9afaee74ab
Old review at: https://webrtc-codereview.appspot.com/43839004/ R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45769004 Cr-Commit-Position: refs/heads/master@{#8788} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
40 lines
1.3 KiB
C++
40 lines
1.3 KiB
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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#include "webrtc/base/checks.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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AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
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const int16_t* audio,
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size_t num_samples_per_channel,
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size_t max_encoded_bytes,
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uint8_t* encoded) {
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CHECK_EQ(num_samples_per_channel,
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static_cast<size_t>(SampleRateHz() / 100));
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EncodedInfo info =
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EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
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CHECK_LE(info.encoded_bytes, max_encoded_bytes);
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return info;
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}
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int AudioEncoder::RtpTimestampRateHz() const {
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return SampleRateHz();
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}
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} // namespace webrtc
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