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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 FakeAudioProcessing() : experimental_ns_enabled_(false) {} | 105 FakeAudioProcessing() : experimental_ns_enabled_(false) {} |
106 | 106 |
107 WEBRTC_STUB(Initialize, ()) | 107 WEBRTC_STUB(Initialize, ()) |
108 WEBRTC_STUB(Initialize, ( | 108 WEBRTC_STUB(Initialize, ( |
109 int input_sample_rate_hz, | 109 int input_sample_rate_hz, |
110 int output_sample_rate_hz, | 110 int output_sample_rate_hz, |
111 int reverse_sample_rate_hz, | 111 int reverse_sample_rate_hz, |
112 webrtc::AudioProcessing::ChannelLayout input_layout, | 112 webrtc::AudioProcessing::ChannelLayout input_layout, |
113 webrtc::AudioProcessing::ChannelLayout output_layout, | 113 webrtc::AudioProcessing::ChannelLayout output_layout, |
114 webrtc::AudioProcessing::ChannelLayout reverse_layout)); | 114 webrtc::AudioProcessing::ChannelLayout reverse_layout)); |
115 WEBRTC_STUB(Initialize, ( | |
116 const webrtc::ProcessingConfig& processing_config)); | |
117 | 115 |
118 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { | 116 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { |
119 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; | 117 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; |
120 } | 118 } |
121 | 119 |
122 WEBRTC_STUB(set_sample_rate_hz, (int rate)); | 120 WEBRTC_STUB(set_sample_rate_hz, (int rate)); |
123 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); | 121 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); |
124 WEBRTC_STUB_CONST(sample_rate_hz, ()); | 122 WEBRTC_STUB_CONST(sample_rate_hz, ()); |
125 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); | 123 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); |
126 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); | 124 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); |
127 WEBRTC_STUB_CONST(num_input_channels, ()); | 125 WEBRTC_STUB_CONST(num_input_channels, ()); |
128 WEBRTC_STUB_CONST(num_output_channels, ()); | 126 WEBRTC_STUB_CONST(num_output_channels, ()); |
129 WEBRTC_STUB_CONST(num_reverse_channels, ()); | 127 WEBRTC_STUB_CONST(num_reverse_channels, ()); |
130 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); | 128 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); |
131 WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ()); | 129 WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ()); |
132 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); | 130 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); |
133 WEBRTC_STUB(ProcessStream, ( | 131 WEBRTC_STUB(ProcessStream, ( |
134 const float* const* src, | 132 const float* const* src, |
135 int samples_per_channel, | 133 int samples_per_channel, |
136 int input_sample_rate_hz, | 134 int input_sample_rate_hz, |
137 webrtc::AudioProcessing::ChannelLayout input_layout, | 135 webrtc::AudioProcessing::ChannelLayout input_layout, |
138 int output_sample_rate_hz, | 136 int output_sample_rate_hz, |
139 webrtc::AudioProcessing::ChannelLayout output_layout, | 137 webrtc::AudioProcessing::ChannelLayout output_layout, |
140 float* const* dest)); | 138 float* const* dest)); |
141 WEBRTC_STUB(ProcessStream, | |
142 (const float* const* src, | |
143 const webrtc::StreamConfig& input_config, | |
144 const webrtc::StreamConfig& output_config, | |
145 float* const* dest)); | |
146 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); | 139 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); |
147 WEBRTC_STUB(AnalyzeReverseStream, ( | 140 WEBRTC_STUB(AnalyzeReverseStream, ( |
148 const float* const* data, | 141 const float* const* data, |
149 int samples_per_channel, | 142 int samples_per_channel, |
150 int sample_rate_hz, | 143 int sample_rate_hz, |
151 webrtc::AudioProcessing::ChannelLayout layout)); | 144 webrtc::AudioProcessing::ChannelLayout layout)); |
152 WEBRTC_STUB(AnalyzeReverseStream, ( | |
153 const float* const* data, | |
154 const webrtc::StreamConfig& reverse_config)); | |
155 WEBRTC_STUB(set_stream_delay_ms, (int delay)); | 145 WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
156 WEBRTC_STUB_CONST(stream_delay_ms, ()); | 146 WEBRTC_STUB_CONST(stream_delay_ms, ()); |
157 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | 147 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |
158 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); | 148 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); |
159 WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ()); | 149 WEBRTC_BOOL_STUB_CONST(stream_key_pressed, ()); |
160 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); | 150 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); |
161 WEBRTC_STUB_CONST(delay_offset_ms, ()); | 151 WEBRTC_STUB_CONST(delay_offset_ms, ()); |
162 WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); | 152 WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); |
163 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); | 153 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); |
164 WEBRTC_STUB(StopDebugRecording, ()); | 154 WEBRTC_STUB(StopDebugRecording, ()); |
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1280 DtmfInfo dtmf_info_; | 1270 DtmfInfo dtmf_info_; |
1281 webrtc::VoEMediaProcess* media_processor_; | 1271 webrtc::VoEMediaProcess* media_processor_; |
1282 FakeAudioProcessing audio_processing_; | 1272 FakeAudioProcessing audio_processing_; |
1283 }; | 1273 }; |
1284 | 1274 |
1285 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID | 1275 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID |
1286 | 1276 |
1287 } // namespace cricket | 1277 } // namespace cricket |
1288 | 1278 |
1289 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 1279 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
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