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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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46
webrtc/api/rtpreceiverinterface.h
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@ -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