mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
0bc58cf876
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
88 lines
3.2 KiB
C++
88 lines
3.2 KiB
C++
/*
|
|
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
// This file contains interfaces for RtpSenders
|
|
// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
|
|
|
|
#ifndef API_RTPSENDERINTERFACE_H_
|
|
#define API_RTPSENDERINTERFACE_H_
|
|
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "api/dtmfsenderinterface.h"
|
|
#include "api/mediastreaminterface.h"
|
|
#include "api/mediatypes.h"
|
|
#include "api/proxy.h"
|
|
#include "api/rtcerror.h"
|
|
#include "api/rtpparameters.h"
|
|
#include "rtc_base/deprecation.h"
|
|
#include "rtc_base/refcount.h"
|
|
#include "rtc_base/scoped_ref_ptr.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpSenderInterface : public rtc::RefCountInterface {
|
|
public:
|
|
// Returns true if successful in setting the track.
|
|
// Fails if an audio track is set on a video RtpSender, or vice-versa.
|
|
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
|
|
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
|
|
|
|
// Returns primary SSRC used by this sender for sending media.
|
|
// Returns 0 if not yet determined.
|
|
// TODO(deadbeef): Change to absl::optional.
|
|
// TODO(deadbeef): Remove? With GetParameters this should be redundant.
|
|
virtual uint32_t ssrc() const = 0;
|
|
|
|
// Audio or video sender?
|
|
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;
|
|
|
|
// Returns a list of media stream ids associated with this sender's track.
|
|
// These are signalled in the SDP so that the remote side can associate
|
|
// tracks.
|
|
virtual std::vector<std::string> stream_ids() const = 0;
|
|
|
|
virtual RtpParameters GetParameters() = 0;
|
|
// Note that only a subset of the parameters can currently be changed. See
|
|
// rtpparameters.h
|
|
// The encodings are in increasing quality order for simulcast.
|
|
virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
|
|
|
|
// Returns null for a video sender.
|
|
virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
|
|
|
|
protected:
|
|
virtual ~RtpSenderInterface() {}
|
|
};
|
|
|
|
// Define proxy for RtpSenderInterface.
|
|
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
|
|
// are called on is an implementation detail.
|
|
BEGIN_SIGNALING_PROXY_MAP(RtpSender)
|
|
PROXY_SIGNALING_THREAD_DESTRUCTOR()
|
|
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
|
|
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
|
|
PROXY_CONSTMETHOD0(uint32_t, ssrc)
|
|
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
|
|
PROXY_CONSTMETHOD0(std::string, id)
|
|
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
|
|
PROXY_METHOD0(RtpParameters, GetParameters);
|
|
PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
|
|
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
|
|
END_PROXY_MAP()
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_RTPSENDERINTERFACE_H_
|