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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
| 11 // This file is for backwards compatibility only! Use |
| 12 // webrtc/api/audio_codecs/audio_decoder.h instead! |
| 13 // TODO(kwiberg): Remove it. |
| 14 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ | 15 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ | 16 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ |
13 | 17 |
14 #include <memory> | 18 #include "webrtc/api/audio_codecs/audio_decoder.h" |
15 #include <vector> | |
16 | 19 |
17 #include "webrtc/base/array_view.h" | |
18 #include "webrtc/base/buffer.h" | |
19 #include "webrtc/base/constructormagic.h" | |
20 #include "webrtc/base/optional.h" | |
21 #include "webrtc/typedefs.h" | |
22 | |
23 namespace webrtc { | |
24 | |
25 // This is the interface class for decoders in NetEQ. Each codec type will have | |
26 // and implementation of this class. | |
27 class AudioDecoder { | |
28 public: | |
29 enum SpeechType { | |
30 kSpeech = 1, | |
31 kComfortNoise = 2 | |
32 }; | |
33 | |
34 // Used by PacketDuration below. Save the value -1 for errors. | |
35 enum { kNotImplemented = -2 }; | |
36 | |
37 AudioDecoder() = default; | |
38 virtual ~AudioDecoder() = default; | |
39 | |
40 class EncodedAudioFrame { | |
41 public: | |
42 struct DecodeResult { | |
43 size_t num_decoded_samples; | |
44 SpeechType speech_type; | |
45 }; | |
46 | |
47 virtual ~EncodedAudioFrame() = default; | |
48 | |
49 // Returns the duration in samples-per-channel of this audio frame. | |
50 // If no duration can be ascertained, returns zero. | |
51 virtual size_t Duration() const = 0; | |
52 | |
53 // Decodes this frame of audio and writes the result in |decoded|. | |
54 // |decoded| must be large enough to store as many samples as indicated by a | |
55 // call to Duration() . On success, returns an rtc::Optional containing the | |
56 // total number of samples across all channels, as well as whether the | |
57 // decoder produced comfort noise or speech. On failure, returns an empty | |
58 // rtc::Optional. Decode may be called at most once per frame object. | |
59 virtual rtc::Optional<DecodeResult> Decode( | |
60 rtc::ArrayView<int16_t> decoded) const = 0; | |
61 }; | |
62 | |
63 struct ParseResult { | |
64 ParseResult(); | |
65 ParseResult(uint32_t timestamp, | |
66 int priority, | |
67 std::unique_ptr<EncodedAudioFrame> frame); | |
68 ParseResult(ParseResult&& b); | |
69 ~ParseResult(); | |
70 | |
71 ParseResult& operator=(ParseResult&& b); | |
72 | |
73 // The timestamp of the frame is in samples per channel. | |
74 uint32_t timestamp; | |
75 // The relative priority of the frame compared to other frames of the same | |
76 // payload and the same timeframe. A higher value means a lower priority. | |
77 // The highest priority is zero - negative values are not allowed. | |
78 int priority; | |
79 std::unique_ptr<EncodedAudioFrame> frame; | |
80 }; | |
81 | |
82 // Let the decoder parse this payload and prepare zero or more decodable | |
83 // frames. Each frame must be between 10 ms and 120 ms long. The caller must | |
84 // ensure that the AudioDecoder object outlives any frame objects returned by | |
85 // this call. The decoder is free to swap or move the data from the |payload| | |
86 // buffer. |timestamp| is the input timestamp, in samples, corresponding to | |
87 // the start of the payload. | |
88 virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, | |
89 uint32_t timestamp); | |
90 | |
91 // Decodes |encode_len| bytes from |encoded| and writes the result in | |
92 // |decoded|. The maximum bytes allowed to be written into |decoded| is | |
93 // |max_decoded_bytes|. Returns the total number of samples across all | |
94 // channels. If the decoder produced comfort noise, |speech_type| | |
95 // is set to kComfortNoise, otherwise it is kSpeech. The desired output | |
96 // sample rate is provided in |sample_rate_hz|, which must be valid for the | |
97 // codec at hand. | |
98 int Decode(const uint8_t* encoded, | |
99 size_t encoded_len, | |
100 int sample_rate_hz, | |
101 size_t max_decoded_bytes, | |
102 int16_t* decoded, | |
103 SpeechType* speech_type); | |
104 | |
105 // Same as Decode(), but interfaces to the decoders redundant decode function. | |
106 // The default implementation simply calls the regular Decode() method. | |
107 int DecodeRedundant(const uint8_t* encoded, | |
108 size_t encoded_len, | |
109 int sample_rate_hz, | |
110 size_t max_decoded_bytes, | |
111 int16_t* decoded, | |
112 SpeechType* speech_type); | |
113 | |
114 // Indicates if the decoder implements the DecodePlc method. | |
115 virtual bool HasDecodePlc() const; | |
116 | |
117 // Calls the packet-loss concealment of the decoder to update the state after | |
118 // one or several lost packets. The caller has to make sure that the | |
119 // memory allocated in |decoded| should accommodate |num_frames| frames. | |
120 virtual size_t DecodePlc(size_t num_frames, int16_t* decoded); | |
121 | |
122 // Resets the decoder state (empty buffers etc.). | |
123 virtual void Reset() = 0; | |
124 | |
125 // Notifies the decoder of an incoming packet to NetEQ. | |
126 virtual int IncomingPacket(const uint8_t* payload, | |
127 size_t payload_len, | |
128 uint16_t rtp_sequence_number, | |
129 uint32_t rtp_timestamp, | |
130 uint32_t arrival_timestamp); | |
131 | |
132 // Returns the last error code from the decoder. | |
133 virtual int ErrorCode(); | |
134 | |
135 // Returns the duration in samples-per-channel of the payload in |encoded| | |
136 // which is |encoded_len| bytes long. Returns kNotImplemented if no duration | |
137 // estimate is available, or -1 in case of an error. | |
138 virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const; | |
139 | |
140 // Returns the duration in samples-per-channel of the redandant payload in | |
141 // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no | |
142 // duration estimate is available, or -1 in case of an error. | |
143 virtual int PacketDurationRedundant(const uint8_t* encoded, | |
144 size_t encoded_len) const; | |
145 | |
146 // Detects whether a packet has forward error correction. The packet is | |
147 // comprised of the samples in |encoded| which is |encoded_len| bytes long. | |
148 // Returns true if the packet has FEC and false otherwise. | |
149 virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; | |
150 | |
151 // Returns the actual sample rate of the decoder's output. This value may not | |
152 // change during the lifetime of the decoder. | |
153 virtual int SampleRateHz() const = 0; | |
154 | |
155 // The number of channels in the decoder's output. This value may not change | |
156 // during the lifetime of the decoder. | |
157 virtual size_t Channels() const = 0; | |
158 | |
159 protected: | |
160 static SpeechType ConvertSpeechType(int16_t type); | |
161 | |
162 virtual int DecodeInternal(const uint8_t* encoded, | |
163 size_t encoded_len, | |
164 int sample_rate_hz, | |
165 int16_t* decoded, | |
166 SpeechType* speech_type) = 0; | |
167 | |
168 virtual int DecodeRedundantInternal(const uint8_t* encoded, | |
169 size_t encoded_len, | |
170 int sample_rate_hz, | |
171 int16_t* decoded, | |
172 SpeechType* speech_type); | |
173 | |
174 private: | |
175 RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder); | |
176 }; | |
177 | |
178 } // namespace webrtc | |
179 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ | 20 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_ |
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