Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(4)

Side by Side Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1402403008: Changed FakeVoiceEngine into a MockVoiceEngine. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: unneeded include Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/audio/audio_receive_stream_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2010 Google Inc. 3 * Copyright 2010 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 758 matching lines...) Expand 10 before | Expand all | Expand 10 after
769 unsigned int& playoutTimestamp, 769 unsigned int& playoutTimestamp,
770 unsigned int* jitter, 770 unsigned int* jitter,
771 unsigned short* fractionLost)); 771 unsigned short* fractionLost));
772 WEBRTC_STUB(GetRemoteRTCPReportBlocks, 772 WEBRTC_STUB(GetRemoteRTCPReportBlocks,
773 (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)); 773 (int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
774 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, 774 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
775 unsigned int& maxJitterMs, 775 unsigned int& maxJitterMs,
776 unsigned int& discardedPackets)); 776 unsigned int& discardedPackets));
777 WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)); 777 WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
778 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) { 778 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
779 return SetFECStatus(channel, enable, redPayloadtype);
780 }
781 // TODO(minyue): remove the below function when transition to SetREDStatus
782 // is finished.
783 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) {
784 WEBRTC_CHECK_CHANNEL(channel); 779 WEBRTC_CHECK_CHANNEL(channel);
785 channels_[channel]->red = enable; 780 channels_[channel]->red = enable;
786 channels_[channel]->red_type = redPayloadtype; 781 channels_[channel]->red_type = redPayloadtype;
787 return 0; 782 return 0;
788 } 783 }
789 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) { 784 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
790 return GetFECStatus(channel, enable, redPayloadtype);
791 }
792 // TODO(minyue): remove the below function when transition to GetREDStatus
793 // is finished.
794 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) {
795 WEBRTC_CHECK_CHANNEL(channel); 785 WEBRTC_CHECK_CHANNEL(channel);
796 enable = channels_[channel]->red; 786 enable = channels_[channel]->red;
797 redPayloadtype = channels_[channel]->red_type; 787 redPayloadtype = channels_[channel]->red_type;
798 return 0; 788 return 0;
799 } 789 }
800 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) { 790 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
801 WEBRTC_CHECK_CHANNEL(channel); 791 WEBRTC_CHECK_CHANNEL(channel);
802 channels_[channel]->nack = enable; 792 channels_[channel]->nack = enable;
803 channels_[channel]->nack_max_packets = maxNoPackets; 793 channels_[channel]->nack_max_packets = maxNoPackets;
804 return 0; 794 return 0;
(...skipping 215 matching lines...) Expand 10 before | Expand all | Expand 10 after
1020 int playout_sample_rate_; 1010 int playout_sample_rate_;
1021 DtmfInfo dtmf_info_; 1011 DtmfInfo dtmf_info_;
1022 FakeAudioProcessing audio_processing_; 1012 FakeAudioProcessing audio_processing_;
1023 }; 1013 };
1024 1014
1025 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID 1015 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
1026 1016
1027 } // namespace cricket 1017 } // namespace cricket
1028 1018
1029 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 1019 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/audio/audio_receive_stream_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698