| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_file_processor.h | 101 const StreamConfig output_config_; member in class:webrtc::final
133 const StreamConfig output_config_; member in class:webrtc::final
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| debug_dump_test.cc | 91 StreamConfig output_config_; member in class:webrtc::test::__anon27885::DebugDumpGenerator
123 output_config_(input_rate_hz, input_channels),
132 output_(new ChannelBuffer<float>(output_config_.num_frames(),
133 output_config_.num_channels())),
174 output_config_.set_sample_rate_hz(rate_hz);
175 MaybeResetBuffer(&output_, output_config_);
179 output_config_.set_num_channels(channels);
180 MaybeResetBuffer(&output_, output_config_);
197 output_config_, output_->channels()));
261 StreamConfig output_config_; member in class:webrtc::test::DebugDumpTest
[all...] |
| audio_file_processor.cc | 51 output_config_(GetStreamConfig(*out_file)),
63 output_config_, out_buf_.channels()));
75 output_config_(GetStreamConfig(*out_file)),
129 {input_config_, output_config_, reverse_config_, reverse_config_}};
153 output_config_, out_buf_.channels()));
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