamp_; |
+ uint32_t source_; |
+}; |
+ |
class TelephoneEventHandler { |
public: |
virtual ~TelephoneEventHandler() {} |
@@ -89,6 +114,9 @@ class RtpReceiver { |
// Returns the current energy of the RTP stream received. |
virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0; |
+ |
+ virtual const std::vector<RtpContributingSource*>& |
+ GetContributingSources() = 0; |
}; |
} // namespace webrtc |