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_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
12#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
13
14#include <algorithm>
15#include <vector>
16
17#include "webrtc/base/array_view.h"
18#include "webrtc/typedefs.h"
19
20namespace webrtc {
21
22// This is the interface class for encoders in AudioCoding module. Each codec
23// type must have an implementation of this class.
24class AudioEncoder {
25 public:
26  struct EncodedInfoLeaf {
27    size_t encoded_bytes = 0;
28    uint32_t encoded_timestamp = 0;
29    int payload_type = 0;
30    bool send_even_if_empty = false;
31    bool speech = true;
32  };
33
34  // This is the main struct for auxiliary encoding information. Each encoded
35  // packet should be accompanied by one EncodedInfo struct, containing the
36  // total number of |encoded_bytes|, the |encoded_timestamp| and the
37  // |payload_type|. If the packet contains redundant encodings, the |redundant|
38  // vector will be populated with EncodedInfoLeaf structs. Each struct in the
39  // vector represents one encoding; the order of structs in the vector is the
40  // same as the order in which the actual payloads are written to the byte
41  // stream. When EncoderInfoLeaf structs are present in the vector, the main
42  // struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
43  // vector.
44  struct EncodedInfo : public EncodedInfoLeaf {
45    EncodedInfo();
46    ~EncodedInfo();
47
48    std::vector<EncodedInfoLeaf> redundant;
49  };
50
51  virtual ~AudioEncoder() = default;
52
53  // Returns the maximum number of bytes that can be produced by the encoder
54  // at each Encode() call. The caller can use the return value to determine
55  // the size of the buffer that needs to be allocated. This value is allowed
56  // to depend on encoder parameters like bitrate, frame size etc., so if
57  // any of these change, the caller of Encode() is responsible for checking
58  // that the buffer is large enough by calling MaxEncodedBytes() again.
59  virtual size_t MaxEncodedBytes() const = 0;
60
61  // Returns the input sample rate in Hz and the number of input channels.
62  // These are constants set at instantiation time.
63  virtual int SampleRateHz() const = 0;
64  virtual size_t NumChannels() const = 0;
65
66  // Returns the rate at which the RTP timestamps are updated. The default
67  // implementation returns SampleRateHz().
68  virtual int RtpTimestampRateHz() const;
69
70  // Returns the number of 10 ms frames the encoder will put in the next
71  // packet. This value may only change when Encode() outputs a packet; i.e.,
72  // the encoder may vary the number of 10 ms frames from packet to packet, but
73  // it must decide the length of the next packet no later than when outputting
74  // the preceding packet.
75  virtual size_t Num10MsFramesInNextPacket() const = 0;
76
77  // Returns the maximum value that can be returned by
78  // Num10MsFramesInNextPacket().
79  virtual size_t Max10MsFramesInAPacket() const = 0;
80
81  // Returns the current target bitrate in bits/s. The value -1 means that the
82  // codec adapts the target automatically, and a current target cannot be
83  // provided.
84  virtual int GetTargetBitrate() const = 0;
85
86  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
87  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
88  // The encoder produces zero or more bytes of output in |encoded| and
89  // returns additional encoding information.
90  // The caller is responsible for making sure that |max_encoded_bytes| is
91  // not smaller than the number of bytes actually produced by the encoder.
92  // Encode() checks some preconditions, calls EncodeInternal() which does the
93  // actual work, and then checks some postconditions.
94  EncodedInfo Encode(uint32_t rtp_timestamp,
95                     rtc::ArrayView<const int16_t> audio,
96                     size_t max_encoded_bytes,
97                     uint8_t* encoded);
98
99  virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
100                                     rtc::ArrayView<const int16_t> audio,
101                                     size_t max_encoded_bytes,
102                                     uint8_t* encoded) = 0;
103
104  // Resets the encoder to its starting state, discarding any input that has
105  // been fed to the encoder but not yet emitted in a packet.
106  virtual void Reset() = 0;
107
108  // Enables or disables codec-internal FEC (forward error correction). Returns
109  // true if the codec was able to comply. The default implementation returns
110  // true when asked to disable FEC and false when asked to enable it (meaning
111  // that FEC isn't supported).
112  virtual bool SetFec(bool enable);
113
114  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
115  // able to comply. The default implementation returns true when asked to
116  // disable DTX and false when asked to enable it (meaning that DTX isn't
117  // supported).
118  virtual bool SetDtx(bool enable);
119
120  // Sets the application mode. Returns true if the codec was able to comply.
121  // The default implementation just returns false.
122  enum class Application { kSpeech, kAudio };
123  virtual bool SetApplication(Application application);
124
125  // Tells the encoder about the highest sample rate the decoder is expected to
126  // use when decoding the bitstream. The encoder would typically use this
127  // information to adjust the quality of the encoding. The default
128  // implementation does nothing.
129  virtual void SetMaxPlaybackRate(int frequency_hz);
130
131  // Tells the encoder what the projected packet loss rate is. The rate is in
132  // the range [0.0, 1.0]. The encoder would typically use this information to
133  // adjust channel coding efforts, such as FEC. The default implementation
134  // does nothing.
135  virtual void SetProjectedPacketLossRate(double fraction);
136
137  // Tells the encoder what average bitrate we'd like it to produce. The
138  // encoder is free to adjust or disregard the given bitrate (the default
139  // implementation does the latter).
140  virtual void SetTargetBitrate(int target_bps);
141};
142}  // namespace webrtc
143#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
144