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Side by Side Diff: webrtc/modules/audio_coding/codecs/audio_decoder.h

Issue 2029273003: AudioDecoder: Document that the sample rate and number of channels is constant (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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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
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86 // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no 86 // |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
87 // duration estimate is available, or -1 in case of an error. 87 // duration estimate is available, or -1 in case of an error.
88 virtual int PacketDurationRedundant(const uint8_t* encoded, 88 virtual int PacketDurationRedundant(const uint8_t* encoded,
89 size_t encoded_len) const; 89 size_t encoded_len) const;
90 90
91 // Detects whether a packet has forward error correction. The packet is 91 // Detects whether a packet has forward error correction. The packet is
92 // comprised of the samples in |encoded| which is |encoded_len| bytes long. 92 // comprised of the samples in |encoded| which is |encoded_len| bytes long.
93 // Returns true if the packet has FEC and false otherwise. 93 // Returns true if the packet has FEC and false otherwise.
94 virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; 94 virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
95 95
96 // Returns the actual sample rate of the decoder's output. 96 // Returns the actual sample rate of the decoder's output. This value may not
97 // change during the lifetime of the decoder.
97 // NOTE: For now, this has a default implementation that returns an unusable 98 // NOTE: For now, this has a default implementation that returns an unusable
98 // value (-1). That default implementation will go away soon, and at the same 99 // value (-1). That default implementation will go away soon, and at the same
99 // time callers will start relying on the return value, so make sure you 100 // time callers will start relying on the return value, so make sure you
100 // override it with something that returns a correct value! 101 // override it with something that returns a correct value!
101 // TODO(kwiberg): Remove the default implementation. 102 // TODO(kwiberg): Remove the default implementation.
102 virtual int SampleRateHz() const; 103 virtual int SampleRateHz() const;
103 104
105 // The number of channels in the decoder's output. This value may not change
106 // during the lifetime of the decoder.
104 virtual size_t Channels() const = 0; 107 virtual size_t Channels() const = 0;
105 108
106 protected: 109 protected:
107 static SpeechType ConvertSpeechType(int16_t type); 110 static SpeechType ConvertSpeechType(int16_t type);
108 111
109 virtual int DecodeInternal(const uint8_t* encoded, 112 virtual int DecodeInternal(const uint8_t* encoded,
110 size_t encoded_len, 113 size_t encoded_len,
111 int sample_rate_hz, 114 int sample_rate_hz,
112 int16_t* decoded, 115 int16_t* decoded,
113 SpeechType* speech_type) = 0; 116 SpeechType* speech_type) = 0;
114 117
115 virtual int DecodeRedundantInternal(const uint8_t* encoded, 118 virtual int DecodeRedundantInternal(const uint8_t* encoded,
116 size_t encoded_len, 119 size_t encoded_len,
117 int sample_rate_hz, 120 int sample_rate_hz,
118 int16_t* decoded, 121 int16_t* decoded,
119 SpeechType* speech_type); 122 SpeechType* speech_type);
120 123
121 private: 124 private:
122 RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder); 125 RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
123 }; 126 };
124 127
125 } // namespace webrtc 128 } // namespace webrtc
126 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_ 129 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_
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