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 |