Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 93 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc
new file mode 100644
index 0000000000..f3b2ddc689
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  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/audio_buffer.h"
+
+#include <cmath>
+
+#include "test/gtest.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 48000u;
+const size_t kStereo = 2u;
+const size_t kMono = 1u;
+
+void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
+  EXPECT_EQ(ab.num_channels(), num_channels);
+}
+
+}  // namespace
+
+TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
+  AudioBuffer ab(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz,
+                 kStereo);
+  ExpectNumChannels(ab, kStereo);
+  ab.set_num_channels(1);
+  ExpectNumChannels(ab, kMono);
+  ab.RestoreNumChannels();
+  ExpectNumChannels(ab, kStereo);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(AudioBufferDeathTest, SetNumChannelsDeathTest) {
+  AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz,
+                 kMono);
+  RTC_EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
+}
+#endif
+
+TEST(AudioBufferTest, CopyWithoutResampling) {
+  AudioBuffer ab1(32000, 2, 32000, 2, 32000, 2);
+  AudioBuffer ab2(32000, 2, 32000, 2, 32000, 2);
+  // Fill first buffer.
+  for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab1.num_frames(); ++i) {
+      ab1.channels()[ch][i] = i + ch;
+    }
+  }
+  // Copy to second buffer.
+  ab1.CopyTo(&ab2);
+  // Verify content of second buffer.
+  for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab2.num_frames(); ++i) {
+      EXPECT_EQ(ab2.channels()[ch][i], i + ch);
+    }
+  }
+}
+
+TEST(AudioBufferTest, CopyWithResampling) {
+  AudioBuffer ab1(32000, 2, 32000, 2, 48000, 2);
+  AudioBuffer ab2(48000, 2, 48000, 2, 48000, 2);
+  float energy_ab1 = 0.f;
+  float energy_ab2 = 0.f;
+  const float pi = std::acos(-1.f);
+  // Put a sine and compute energy of first buffer.
+  for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab1.num_frames(); ++i) {
+      ab1.channels()[ch][i] = std::sin(2 * pi * 100.f / 32000.f * i);
+      energy_ab1 += ab1.channels()[ch][i] * ab1.channels()[ch][i];
+    }
+  }
+  // Copy to second buffer.
+  ab1.CopyTo(&ab2);
+  // Compute energy of second buffer.
+  for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab2.num_frames(); ++i) {
+      energy_ab2 += ab2.channels()[ch][i] * ab2.channels()[ch][i];
+    }
+  }
+  // Verify that energies match.
+  EXPECT_NEAR(energy_ab1, energy_ab2 * 32000.f / 48000.f, .01f * energy_ab1);
+}
+}  // namespace webrtc