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 94 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
105 (const float* const* src, | 105 (const float* const* src, |
106 const webrtc::StreamConfig& reverse_input_config, | 106 const webrtc::StreamConfig& reverse_input_config, |
107 const webrtc::StreamConfig& reverse_output_config, | 107 const webrtc::StreamConfig& reverse_output_config, |
108 float* const* dest)); | 108 float* const* dest)); |
109 WEBRTC_STUB(set_stream_delay_ms, (int delay)); | 109 WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
110 WEBRTC_STUB_CONST(stream_delay_ms, ()); | 110 WEBRTC_STUB_CONST(stream_delay_ms, ()); |
111 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | 111 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |
112 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); | 112 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); |
113 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); | 113 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); |
114 WEBRTC_STUB_CONST(delay_offset_ms, ()); | 114 WEBRTC_STUB_CONST(delay_offset_ms, ()); |
115 WEBRTC_STUB(StartDebugRecording, | 115 WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); |
116 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); | 116 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); |
117 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); | |
118 WEBRTC_STUB(StopDebugRecording, ()); | 117 WEBRTC_STUB(StopDebugRecording, ()); |
119 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | 118 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); |
120 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | 119 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } |
121 webrtc::EchoControlMobile* echo_control_mobile() const override { | 120 webrtc::EchoControlMobile* echo_control_mobile() const override { |
122 return NULL; | 121 return NULL; |
123 } | 122 } |
124 webrtc::GainControl* gain_control() const override { return NULL; } | 123 webrtc::GainControl* gain_control() const override { return NULL; } |
125 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | 124 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } |
126 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | 125 webrtc::LevelEstimator* level_estimator() const override { return NULL; } |
127 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | 126 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } |
(...skipping 686 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
814 int playout_fail_channel_; | 813 int playout_fail_channel_; |
815 int send_fail_channel_; | 814 int send_fail_channel_; |
816 int recording_sample_rate_; | 815 int recording_sample_rate_; |
817 int playout_sample_rate_; | 816 int playout_sample_rate_; |
818 FakeAudioProcessing audio_processing_; | 817 FakeAudioProcessing audio_processing_; |
819 }; | 818 }; |
820 | 819 |
821 } // namespace cricket | 820 } // namespace cricket |
822 | 821 |
823 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 822 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
OLD | NEW |