b86d4e4a8d
This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
49 lines
2.1 KiB
C++
49 lines
2.1 KiB
C++
/*
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* Copyright (c) 2015 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_VOE_CHANNEL_PROXY_H_
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#define WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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#include <string>
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#include "testing/gmock/include/gmock/gmock.h"
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#include "webrtc/voice_engine/channel_proxy.h"
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namespace webrtc {
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namespace test {
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class MockVoEChannelProxy : public voe::ChannelProxy {
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public:
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MOCK_METHOD1(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
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MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
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MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD3(SetCongestionControlObjects,
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void(RtpPacketSender* rtp_packet_sender,
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TransportFeedbackObserver* transport_feedback_observer,
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PacketRouter* seq_num_allocator));
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
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MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
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uint32_t duration_ms));
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_
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