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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |
13 | 13 |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" | |
17 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 16 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
18 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h" | 17 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h" |
19 | 18 |
20 namespace webrtc { | 19 namespace webrtc { |
21 | 20 |
22 struct CodecInst; | 21 struct CodecInst; |
23 | 22 |
24 template <typename T> | 23 template <typename T> |
25 class AudioEncoderIsacT final : public AudioEncoder { | 24 class AudioEncoderIsacT final : public AudioEncoder { |
26 public: | 25 public: |
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87 | 86 |
88 // Timestamp of the first input of the currently in-progress packet. | 87 // Timestamp of the first input of the currently in-progress packet. |
89 uint32_t packet_timestamp_; | 88 uint32_t packet_timestamp_; |
90 | 89 |
91 // Timestamp of the previously encoded packet. | 90 // Timestamp of the previously encoded packet. |
92 uint32_t last_encoded_timestamp_; | 91 uint32_t last_encoded_timestamp_; |
93 | 92 |
94 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); | 93 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT); |
95 }; | 94 }; |
96 | 95 |
97 template <typename T> | |
98 class AudioDecoderIsacT final : public AudioDecoder { | |
99 public: | |
100 AudioDecoderIsacT(); | |
101 explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo); | |
102 ~AudioDecoderIsacT() override; | |
103 | |
104 bool HasDecodePlc() const override; | |
105 size_t DecodePlc(size_t num_frames, int16_t* decoded) override; | |
106 void Reset() override; | |
107 int IncomingPacket(const uint8_t* payload, | |
108 size_t payload_len, | |
109 uint16_t rtp_sequence_number, | |
110 uint32_t rtp_timestamp, | |
111 uint32_t arrival_timestamp) override; | |
112 int ErrorCode() override; | |
113 size_t Channels() const override; | |
114 int DecodeInternal(const uint8_t* encoded, | |
115 size_t encoded_len, | |
116 int sample_rate_hz, | |
117 int16_t* decoded, | |
118 SpeechType* speech_type) override; | |
119 | |
120 private: | |
121 typename T::instance_type* isac_state_; | |
122 LockedIsacBandwidthInfo* bwinfo_; | |
123 int decoder_sample_rate_hz_; | |
124 | |
125 RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT); | |
126 }; | |
127 | |
128 } // namespace webrtc | 96 } // namespace webrtc |
129 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ | 97 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ |
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