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d47941e018
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
51 lines
2 KiB
C++
51 lines
2 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "test/gtest.h"
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namespace webrtc {
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TEST(AudioFrameTest, ConstructFromAudioBuffer) {
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constexpr int kSampleRateHz = 48000;
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constexpr int kNumChannels = 2;
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constexpr float kFloatConstant = 1272.f;
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constexpr float kIntConstant = 17252;
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const webrtc::StreamConfig stream_config(kSampleRateHz, kNumChannels, false);
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webrtc::AudioBuffer buffer(
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stream_config.sample_rate_hz(), stream_config.num_channels(),
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stream_config.sample_rate_hz(), stream_config.num_channels(),
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stream_config.sample_rate_hz(), stream_config.num_channels());
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AudioFrameView<float> non_const_view(buffer.channels(), buffer.num_channels(),
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buffer.num_frames());
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// Modification is allowed.
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non_const_view.channel(0)[0] = kFloatConstant;
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EXPECT_EQ(buffer.channels()[0][0], kFloatConstant);
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AudioFrameView<const float> const_view(
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buffer.channels(), buffer.num_channels(), buffer.num_frames());
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// Modification is not allowed.
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// const_view.channel(0)[0] = kFloatConstant;
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// Assignment is allowed.
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AudioFrameView<const float> other_const_view = non_const_view;
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static_cast<void>(other_const_view);
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// But not the other way. The following will fail:
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// non_const_view = other_const_view;
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AudioFrameView<float> non_const_float_view(
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buffer.channels(), buffer.num_channels(), buffer.num_frames());
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non_const_float_view.channel(0)[0] = kIntConstant;
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EXPECT_EQ(buffer.channels()[0][0], kIntConstant);
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}
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} // namespace webrtc
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