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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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125 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); | 125 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); |
126 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); | 126 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); |
127 WEBRTC_STUB_CONST(num_input_channels, ()); | 127 WEBRTC_STUB_CONST(num_input_channels, ()); |
128 WEBRTC_STUB_CONST(num_output_channels, ()); | 128 WEBRTC_STUB_CONST(num_output_channels, ()); |
129 WEBRTC_STUB_CONST(num_reverse_channels, ()); | 129 WEBRTC_STUB_CONST(num_reverse_channels, ()); |
130 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); | 130 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); |
131 WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ()); | 131 WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ()); |
132 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); | 132 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); |
133 WEBRTC_STUB(ProcessStream, ( | 133 WEBRTC_STUB(ProcessStream, ( |
134 const float* const* src, | 134 const float* const* src, |
135 int samples_per_channel, | 135 size_t samples_per_channel, |
136 int input_sample_rate_hz, | 136 int input_sample_rate_hz, |
137 webrtc::AudioProcessing::ChannelLayout input_layout, | 137 webrtc::AudioProcessing::ChannelLayout input_layout, |
138 int output_sample_rate_hz, | 138 int output_sample_rate_hz, |
139 webrtc::AudioProcessing::ChannelLayout output_layout, | 139 webrtc::AudioProcessing::ChannelLayout output_layout, |
140 float* const* dest)); | 140 float* const* dest)); |
141 WEBRTC_STUB(ProcessStream, | 141 WEBRTC_STUB(ProcessStream, |
142 (const float* const* src, | 142 (const float* const* src, |
143 const webrtc::StreamConfig& input_config, | 143 const webrtc::StreamConfig& input_config, |
144 const webrtc::StreamConfig& output_config, | 144 const webrtc::StreamConfig& output_config, |
145 float* const* dest)); | 145 float* const* dest)); |
146 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); | 146 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); |
147 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); | 147 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); |
148 WEBRTC_STUB(AnalyzeReverseStream, ( | 148 WEBRTC_STUB(AnalyzeReverseStream, ( |
149 const float* const* data, | 149 const float* const* data, |
150 int samples_per_channel, | 150 size_t samples_per_channel, |
151 int sample_rate_hz, | 151 int sample_rate_hz, |
152 webrtc::AudioProcessing::ChannelLayout layout)); | 152 webrtc::AudioProcessing::ChannelLayout layout)); |
153 WEBRTC_STUB(ProcessReverseStream, | 153 WEBRTC_STUB(ProcessReverseStream, |
154 (const float* const* src, | 154 (const float* const* src, |
155 const webrtc::StreamConfig& reverse_input_config, | 155 const webrtc::StreamConfig& reverse_input_config, |
156 const webrtc::StreamConfig& reverse_output_config, | 156 const webrtc::StreamConfig& reverse_output_config, |
157 float* const* dest)); | 157 float* const* dest)); |
158 WEBRTC_STUB(set_stream_delay_ms, (int delay)); | 158 WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
159 WEBRTC_STUB_CONST(stream_delay_ms, ()); | 159 WEBRTC_STUB_CONST(stream_delay_ms, ()); |
160 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); | 160 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |
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1283 DtmfInfo dtmf_info_; | 1283 DtmfInfo dtmf_info_; |
1284 webrtc::VoEMediaProcess* media_processor_; | 1284 webrtc::VoEMediaProcess* media_processor_; |
1285 FakeAudioProcessing audio_processing_; | 1285 FakeAudioProcessing audio_processing_; |
1286 }; | 1286 }; |
1287 | 1287 |
1288 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID | 1288 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID |
1289 | 1289 |
1290 } // namespace cricket | 1290 } // namespace cricket |
1291 | 1291 |
1292 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | 1292 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ |
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