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On top of adding unittests for the remixing, the CL moves the code tested to a separate file in order to allow it to be tested. Bug: webrtc:11007 Change-Id: I531736517bbcc715b3c1bf3a4256c42208c5b778 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155740 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29839}
34 lines
1.3 KiB
C++
34 lines
1.3 KiB
C++
/*
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* Copyright (c) 2019 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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#ifndef MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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#include <vector>
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#include "api/audio/audio_frame.h"
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namespace webrtc {
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// Stereo-to-mono downmixing. The length of the output must equal to the number
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// of samples per channel in the input.
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void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output);
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// Remixes the interleaved input frame to an interleaved output data vector. The
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// remixed data replaces the data in the output vector which is resized if
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// needed. The remixing supports any combination of input and output channels,
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// as well as any number of samples per channel.
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void ReMixFrame(const AudioFrame& input,
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size_t num_output_channels,
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std::vector<int16_t>* output);
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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