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Side by Side Diff: webrtc/modules/audio_coding/codecs/audio_encoder.cc

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
11 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" 11 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
12 #include "webrtc/base/checks.h" 12 #include "webrtc/base/checks.h"
13 13
14 namespace webrtc { 14 namespace webrtc {
15 15
16 AudioEncoder::EncodedInfo::EncodedInfo() = default; 16 AudioEncoder::EncodedInfo::EncodedInfo() = default;
17 17
18 AudioEncoder::EncodedInfo::~EncodedInfo() = default; 18 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
19 19
20 int AudioEncoder::RtpTimestampRateHz() const { 20 int AudioEncoder::RtpTimestampRateHz() const {
21 return SampleRateHz(); 21 return SampleRateHz();
22 } 22 }
23 23
24 AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp, 24 AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
25 const int16_t* audio, 25 const int16_t* audio,
26 size_t num_samples_per_channel, 26 size_t num_samples_per_channel,
27 size_t max_encoded_bytes, 27 size_t max_encoded_bytes,
28 uint8_t* encoded) { 28 uint8_t* encoded) {
29 CHECK_EQ(num_samples_per_channel, 29 RTC_CHECK_EQ(num_samples_per_channel,
30 static_cast<size_t>(SampleRateHz() / 100)); 30 static_cast<size_t>(SampleRateHz() / 100));
31 EncodedInfo info = 31 EncodedInfo info =
32 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded); 32 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
33 CHECK_LE(info.encoded_bytes, max_encoded_bytes); 33 RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
34 return info; 34 return info;
35 } 35 }
36 36
37 bool AudioEncoder::SetFec(bool enable) { 37 bool AudioEncoder::SetFec(bool enable) {
38 return !enable; 38 return !enable;
39 } 39 }
40 40
41 bool AudioEncoder::SetDtx(bool enable) { 41 bool AudioEncoder::SetDtx(bool enable) {
42 return !enable; 42 return !enable;
43 } 43 }
44 44
45 bool AudioEncoder::SetApplication(Application application) { 45 bool AudioEncoder::SetApplication(Application application) {
46 return false; 46 return false;
47 } 47 }
48 48
49 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} 49 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
50 50
51 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {} 51 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
52 52
53 void AudioEncoder::SetTargetBitrate(int target_bps) {} 53 void AudioEncoder::SetTargetBitrate(int target_bps) {}
54 54
55 } // namespace webrtc 55 } // namespace webrtc
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