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: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
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}
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index fd233ab..8607d93 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -15,7 +15,6 @@
#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
#include <string>
-#include <vector>
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
@@ -26,41 +25,6 @@
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,
@@ -97,13 +61,6 @@
// 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() {}
};
@@ -119,8 +76,7 @@
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
- PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
- END_PROXY_MAP()
+END_PROXY_MAP()
} // namespace webrtc