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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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