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  /external/webrtc/webrtc/modules/audio_processing/test/
debug_dump_test.cc 89 StreamConfig input_config_; member in class:webrtc::test::__anon27047::DebugDumpGenerator
121 : input_config_(input_rate_hz, input_channels),
128 input_(new ChannelBuffer<float>(input_config_.num_frames(),
129 input_config_.num_channels())),
151 input_config_.set_sample_rate_hz(rate_hz);
152 MaybeResetBuffer(&input_, input_config_);
157 input_config_.set_num_channels(channels);
158 MaybeResetBuffer(&input_, input_config_);
191 ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
196 apm_->ProcessStream(input_->channels(), input_config_,
259 StreamConfig input_config_; member in class:webrtc::test::DebugDumpTest
    [all...]
audio_file_processor.h 100 const StreamConfig input_config_; member in class:webrtc::final
131 StreamConfig input_config_; member in class:webrtc::final
audio_file_processor.cc 50 input_config_(GetStreamConfig(*in_file)),
62 ap_->ProcessStream(in_buf_.channels(), input_config_,
124 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
129 {input_config_, output_config_, reverse_config_, reverse_config_}};
152 ap_->ProcessStream(in_buf_->channels(), input_config_,

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