| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ |
| #define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ |
| |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| class AudioFrame; |
| |
| class AudioCoder : public AudioPacketizationCallback |
| { |
| public: |
| AudioCoder(uint32_t instanceID); |
| ~AudioCoder(); |
| |
| int32_t SetEncodeCodec(const CodecInst& codecInst); |
| |
| int32_t SetDecodeCodec(const CodecInst& codecInst); |
| |
| int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz, |
| const int8_t* incomingPayload, size_t payloadLength); |
| |
| int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz); |
| |
| int32_t Encode(const AudioFrame& audio, int8_t* encodedData, |
| size_t& encodedLengthInBytes); |
| |
| protected: |
| int32_t SendData(FrameType frameType, |
| uint8_t payloadType, |
| uint32_t timeStamp, |
| const uint8_t* payloadData, |
| size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| private: |
| rtc::scoped_ptr<AudioCodingModule> _acm; |
| |
| CodecInst _receiveCodec; |
| |
| uint32_t _encodeTimestamp; |
| int8_t* _encodedData; |
| size_t _encodedLengthInBytes; |
| |
| uint32_t _decodeTimestamp; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ |