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

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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
Download Patch File Download Diff File Expand all files Collapse all files

46
webrtc/api/rtpreceiverinterface.h
View File

@ -15,6 +15,7 @@
#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
#include <string>
#include <vector>
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
@ -25,6 +26,41 @@
namespace webrtc {
enum class RtpSourceType {
SSRC,
CSRC,
};
class RtpSource {
public:
RtpSource() = delete;
RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type)
: timestamp_ms_(timestamp_ms),
source_id_(source_id),
source_type_(source_type) {}
int64_t timestamp_ms() const { return timestamp_ms_; }
void update_timestamp_ms(int64_t timestamp_ms) {
RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
timestamp_ms_ = timestamp_ms;
}
// The identifier of the source can be the CSRC or the SSRC.
uint32_t source_id() const { return source_id_; }
// The source can be either a contributing source or a synchronization source.
RtpSourceType source_type() const { return source_type_; }
// This isn't implemented yet and will always return an empty Optional.
// TODO(zhihuang): Implement this to return real audio level.
rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
private:
int64_t timestamp_ms_;
uint32_t source_id_;
RtpSourceType source_type_;
};
class RtpReceiverObserverInterface {
public:
// Note: Currently if there are multiple RtpReceivers of the same media type,
@ -61,6 +97,13 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the
// content::FakeRtpReceiver in Chromium.
virtual std::vector<RtpSource> GetSources() const {
return std::vector<RtpSource>();
}
protected:
virtual ~RtpReceiverInterface() {}
};
@ -76,7 +119,8 @@ BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
END_PROXY_MAP()
PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
END_PROXY_MAP()
} // namespace webrtc