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| 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 |
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| 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)); |
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| 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); |
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| 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_ |
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