Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(218)

Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after
43 int16_t* decoded, 43 int16_t* decoded,
44 int16_t* speech_type) { 44 int16_t* speech_type) {
45 return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type); 45 return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
46 } 46 }
47 static inline size_t DecodePlc(instance_type* inst, 47 static inline size_t DecodePlc(instance_type* inst,
48 int16_t* decoded, 48 int16_t* decoded,
49 size_t num_lost_frames) { 49 size_t num_lost_frames) {
50 return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames); 50 return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
51 } 51 }
52 52
53 static inline int16_t DecoderInit(instance_type* inst) { 53 static inline void DecoderInit(instance_type* inst) {
54 return WebRtcIsac_DecoderInit(inst); 54 WebRtcIsac_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 WebRtcIsac_Encode(inst, speech_in, encoded); 59 return WebRtcIsac_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 WebRtcIsac_EncoderInit(inst, coding_mode); 62 return WebRtcIsac_EncoderInit(inst, coding_mode);
63 } 63 }
64 static inline uint16_t EncSampRate(instance_type* inst) { 64 static inline uint16_t EncSampRate(instance_type* inst) {
(...skipping 62 matching lines...) Expand 10 before | Expand all | Expand 10 after
127 : public AudioEncoderMutableImpl<AudioEncoderIsac> { 127 : public AudioEncoderMutableImpl<AudioEncoderIsac> {
128 public: 128 public:
129 AudioEncoderMutableIsacFloat(const CodecInst& codec_inst, 129 AudioEncoderMutableIsacFloat(const CodecInst& codec_inst,
130 LockedIsacBandwidthInfo* bwinfo); 130 LockedIsacBandwidthInfo* bwinfo);
131 void SetMaxPayloadSize(int max_payload_size_bytes) override; 131 void SetMaxPayloadSize(int max_payload_size_bytes) override;
132 void SetMaxRate(int max_rate_bps) override; 132 void SetMaxRate(int max_rate_bps) override;
133 }; 133 };
134 134
135 } // namespace webrtc 135 } // namespace webrtc
136 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ ISAC_H_ 136 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ ISAC_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698