Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(160)

Side by Side Diff: webrtc/media/base/fakemediaengine.h

Issue 2059403002: Removed GetOutputVolume() and SetOutputVolume(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 6 months 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/media/base/mediaengine.h » ('j') | webrtc/pc/channelmanager.cc » ('J')
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 721 matching lines...) Expand 10 before | Expand all | Expand 10 after
732 bool options_changed_; 732 bool options_changed_;
733 bool fail_create_channel_; 733 bool fail_create_channel_;
734 RtpCapabilities capabilities_; 734 RtpCapabilities capabilities_;
735 }; 735 };
736 736
737 class FakeVoiceEngine : public FakeBaseEngine { 737 class FakeVoiceEngine : public FakeBaseEngine {
738 public: 738 public:
739 FakeVoiceEngine( 739 FakeVoiceEngine(
740 webrtc::AudioDeviceModule* adm, 740 webrtc::AudioDeviceModule* adm,
741 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& 741 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
742 audio_decoder_factory) 742 audio_decoder_factory) {
743 : output_volume_(-1) {
744 // Add a fake audio codec. Note that the name must not be "" as there are 743 // Add a fake audio codec. Note that the name must not be "" as there are
745 // sanity checks against that. 744 // sanity checks against that.
746 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1)); 745 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
747 } 746 }
748 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const { 747 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
749 return rtc::scoped_refptr<webrtc::AudioState>(); 748 return rtc::scoped_refptr<webrtc::AudioState>();
750 } 749 }
751 750
752 VoiceMediaChannel* CreateChannel(webrtc::Call* call, 751 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
753 const MediaConfig& config, 752 const MediaConfig& config,
754 const AudioOptions& options) { 753 const AudioOptions& options) {
755 if (fail_create_channel_) { 754 if (fail_create_channel_) {
756 return nullptr; 755 return nullptr;
757 } 756 }
758 757
759 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options); 758 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
760 channels_.push_back(ch); 759 channels_.push_back(ch);
761 return ch; 760 return ch;
762 } 761 }
763 FakeVoiceMediaChannel* GetChannel(size_t index) { 762 FakeVoiceMediaChannel* GetChannel(size_t index) {
764 return (channels_.size() > index) ? channels_[index] : NULL; 763 return (channels_.size() > index) ? channels_[index] : NULL;
765 } 764 }
766 void UnregisterChannel(VoiceMediaChannel* channel) { 765 void UnregisterChannel(VoiceMediaChannel* channel) {
767 channels_.erase(std::find(channels_.begin(), channels_.end(), channel)); 766 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
768 } 767 }
769 768
770 const std::vector<AudioCodec>& codecs() { return codecs_; } 769 const std::vector<AudioCodec>& codecs() { return codecs_; }
771 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; } 770 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
772 771
773 bool GetOutputVolume(int* level) {
774 *level = output_volume_;
775 return true;
776 }
777 bool SetOutputVolume(int level) {
778 output_volume_ = level;
779 return true;
780 }
781
782 int GetInputLevel() { return 0; } 772 int GetInputLevel() { return 0; }
783 773
784 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) { 774 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
785 return false; 775 return false;
786 } 776 }
787 777
788 void StopAecDump() {} 778 void StopAecDump() {}
789 779
790 bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes) { 780 bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes) {
791 return false; 781 return false;
792 } 782 }
793 783
794 void StopRtcEventLog() {} 784 void StopRtcEventLog() {}
795 785
796 private: 786 private:
797 std::vector<FakeVoiceMediaChannel*> channels_; 787 std::vector<FakeVoiceMediaChannel*> channels_;
798 std::vector<AudioCodec> codecs_; 788 std::vector<AudioCodec> codecs_;
799 int output_volume_;
800 789
801 friend class FakeMediaEngine; 790 friend class FakeMediaEngine;
802 }; 791 };
803 792
804 class FakeVideoEngine : public FakeBaseEngine { 793 class FakeVideoEngine : public FakeBaseEngine {
805 public: 794 public:
806 FakeVideoEngine() : capture_(false) { 795 FakeVideoEngine() : capture_(false) {
807 // Add a fake video codec. Note that the name must not be "" as there are 796 // Add a fake video codec. Note that the name must not be "" as there are
808 // sanity checks against that. 797 // sanity checks against that.
809 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0)); 798 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0));
(...skipping 73 matching lines...) Expand 10 before | Expand all | Expand 10 after
883 video_.set_rtp_header_extensions(extensions); 872 video_.set_rtp_header_extensions(extensions);
884 } 873 }
885 874
886 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) { 875 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
887 return voice_.GetChannel(index); 876 return voice_.GetChannel(index);
888 } 877 }
889 FakeVideoMediaChannel* GetVideoChannel(size_t index) { 878 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
890 return video_.GetChannel(index); 879 return video_.GetChannel(index);
891 } 880 }
892 881
893 int output_volume() const { return voice_.output_volume_; }
894 bool capture() const { return video_.capture_; } 882 bool capture() const { return video_.capture_; }
895 bool options_changed() const { 883 bool options_changed() const {
896 return video_.options_changed_; 884 return video_.options_changed_;
897 } 885 }
898 void clear_options_changed() { 886 void clear_options_changed() {
899 video_.options_changed_ = false; 887 video_.options_changed_ = false;
900 } 888 }
901 void set_fail_create_channel(bool fail) { 889 void set_fail_create_channel(bool fail) {
902 voice_.set_fail_create_channel(fail); 890 voice_.set_fail_create_channel(fail);
903 video_.set_fail_create_channel(fail); 891 video_.set_fail_create_channel(fail);
(...skipping 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
960 948
961 private: 949 private:
962 std::vector<FakeDataMediaChannel*> channels_; 950 std::vector<FakeDataMediaChannel*> channels_;
963 std::vector<DataCodec> data_codecs_; 951 std::vector<DataCodec> data_codecs_;
964 DataChannelType last_channel_type_; 952 DataChannelType last_channel_type_;
965 }; 953 };
966 954
967 } // namespace cricket 955 } // namespace cricket
968 956
969 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 957 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/media/base/mediaengine.h » ('j') | webrtc/pc/channelmanager.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698