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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc')
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diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc
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+++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2018 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/include/audio_frame_view.h"
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+TEST(AudioFrameTest, ConstructFromAudioBuffer) {
+ constexpr int kSampleRateHz = 48000;
+ constexpr int kNumChannels = 2;
+ constexpr float kFloatConstant = 1272.f;
+ constexpr float kIntConstant = 17252;
+ const webrtc::StreamConfig stream_config(kSampleRateHz, kNumChannels);
+ webrtc::AudioBuffer buffer(
+ stream_config.sample_rate_hz(), stream_config.num_channels(),
+ stream_config.sample_rate_hz(), stream_config.num_channels(),
+ stream_config.sample_rate_hz(), stream_config.num_channels());
+
+ AudioFrameView<float> non_const_view(buffer.channels(), buffer.num_channels(),
+ buffer.num_frames());
+ // Modification is allowed.
+ non_const_view.channel(0)[0] = kFloatConstant;
+ EXPECT_EQ(buffer.channels()[0][0], kFloatConstant);
+
+ AudioFrameView<const float> const_view(
+ buffer.channels(), buffer.num_channels(), buffer.num_frames());
+ // Modification is not allowed.
+ // const_view.channel(0)[0] = kFloatConstant;
+
+ // Assignment is allowed.
+ AudioFrameView<const float> other_const_view = non_const_view;
+ static_cast<void>(other_const_view);
+
+ // But not the other way. The following will fail:
+ // non_const_view = other_const_view;
+
+ AudioFrameView<float> non_const_float_view(
+ buffer.channels(), buffer.num_channels(), buffer.num_frames());
+ non_const_float_view.channel(0)[0] = kIntConstant;
+ EXPECT_EQ(buffer.channels()[0][0], kIntConstant);
+}
+} // namespace webrtc