3e6db2321c
This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
62 lines
1.8 KiB
C++
62 lines
1.8 KiB
C++
/*
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* Copyright (c) 2011 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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#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioFrame;
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class AudioCoder : public AudioPacketizationCallback
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{
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public:
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AudioCoder(uint32_t instanceID);
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~AudioCoder();
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int32_t SetEncodeCodec(const CodecInst& codecInst);
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int32_t SetDecodeCodec(const CodecInst& codecInst);
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int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
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const int8_t* incomingPayload, size_t payloadLength);
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int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
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int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
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size_t& encodedLengthInBytes);
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protected:
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int32_t SendData(FrameType frameType,
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uint8_t payloadType,
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uint32_t timeStamp,
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const uint8_t* payloadData,
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size_t payloadSize,
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const RTPFragmentationHeader* fragmentation) override;
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private:
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rtc::scoped_ptr<AudioCodingModule> _acm;
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CodecInst _receiveCodec;
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uint32_t _encodeTimestamp;
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int8_t* _encodedData;
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size_t _encodedLengthInBytes;
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uint32_t _decodeTimestamp;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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