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Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h

Issue 1319683002: AudioDecoder: Replace Init() with Reset() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@buffer
Patch Set: review fixes Created 5 years, 3 months ago
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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
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43 size_t len, 43 size_t len,
44 int16_t* decoded, 44 int16_t* decoded,
45 int16_t* speech_type) { 45 int16_t* speech_type) {
46 return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type); 46 return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type);
47 } 47 }
48 static inline size_t DecodePlc(instance_type* inst, 48 static inline size_t DecodePlc(instance_type* inst,
49 int16_t* decoded, 49 int16_t* decoded,
50 size_t num_lost_frames) { 50 size_t num_lost_frames) {
51 return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames); 51 return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames);
52 } 52 }
53 static inline int16_t DecoderInit(instance_type* inst) { 53 static inline void DecoderInit(instance_type* inst) {
54 return WebRtcIsacfix_DecoderInit(inst); 54 WebRtcIsacfix_DecoderInit(inst);
55 } 55 }
56 static inline int Encode(instance_type* inst, 56 static inline int Encode(instance_type* inst,
57 const int16_t* speech_in, 57 const int16_t* speech_in,
58 uint8_t* encoded) { 58 uint8_t* encoded) {
59 return WebRtcIsacfix_Encode(inst, speech_in, encoded); 59 return WebRtcIsacfix_Encode(inst, speech_in, encoded);
60 } 60 }
61 static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) { 61 static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
62 return WebRtcIsacfix_EncoderInit(inst, coding_mode); 62 return WebRtcIsacfix_EncoderInit(inst, coding_mode);
63 } 63 }
64 static inline uint16_t EncSampRate(instance_type* inst) { 64 static inline uint16_t EncSampRate(instance_type* inst) {
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129 : public AudioEncoderMutableImpl<AudioEncoderIsacFix> { 129 : public AudioEncoderMutableImpl<AudioEncoderIsacFix> {
130 public: 130 public:
131 explicit AudioEncoderMutableIsacFix(const CodecInst& codec_inst, 131 explicit AudioEncoderMutableIsacFix(const CodecInst& codec_inst,
132 LockedIsacBandwidthInfo* bwinfo); 132 LockedIsacBandwidthInfo* bwinfo);
133 void SetMaxPayloadSize(int max_payload_size_bytes) override; 133 void SetMaxPayloadSize(int max_payload_size_bytes) override;
134 void SetMaxRate(int max_rate_bps) override; 134 void SetMaxRate(int max_rate_bps) override;
135 }; 135 };
136 136
137 } // namespace webrtc 137 } // namespace webrtc
138 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_I SACFIX_H_ 138 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_I SACFIX_H_
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