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}
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index 8607d93..fd233ab 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -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 @@
// 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 @@
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