Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 105 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc
new file mode 100644
index 0000000000..011ab49651
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2017 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/aec3/block_delay_buffer.h"
+
+#include <string>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+float SampleValue(size_t sample_index) {
+  return sample_index % 32768;
+}
+
+// Populates the frame with linearly increasing sample values for each band.
+void PopulateInputFrame(size_t frame_length,
+                        size_t num_bands,
+                        size_t first_sample_index,
+                        float* const* frame) {
+  for (size_t k = 0; k < num_bands; ++k) {
+    for (size_t i = 0; i < frame_length; ++i) {
+      frame[k][i] = SampleValue(first_sample_index + i);
+    }
+  }
+}
+
+std::string ProduceDebugText(int sample_rate_hz, size_t delay) {
+  char log_stream_buffer[8 * 1024];
+  rtc::SimpleStringBuilder ss(log_stream_buffer);
+  ss << "Sample rate: " << sample_rate_hz;
+  ss << ", Delay: " << delay;
+  return ss.str();
+}
+
+}  // namespace
+
+class BlockDelayBufferTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, int, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    ParameterCombinations,
+    BlockDelayBufferTest,
+    ::testing::Combine(::testing::Values(0, 1, 27, 160, 4321, 7021),
+                       ::testing::Values(16000, 32000, 48000),
+                       ::testing::Values(1, 2, 4)));
+
+// Verifies that the correct signal delay is achived.
+TEST_P(BlockDelayBufferTest, CorrectDelayApplied) {
+  const size_t delay = std::get<0>(GetParam());
+  const int rate = std::get<1>(GetParam());
+  const size_t num_channels = std::get<2>(GetParam());
+
+  SCOPED_TRACE(ProduceDebugText(rate, delay));
+  size_t num_bands = NumBandsForRate(rate);
+  size_t subband_frame_length = 160;
+
+  BlockDelayBuffer delay_buffer(num_channels, num_bands, subband_frame_length,
+                                delay);
+
+  static constexpr size_t kNumFramesToProcess = 20;
+  for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+       ++frame_index) {
+    AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate,
+                             num_channels);
+    if (rate > 16000) {
+      audio_buffer.SplitIntoFrequencyBands();
+    }
+    size_t first_sample_index = frame_index * subband_frame_length;
+    for (size_t ch = 0; ch < num_channels; ++ch) {
+      PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
+                         &audio_buffer.split_bands(ch)[0]);
+    }
+    delay_buffer.DelaySignal(&audio_buffer);
+
+    for (size_t ch = 0; ch < num_channels; ++ch) {
+      for (size_t band = 0; band < num_bands; ++band) {
+        size_t sample_index = first_sample_index;
+        for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) {
+          if (sample_index < delay) {
+            EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]);
+          } else {
+            EXPECT_EQ(SampleValue(sample_index - delay),
+                      audio_buffer.split_bands(ch)[band][i]);
+          }
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc