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Side by Side Diff: webrtc/modules/audio_coding/neteq/delay_manager.cc

Issue 2243403007: NetEq: Changing checked_cast to saturated_cast (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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91 if (!IsNewerTimestamp(timestamp, last_timestamp_) || 91 if (!IsNewerTimestamp(timestamp, last_timestamp_) ||
92 !IsNewerSequenceNumber(sequence_number, last_seq_no_)) { 92 !IsNewerSequenceNumber(sequence_number, last_seq_no_)) {
93 // Wrong timestamp or sequence order; use stored value. 93 // Wrong timestamp or sequence order; use stored value.
94 packet_len_ms = packet_len_ms_; 94 packet_len_ms = packet_len_ms_;
95 } else { 95 } else {
96 // Calculate timestamps per packet and derive packet length in ms. 96 // Calculate timestamps per packet and derive packet length in ms.
97 int64_t packet_len_samp = 97 int64_t packet_len_samp =
98 static_cast<uint32_t>(timestamp - last_timestamp_) / 98 static_cast<uint32_t>(timestamp - last_timestamp_) /
99 static_cast<uint16_t>(sequence_number - last_seq_no_); 99 static_cast<uint16_t>(sequence_number - last_seq_no_);
100 packet_len_ms = 100 packet_len_ms =
101 rtc::checked_cast<int>(1000 * packet_len_samp / sample_rate_hz); 101 rtc::saturated_cast<int>(1000 * packet_len_samp / sample_rate_hz);
102 } 102 }
103 103
104 if (packet_len_ms > 0) { 104 if (packet_len_ms > 0) {
105 // Cannot update statistics unless |packet_len_ms| is valid. 105 // Cannot update statistics unless |packet_len_ms| is valid.
106 // Calculate inter-arrival time (IAT) in integer "packet times" 106 // Calculate inter-arrival time (IAT) in integer "packet times"
107 // (rounding down). This is the value used as index to the histogram 107 // (rounding down). This is the value used as index to the histogram
108 // vector |iat_vector_|. 108 // vector |iat_vector_|.
109 int iat_packets = packet_iat_stopwatch_->ElapsedMs() / packet_len_ms; 109 int iat_packets = packet_iat_stopwatch_->ElapsedMs() / packet_len_ms;
110 110
111 if (streaming_mode_) { 111 if (streaming_mode_) {
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417 int DelayManager::base_target_level() const { return base_target_level_; } 417 int DelayManager::base_target_level() const { return base_target_level_; }
418 void DelayManager::set_streaming_mode(bool value) { streaming_mode_ = value; } 418 void DelayManager::set_streaming_mode(bool value) { streaming_mode_ = value; }
419 int DelayManager::last_pack_cng_or_dtmf() const { 419 int DelayManager::last_pack_cng_or_dtmf() const {
420 return last_pack_cng_or_dtmf_; 420 return last_pack_cng_or_dtmf_;
421 } 421 }
422 422
423 void DelayManager::set_last_pack_cng_or_dtmf(int value) { 423 void DelayManager::set_last_pack_cng_or_dtmf(int value) {
424 last_pack_cng_or_dtmf_ = value; 424 last_pack_cng_or_dtmf_ = value;
425 } 425 }
426 } // namespace webrtc 426 } // namespace webrtc
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