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Unified Diff: webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h

Issue 1339253003: Move AudioDecoderIsac* to its own files (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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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_

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