
The following three methods are added: rtp_timestamp_rate_hz() SetTargetBitrate() SetProjectedPacketLossRate() Default implementations are provided, and a few overrides are implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new methods to the underlying speech codec. BUG=3926 COAUTHOR:kwiberg@webrtc.org R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34049004 Cr-Commit-Position: refs/heads/master@{#8171} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
26 lines
713 B
C++
26 lines
713 B
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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namespace webrtc {
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AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
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}
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AudioEncoder::EncodedInfo::~EncodedInfo() {
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}
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int AudioEncoder::rtp_timestamp_rate_hz() const {
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return sample_rate_hz();
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}
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} // namespace webrtc
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