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Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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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88 /* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */ 88 /* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
89 /* pksize - size of packet in bytes, from NetEq */ 89 /* pksize - size of packet in bytes, from NetEq */
90 /* Index - integer (range 0...23) indicating bottle neck & jitter as e stimated by other side */ 90 /* Index - integer (range 0...23) indicating bottle neck & jitter as e stimated by other side */
91 /* returns 0 if everything went fine, -1 otherwise */ 91 /* returns 0 if everything went fine, -1 otherwise */
92 int16_t WebRtcIsac_UpdateBandwidthEstimator( 92 int16_t WebRtcIsac_UpdateBandwidthEstimator(
93 BwEstimatorstr* bwest_str, 93 BwEstimatorstr* bwest_str,
94 const uint16_t rtp_number, 94 const uint16_t rtp_number,
95 const int32_t frame_length, 95 const int32_t frame_length,
96 const uint32_t send_ts, 96 const uint32_t send_ts,
97 const uint32_t arr_ts, 97 const uint32_t arr_ts,
98 const int32_t pksize); 98 const size_t pksize);
99 99
100 /* Update receiving estimates. Used when we only receive BWE index, no iSAC da ta packet. */ 100 /* Update receiving estimates. Used when we only receive BWE index, no iSAC da ta packet. */
101 int16_t WebRtcIsac_UpdateUplinkBwImpl( 101 int16_t WebRtcIsac_UpdateUplinkBwImpl(
102 BwEstimatorstr* bwest_str, 102 BwEstimatorstr* bwest_str,
103 int16_t Index, 103 int16_t Index,
104 enum IsacSamplingRate encoderSamplingFreq); 104 enum IsacSamplingRate encoderSamplingFreq);
105 105
106 /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */ 106 /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
107 void WebRtcIsac_GetDownlinkBwJitIndexImpl( 107 void WebRtcIsac_GetDownlinkBwJitIndexImpl(
108 BwEstimatorstr* bwest_str, 108 BwEstimatorstr* bwest_str,
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175 int16_t WebRtcIsac_UpdateUplinkJitter( 175 int16_t WebRtcIsac_UpdateUplinkJitter(
176 BwEstimatorstr* bwest_str, 176 BwEstimatorstr* bwest_str,
177 int32_t index); 177 int32_t index);
178 178
179 #if defined(__cplusplus) 179 #if defined(__cplusplus)
180 } 180 }
181 #endif 181 #endif
182 182
183 183
184 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATO R_H_ */ 184 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATO R_H_ */
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