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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc | |
parent | Initial commit. (diff) | |
download | firefox-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')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc | 51 |
1 files changed, 51 insertions, 0 deletions
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 new file mode 100644 index 0000000000..fd25bc3b0b --- /dev/null +++ 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 |