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Unified Diff: webrtc/modules/audio_processing/test/audio_buffer_tools.cc

Issue 1510493004: Bitexactness test for the highpass filter (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Reverted changes to audio_processing_impl.cc Created 4 years, 10 months ago
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Index: webrtc/modules/audio_processing/test/audio_buffer_tools.cc
diff --git a/webrtc/modules/audio_processing/test/audio_buffer_tools.cc b/webrtc/modules/audio_processing/test/audio_buffer_tools.cc
new file mode 100644
index 0000000000000000000000000000000000000000..a8cb09ca6a4b7a3da898163b647c8d42e9ce7d5f
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/audio_buffer_tools.cc
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
+
+namespace webrtc {
+namespace test {
+
+void SetupFrame(StreamConfig stream_config,
+ std::vector<float*>* frame,
+ std::vector<float>* frame_samples) {
+ frame_samples->resize(stream_config.num_channels() *
+ stream_config.num_frames());
+ frame->resize(stream_config.num_channels());
+ for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) {
+ (*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
+ }
+}
+
+void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
+ const std::vector<float>& source,
+ AudioBuffer* destination) {
+ std::vector<float*> input;
+ std::vector<float> input_samples;
+
+ SetupFrame(stream_config, &input, &input_samples);
+
+ RTC_DCHECK_EQ(input_samples.size(), source.size());
+ input_samples = source;
+
+ destination->CopyFrom(&input[0], stream_config);
+}
+
+std::vector<float> ExtractVectorFromAudioBuffer(
+ const StreamConfig& stream_config,
+ AudioBuffer* source) {
+ std::vector<float*> output;
+ std::vector<float> output_samples;
+
+ SetupFrame(stream_config, &output, &output_samples);
+
+ source->CopyTo(stream_config, &output[0]);
+
+ return output_samples;
+}
+
+} // namespace test
+} // namespace webrtc
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