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_ |