0cb612b43b
https://webrtc-codereview.appspot.com/38279004/ Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43839004 Cr-Commit-Position: refs/heads/master@{#8749} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
42 lines
1.4 KiB
C++
42 lines
1.4 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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const AudioEncoder::EncodedInfo AudioEncoder::kZeroEncodedBytes;
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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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