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| 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, |
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| 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, (const char filename[kMaxFilenameSize])); | 115 WEBRTC_STUB(StartDebugRecording, |
| 116 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); | 116 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); |
| 117 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); |
| 117 WEBRTC_STUB(StopDebugRecording, ()); | 118 WEBRTC_STUB(StopDebugRecording, ()); |
| 118 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | 119 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); |
| 119 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | 120 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } |
| 120 webrtc::EchoControlMobile* echo_control_mobile() const override { | 121 webrtc::EchoControlMobile* echo_control_mobile() const override { |
| 121 return NULL; | 122 return NULL; |
| 122 } | 123 } |
| 123 webrtc::GainControl* gain_control() const override { return NULL; } | 124 webrtc::GainControl* gain_control() const override { return NULL; } |
| 124 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | 125 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } |
| 125 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | 126 webrtc::LevelEstimator* level_estimator() const override { return NULL; } |
| 126 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | 127 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } |
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| 813 int playout_fail_channel_; | 814 int playout_fail_channel_; |
| 814 int send_fail_channel_; | 815 int send_fail_channel_; |
| 815 int recording_sample_rate_; | 816 int recording_sample_rate_; |
| 816 int playout_sample_rate_; | 817 int playout_sample_rate_; |
| 817 FakeAudioProcessing audio_processing_; | 818 FakeAudioProcessing audio_processing_; |
| 818 }; | 819 }; |
| 819 | 820 |
| 820 } // namespace cricket | 821 } // namespace cricket |
| 821 | 822 |
| 822 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 823 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
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