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Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )

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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}
This commit is contained in:
olka
2017-04-10 04:38:13 -07:00
committed by Commit bot
parent 925e9d762c
commit fbcc5cb386
25 changed files with 44 additions and 563 deletions
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22
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
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@ -11,10 +11,7 @@
#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"
@ -59,16 +56,6 @@ 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;
@ -79,9 +66,6 @@ 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_;
@ -100,12 +84,6 @@ 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_