OLD | NEW |
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 59 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
70 WEBRTC_STUB(Initialize, ( | 70 WEBRTC_STUB(Initialize, ( |
71 const webrtc::ProcessingConfig& processing_config)); | 71 const webrtc::ProcessingConfig& processing_config)); |
72 | 72 |
73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { | 73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { |
74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; | 74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; |
75 } | 75 } |
76 | 76 |
77 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); | 77 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); |
78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); | 78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); |
79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); | 79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); |
80 WEBRTC_STUB_CONST(num_input_channels, ()); | 80 size_t num_input_channels() const override { return 0; } |
81 WEBRTC_STUB_CONST(num_proc_channels, ()); | 81 size_t num_proc_channels() const override { return 0; } |
82 WEBRTC_STUB_CONST(num_output_channels, ()); | 82 size_t num_output_channels() const override { return 0; } |
83 WEBRTC_STUB_CONST(num_reverse_channels, ()); | 83 size_t num_reverse_channels() const override { return 0; } |
84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); | 84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); |
85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); | 85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); |
86 WEBRTC_STUB(ProcessStream, ( | 86 WEBRTC_STUB(ProcessStream, ( |
87 const float* const* src, | 87 const float* const* src, |
88 size_t samples_per_channel, | 88 size_t samples_per_channel, |
89 int input_sample_rate_hz, | 89 int input_sample_rate_hz, |
90 webrtc::AudioProcessing::ChannelLayout input_layout, | 90 webrtc::AudioProcessing::ChannelLayout input_layout, |
91 int output_sample_rate_hz, | 91 int output_sample_rate_hz, |
92 webrtc::AudioProcessing::ChannelLayout output_layout, | 92 webrtc::AudioProcessing::ChannelLayout output_layout, |
93 float* const* dest)); | 93 float* const* dest)); |
(...skipping 721 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
815 int playout_fail_channel_; | 815 int playout_fail_channel_; |
816 int send_fail_channel_; | 816 int send_fail_channel_; |
817 int recording_sample_rate_; | 817 int recording_sample_rate_; |
818 int playout_sample_rate_; | 818 int playout_sample_rate_; |
819 FakeAudioProcessing audio_processing_; | 819 FakeAudioProcessing audio_processing_; |
820 }; | 820 }; |
821 | 821 |
822 } // namespace cricket | 822 } // namespace cricket |
823 | 823 |
824 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 824 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
OLD | NEW |