Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )

Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > 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}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
This commit is contained in:
hbos
2017-04-10 07:39:05 -07:00
committed by Commit bot
parent b0f7e39fd4
commit 8d609f6b6d
25 changed files with 563 additions and 44 deletions

View File

@ -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_