a8eb756a34
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
29 lines
913 B
C++
29 lines
913 B
C++
/*
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* Copyright (c) 2013 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 WEBRTC_TEST_MOCK_TRANSPORT_H_
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#define WEBRTC_TEST_MOCK_TRANSPORT_H_
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#include "webrtc/api/call/transport.h"
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#include "webrtc/test/gmock.h"
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namespace webrtc {
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class MockTransport : public Transport {
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public:
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MOCK_METHOD3(SendRtp,
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bool(const uint8_t* data,
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size_t len,
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const PacketOptions& options));
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MOCK_METHOD2(SendRtcp, bool(const uint8_t* data, size_t len));
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};
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} // namespace webrtc
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#endif // WEBRTC_TEST_MOCK_TRANSPORT_H_
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