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 138 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
149 (const float* const* src, | 149 (const float* const* src, |
150 const webrtc::StreamConfig& reverse_input_config, | 150 const webrtc::StreamConfig& reverse_input_config, |
151 const webrtc::StreamConfig& reverse_output_config, | 151 const webrtc::StreamConfig& reverse_output_config, |
152 float* const* dest)); | 152 float* const* dest)); |
153 WEBRTC_STUB(set_stream_delay_ms, (int delay)); | 153 WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
154 WEBRTC_STUB_CONST(stream_delay_ms, ()); | 154 WEBRTC_STUB_CONST(stream_delay_ms, ()); |
155 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | 155 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |
156 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); | 156 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); |
157 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); | 157 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); |
158 WEBRTC_STUB_CONST(delay_offset_ms, ()); | 158 WEBRTC_STUB_CONST(delay_offset_ms, ()); |
159 WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); | 159 WEBRTC_STUB(StartDebugRecording, |
160 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); | 160 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); |
| 161 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); |
161 WEBRTC_STUB(StopDebugRecording, ()); | 162 WEBRTC_STUB(StopDebugRecording, ()); |
162 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | 163 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); |
163 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | 164 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } |
164 webrtc::EchoControlMobile* echo_control_mobile() const override { | 165 webrtc::EchoControlMobile* echo_control_mobile() const override { |
165 return NULL; | 166 return NULL; |
166 } | 167 } |
167 webrtc::GainControl* gain_control() const override { return NULL; } | 168 webrtc::GainControl* gain_control() const override { return NULL; } |
168 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | 169 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } |
169 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | 170 webrtc::LevelEstimator* level_estimator() const override { return NULL; } |
170 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | 171 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } |
(...skipping 947 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1118 int playout_sample_rate_; | 1119 int playout_sample_rate_; |
1119 DtmfInfo dtmf_info_; | 1120 DtmfInfo dtmf_info_; |
1120 FakeAudioProcessing audio_processing_; | 1121 FakeAudioProcessing audio_processing_; |
1121 }; | 1122 }; |
1122 | 1123 |
1123 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID | 1124 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID |
1124 | 1125 |
1125 } // namespace cricket | 1126 } // namespace cricket |
1126 | 1127 |
1127 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 1128 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
OLD | NEW |