1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc
new file mode 100644
index 0000000000..458f6ced76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2016 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 "modules/audio_processing/test/simulator_buffers.h"
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz,
+                                   int capture_input_sample_rate_hz,
+                                   int render_output_sample_rate_hz,
+                                   int capture_output_sample_rate_hz,
+                                   size_t num_render_input_channels,
+                                   size_t num_capture_input_channels,
+                                   size_t num_render_output_channels,
+                                   size_t num_capture_output_channels) {
+  Random rand_gen(42);
+  CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels,
+                        &rand_gen, &render_input_buffer, &render_input_config,
+                        &render_input, &render_input_samples);
+
+  CreateConfigAndBuffer(render_output_sample_rate_hz,
+                        num_render_output_channels, &rand_gen,
+                        &render_output_buffer, &render_output_config,
+                        &render_output, &render_output_samples);
+
+  CreateConfigAndBuffer(capture_input_sample_rate_hz,
+                        num_capture_input_channels, &rand_gen,
+                        &capture_input_buffer, &capture_input_config,
+                        &capture_input, &capture_input_samples);
+
+  CreateConfigAndBuffer(capture_output_sample_rate_hz,
+                        num_capture_output_channels, &rand_gen,
+                        &capture_output_buffer, &capture_output_config,
+                        &capture_output, &capture_output_samples);
+
+  UpdateInputBuffers();
+}
+
+SimulatorBuffers::~SimulatorBuffers() = default;
+
+void SimulatorBuffers::CreateConfigAndBuffer(
+    int sample_rate_hz,
+    size_t num_channels,
+    Random* rand_gen,
+    std::unique_ptr<AudioBuffer>* buffer,
+    StreamConfig* config,
+    std::vector<float*>* buffer_data,
+    std::vector<float>* buffer_data_samples) {
+  int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
+  *config = StreamConfig(sample_rate_hz, num_channels);
+  buffer->reset(
+      new AudioBuffer(config->sample_rate_hz(), config->num_channels(),
+                      config->sample_rate_hz(), config->num_channels(),
+                      config->sample_rate_hz(), config->num_channels()));
+
+  buffer_data_samples->resize(samples_per_channel * num_channels);
+  for (auto& v : *buffer_data_samples) {
+    v = rand_gen->Rand<float>();
+  }
+
+  buffer_data->resize(num_channels);
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    (*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
+  }
+}
+
+void SimulatorBuffers::UpdateInputBuffers() {
+  test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
+                                capture_input_buffer.get());
+  test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
+                                render_input_buffer.get());
+}
+
+}  // namespace test
+}  // namespace webrtc