Line data Source code
1 : /*
2 : * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 : *
4 : * Use of this source code is governed by a BSD-style license
5 : * that can be found in the LICENSE file in the root of the source
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : */
10 :
11 : #ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
12 : #define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
13 :
14 : #include <memory>
15 :
16 : #include "webrtc/common_audio/audio_ring_buffer.h"
17 : #include "webrtc/common_audio/channel_buffer.h"
18 :
19 : namespace webrtc {
20 :
21 : // The callback function to process audio in the time domain. Input has already
22 : // been windowed, and output will be windowed. The number of input channels
23 : // must be >= the number of output channels.
24 0 : class BlockerCallback {
25 : public:
26 0 : virtual ~BlockerCallback() {}
27 :
28 : virtual void ProcessBlock(const float* const* input,
29 : size_t num_frames,
30 : size_t num_input_channels,
31 : size_t num_output_channels,
32 : float* const* output) = 0;
33 : };
34 :
35 : // The main purpose of Blocker is to abstract away the fact that often we
36 : // receive a different number of audio frames than our transform takes. For
37 : // example, most FFTs work best when the fft-size is a power of 2, but suppose
38 : // we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
39 : // of audio, which is not a power of 2. Blocker allows us to specify the
40 : // transform and all other necessary processing via the Process() callback
41 : // function without any constraints on the transform-size
42 : // (read: |block_size_|) or received-audio-size (read: |chunk_size_|).
43 : // We handle this for the multichannel audio case, allowing for different
44 : // numbers of input and output channels (for example, beamforming takes 2 or
45 : // more input channels and returns 1 output channel). Audio signals are
46 : // represented as deinterleaved floats in the range [-1, 1].
47 : //
48 : // Blocker is responsible for:
49 : // - blocking audio while handling potential discontinuities on the edges
50 : // of chunks
51 : // - windowing blocks before sending them to Process()
52 : // - windowing processed blocks, and overlap-adding them together before
53 : // sending back a processed chunk
54 : //
55 : // To use blocker:
56 : // 1. Impelment a BlockerCallback object |bc|.
57 : // 2. Instantiate a Blocker object |b|, passing in |bc|.
58 : // 3. As you receive audio, call b.ProcessChunk() to get processed audio.
59 : //
60 : // A small amount of delay is added to the first received chunk to deal with
61 : // the difference in chunk/block sizes. This delay is <= chunk_size.
62 : //
63 : // Ownership of window is retained by the caller. That is, Blocker makes a
64 : // copy of window and does not attempt to delete it.
65 0 : class Blocker {
66 : public:
67 : Blocker(size_t chunk_size,
68 : size_t block_size,
69 : size_t num_input_channels,
70 : size_t num_output_channels,
71 : const float* window,
72 : size_t shift_amount,
73 : BlockerCallback* callback);
74 : ~Blocker();
75 :
76 : void ProcessChunk(const float* const* input,
77 : size_t chunk_size,
78 : size_t num_input_channels,
79 : size_t num_output_channels,
80 : float* const* output);
81 :
82 : size_t initial_delay() const { return initial_delay_; }
83 :
84 : private:
85 : const size_t chunk_size_;
86 : const size_t block_size_;
87 : const size_t num_input_channels_;
88 : const size_t num_output_channels_;
89 :
90 : // The number of frames of delay to add at the beginning of the first chunk.
91 : const size_t initial_delay_;
92 :
93 : // The frame index into the input buffer where the first block should be read
94 : // from. This is necessary because shift_amount_ is not necessarily a
95 : // multiple of chunk_size_, so blocks won't line up at the start of the
96 : // buffer.
97 : size_t frame_offset_;
98 :
99 : // Since blocks nearly always overlap, there are certain blocks that require
100 : // frames from the end of one chunk and the beginning of the next chunk. The
101 : // input and output buffers are responsible for saving those frames between
102 : // calls to ProcessChunk().
103 : //
104 : // Both contain |initial delay| + |chunk_size| frames. The input is a fairly
105 : // standard FIFO, but due to the overlap-add it's harder to use an
106 : // AudioRingBuffer for the output.
107 : AudioRingBuffer input_buffer_;
108 : ChannelBuffer<float> output_buffer_;
109 :
110 : // Space for the input block (can't wrap because of windowing).
111 : ChannelBuffer<float> input_block_;
112 :
113 : // Space for the output block (can't wrap because of overlap/add).
114 : ChannelBuffer<float> output_block_;
115 :
116 : std::unique_ptr<float[]> window_;
117 :
118 : // The amount of frames between the start of contiguous blocks. For example,
119 : // |shift_amount_| = |block_size_| / 2 for a Hann window.
120 : size_t shift_amount_;
121 :
122 : BlockerCallback* callback_;
123 : };
124 :
125 : } // namespace webrtc
126 :
127 : #endif // WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
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