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Side by Side Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1533913004: Revert of Added option to specify a maximum file size when recording an AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years ago
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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, 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; }
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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_
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