mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00

This is a reland ofa66395e72f
Original change's description: > Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3." > > This is a reland off3a197e553
> > Original change's description: > > Add core multi-channel pipeline in AEC3 > > This CL adds basic the basic pipeline to support multi-channel > > processing in AEC3. > > > > Apart from that, it removes the 8 kHz processing support in several > > places of the AEC3 code. > > > > Bug: webrtc:10913 > > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332 > > Commit-Queue: Per Åhgren <peah@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#29017} > > Bug: webrtc:10913 > Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124 > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Commit-Queue: Per Åhgren <peah@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29034} Bug: webrtc:10913 Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29042}
48 lines
1.8 KiB
C++
48 lines
1.8 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for producing frames consisting of 2 subframes of 80 samples each
|
|
// from 64 sample blocks. The class is designed to work together with the
|
|
// FrameBlocker class which performs the reverse conversion. Used together with
|
|
// that, this class produces output frames are the same rate as frames are
|
|
// received by the FrameBlocker class. Note that the internal buffers will
|
|
// overrun if any other rate of packets insertion is used.
|
|
class BlockFramer {
|
|
public:
|
|
BlockFramer(size_t num_bands, size_t num_channels);
|
|
~BlockFramer();
|
|
BlockFramer(const BlockFramer&) = delete;
|
|
BlockFramer& operator=(const BlockFramer&) = delete;
|
|
|
|
// Adds a 64 sample block into the data that will form the next output frame.
|
|
void InsertBlock(const std::vector<std::vector<std::vector<float>>>& block);
|
|
// Adds a 64 sample block and extracts an 80 sample subframe.
|
|
void InsertBlockAndExtractSubFrame(
|
|
const std::vector<std::vector<std::vector<float>>>& block,
|
|
std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame);
|
|
|
|
private:
|
|
const size_t num_bands_;
|
|
const size_t num_channels_;
|
|
std::vector<std::vector<std::vector<float>>> buffer_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
|