webrtc_m130/webrtc/api/test/mock_rtpsender.h
deadbeef 20cb0c1c85 Move DTMF sender to RtpSender (as opposed to WebRtcSession).
Previously in the spec, there was a createDtmfSender method on
PeerConnection, but that's been replaced by a "dtmf" attribute
on RtpSender, which allows getting a DTMF sender without having
an audio track.

This also simplifies the code slightly, since tracks are now not
necessary for identification.

BUG=webrtc:4180

Review-Url: https://codereview.webrtc.org/2666853002
Cr-Commit-Position: refs/heads/master@{#16409}
2017-02-02 04:27:00 +00:00

38 lines
1.3 KiB
C++

/*
* Copyright 2016 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.
*/
#ifndef WEBRTC_API_TEST_MOCK_RTPSENDER_H_
#define WEBRTC_API_TEST_MOCK_RTPSENDER_H_
#include <string>
#include <vector>
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
class MockRtpSender : public rtc::RefCountedObject<RtpSenderInterface> {
public:
MOCK_METHOD1(SetTrack, bool(MediaStreamTrackInterface*));
MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
MOCK_CONST_METHOD0(ssrc, uint32_t());
MOCK_CONST_METHOD0(media_type, cricket::MediaType());
MOCK_CONST_METHOD0(id, std::string());
MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
MOCK_CONST_METHOD0(GetParameters, RtpParameters());
MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
};
} // namespace webrtc
#endif // WEBRTC_API_TEST_MOCK_RTPSENDER_H_