| Index: webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
|
| similarity index 85%
|
| copy from webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
|
| copy to webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
|
| index 5bca23ec4e72fa8f222a86cb09386438977846b8..69c73d6904489e024c6800641ceaa2d48ac4e780 100644
|
| --- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
|
| +++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
|
| @@ -1,5 +1,5 @@
|
| /*
|
| - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
|
| + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
| *
|
| * Use of this source code is governed by a BSD-style license
|
| * that can be found in the LICENSE file in the root of the source
|
| @@ -8,20 +8,18 @@
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
|
|
| -#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
|
| -#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
|
| +#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
|
| +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
|
|
|
| #include "webrtc/base/checks.h"
|
| -#include "webrtc/base/scoped_ptr.h"
|
| -#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
|
| #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
|
|
|
| namespace webrtc {
|
|
|
| -struct IsacFix {
|
| - typedef ISACFIX_MainStruct instance_type;
|
| +class IsacFix {
|
| + public:
|
| + using instance_type = ISACFIX_MainStruct;
|
| static const bool has_swb = false;
|
| - static const uint16_t kFixSampleRate = 16000;
|
| static inline int16_t Control(instance_type* inst,
|
| int32_t rate,
|
| int framesize) {
|
| @@ -96,9 +94,8 @@ struct IsacFix {
|
| uint16_t sample_rate_hz) {
|
| RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
|
| }
|
| - static inline void SetInitialBweBottleneck(
|
| - instance_type* inst,
|
| - int bottleneck_bits_per_second) {
|
| + static inline void SetInitialBweBottleneck(instance_type* inst,
|
| + int bottleneck_bits_per_second) {
|
| WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
|
| }
|
| static inline int16_t UpdateBwEstimate(instance_type* inst,
|
| @@ -117,10 +114,10 @@ struct IsacFix {
|
| static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
|
| return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate);
|
| }
|
| -};
|
|
|
| -using AudioEncoderIsacFix = AudioEncoderIsacT<IsacFix>;
|
| -using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
|
| + private:
|
| + enum { kFixSampleRate = 16000 };
|
| +};
|
|
|
| } // namespace webrtc
|
| -#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
|
| +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
|
|
|