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Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )

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Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.

Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
This commit is contained in:
deadbeef
2016-06-24 14:13:06 -07:00
committed by Commit bot
parent bc5831999d
commit 1a7162dbc9
18 changed files with 905 additions and 654 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
webrtc/api/rtpreceiverinterface.h
View File

@ -18,7 +18,6 @@
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/pc/mediasession.h"
@ -27,12 +26,6 @@ namespace webrtc {
class RtpReceiverObserverInterface {
public:
// Note: Currently if there are multiple RtpReceivers of the same media type,
// they will all call OnFirstPacketReceived at once.
//
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
protected:
@ -43,9 +36,6 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
// Audio or video receiver?
virtual cricket::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
@ -56,10 +46,10 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
virtual RtpParameters GetParameters() const = 0;
virtual bool SetParameters(const RtpParameters& parameters) = 0;
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
virtual cricket::MediaType media_type() = 0;
protected:
virtual ~RtpReceiverInterface() {}
};
@ -67,11 +57,11 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
// Define proxy for RtpReceiverInterface.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
PROXY_METHOD0(cricket::MediaType, media_type);
END_SIGNALING_PROXY()
} // namespace webrtc