1ff16c87aa
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes to avoid existing chromium tests to fail. Instead of replacing the existing RtpSender::set_stream_ids() to also fire OnRenegotiationNeeded(), this CL keeps the old set_stream_ids() and adds the new RtpSender::SetStreams() which sets the stream IDs and fires the callback. This allows existing callsites to maintain behavior, and reserve SetStreams() for the cases when we want OnRenegotiationNeeded() to fire. Using the SetStreams() name instead of SetStreamIDs() to match the W3C spec and to make it more different that the existing set_stream_ids(). Original change's description: > Improve spec compliance of SetStreamIDs in RtpSenderInterface > > This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded > event if needed and exposes the method on RtpSenderInterface. > > This is a spec-compliance change. > > Bug: webrtc:10129 > Change-Id: I2b98b92665c847102838b094516a79b24de0e47e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121 > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27974} Bug: webrtc:10129 Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439 Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27992}
62 lines
2.5 KiB
C++
62 lines
2.5 KiB
C++
/*
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* Copyright 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
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#define PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
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#include <string>
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#include <vector>
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#include "pc/rtp_sender.h"
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#include "test/gmock.h"
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namespace webrtc {
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// The definition of MockRtpSender is copied in to avoid multiple inheritance.
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class MockRtpSenderInternal : public RtpSenderInternal {
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public:
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// RtpSenderInterface methods.
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MOCK_METHOD1(SetTrack, bool(MediaStreamTrackInterface*));
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MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
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MOCK_CONST_METHOD0(ssrc, uint32_t());
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MOCK_CONST_METHOD0(dtls_transport,
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rtc::scoped_refptr<DtlsTransportInterface>());
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MOCK_CONST_METHOD0(media_type, cricket::MediaType());
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MOCK_CONST_METHOD0(id, std::string());
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MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
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MOCK_CONST_METHOD0(init_send_encodings, std::vector<RtpEncodingParameters>());
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MOCK_METHOD1(set_transport, void(rtc::scoped_refptr<DtlsTransportInterface>));
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MOCK_CONST_METHOD0(GetParameters, RtpParameters());
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MOCK_CONST_METHOD0(GetParametersInternal, RtpParameters());
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MOCK_METHOD1(SetParameters, RTCError(const RtpParameters&));
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MOCK_METHOD1(SetParametersInternal, RTCError(const RtpParameters&));
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MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
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MOCK_METHOD1(SetFrameEncryptor,
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void(rtc::scoped_refptr<FrameEncryptorInterface>));
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MOCK_CONST_METHOD0(GetFrameEncryptor,
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rtc::scoped_refptr<FrameEncryptorInterface>());
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// RtpSenderInternal methods.
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MOCK_METHOD1(SetMediaChannel, void(cricket::MediaChannel*));
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MOCK_METHOD1(SetSsrc, void(uint32_t));
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MOCK_METHOD1(set_stream_ids, void(const std::vector<std::string>&));
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MOCK_METHOD1(SetStreams, void(const std::vector<std::string>&));
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MOCK_METHOD1(set_init_send_encodings,
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void(const std::vector<RtpEncodingParameters>&));
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MOCK_METHOD0(Stop, void());
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MOCK_CONST_METHOD0(AttachmentId, int());
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MOCK_METHOD1(DisableEncodingLayers,
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RTCError(const std::vector<std::string>&));
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};
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} // namespace webrtc
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#endif // PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
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