019955d770
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium. See audio_encoder.cc and 'sizes' regression here: http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186 > We changed Encode() and EncodeInternal() return type from bool to void in this issue: > 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 TBR=jmarusic@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49449004 Cr-Commit-Position: refs/heads/master@{#8772} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
39 lines
1.2 KiB
C++
39 lines
1.2 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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void 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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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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}
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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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