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Side by Side Diff: webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc

Issue 1881003003: Reland Remove the deprecated EncodeInternal interface from AudioEncoder (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Renamed ApproximateEncodedBytes to SufficientOutputBufferSize in Opus Created 4 years, 8 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
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28 AudioEncoderCopyRed::Config::~Config() = default; 28 AudioEncoderCopyRed::Config::~Config() = default;
29 29
30 AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config) 30 AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
31 : speech_encoder_(std::move(config.speech_encoder)), 31 : speech_encoder_(std::move(config.speech_encoder)),
32 red_payload_type_(config.payload_type) { 32 red_payload_type_(config.payload_type) {
33 RTC_CHECK(speech_encoder_) << "Speech encoder not provided."; 33 RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
34 } 34 }
35 35
36 AudioEncoderCopyRed::~AudioEncoderCopyRed() = default; 36 AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
37 37
38 size_t AudioEncoderCopyRed::MaxEncodedBytes() const {
39 return 2 * speech_encoder_->MaxEncodedBytes();
40 }
41
42 int AudioEncoderCopyRed::SampleRateHz() const { 38 int AudioEncoderCopyRed::SampleRateHz() const {
43 return speech_encoder_->SampleRateHz(); 39 return speech_encoder_->SampleRateHz();
44 } 40 }
45 41
46 size_t AudioEncoderCopyRed::NumChannels() const { 42 size_t AudioEncoderCopyRed::NumChannels() const {
47 return speech_encoder_->NumChannels(); 43 return speech_encoder_->NumChannels();
48 } 44 }
49 45
50 int AudioEncoderCopyRed::RtpTimestampRateHz() const { 46 int AudioEncoderCopyRed::RtpTimestampRateHz() const {
51 return speech_encoder_->RtpTimestampRateHz(); 47 return speech_encoder_->RtpTimestampRateHz();
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126 122
127 void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) { 123 void AudioEncoderCopyRed::SetProjectedPacketLossRate(double fraction) {
128 speech_encoder_->SetProjectedPacketLossRate(fraction); 124 speech_encoder_->SetProjectedPacketLossRate(fraction);
129 } 125 }
130 126
131 void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) { 127 void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
132 speech_encoder_->SetTargetBitrate(bits_per_second); 128 speech_encoder_->SetTargetBitrate(bits_per_second);
133 } 129 }
134 130
135 } // namespace webrtc 131 } // namespace webrtc
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