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Added the GetSources() to the RtpReceiverInterface and implemented

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it for the AudioRtpReceiver.

This method returns a vector of RtpSource(both CSRC source and SSRC
source) which contains the ID of a source, the timestamp, the source
type (SSRC or CSRC) and the audio level.

The RtpSource objects are buffered and maintained by the
RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
the info of the contributing source will be pulled along the object
chain:
AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
AudioReceiveStream -> voe::Channel -> RtpRtcp module

Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource

BUG=chromium:703122
TBR=stefan@webrtc.org, danilchap@webrtc.org

Review-Url: https://codereview.webrtc.org/2770233003
Cr-Commit-Position: refs/heads/master@{#17591}
This commit is contained in:
zhihuang
2017-04-07 10:57:22 -07:00
committed by Commit bot
parent bb16a483f2
commit 292084c376
25 changed files with 563 additions and 44 deletions
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22
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
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@ -11,7 +11,10 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
#include <list>
#include <memory>
#include <unordered_map>
#include <vector>
#include "webrtc/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
@ -56,6 +59,16 @@ class RtpReceiverImpl : public RtpReceiver {
TelephoneEventHandler* GetTelephoneEventHandler() override;
std::vector<RtpSource> GetSources() const override;
const std::vector<RtpSource>& ssrc_sources_for_testing() const {
return ssrc_sources_;
}
const std::list<RtpSource>& csrc_sources_for_testing() const {
return csrc_sources_;
}
private:
bool HaveReceivedFrame() const;
@ -66,6 +79,9 @@ class RtpReceiverImpl : public RtpReceiver {
bool* is_red,
PayloadUnion* payload);
void UpdateSources();
void RemoveOutdatedSources(int64_t now_ms);
Clock* clock_;
RTPPayloadRegistry* rtp_payload_registry_;
std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
@ -84,6 +100,12 @@ class RtpReceiverImpl : public RtpReceiver {
uint32_t last_received_timestamp_;
int64_t last_received_frame_time_ms_;
uint16_t last_received_sequence_number_;
std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
iterator_by_csrc_;
// The RtpSource objects are sorted chronologically.
std::list<RtpSource> csrc_sources_;
std::vector<RtpSource> ssrc_sources_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_