0521127779
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz() are simple accessors for almost all implementations of AudioEncoder, they are virtual and not guaranteed to be just simple accessors. Thus, it makes more sense to use the normal CamelCase naming scheme. BUG=4235 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34239004 Cr-Commit-Position: refs/heads/master@{#8407} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
41 lines
1.3 KiB
C++
41 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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bool 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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EncodedInfo* info) {
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CHECK_EQ(num_samples_per_channel,
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static_cast<size_t>(SampleRateHz() / 100));
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bool ret =
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EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info);
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CHECK_LE(info->encoded_bytes, max_encoded_bytes);
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return ret;
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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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