92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
69 lines
2.0 KiB
C++
69 lines
2.0 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 VOICE_ENGINE_CODER_H_
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#define VOICE_ENGINE_CODER_H_
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#include <memory>
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#include "common_types.h"
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#include "modules/audio_coding/acm2/codec_manager.h"
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#include "modules/audio_coding/acm2/rent_a_codec.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "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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public:
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explicit AudioCoder(uint32_t instance_id);
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~AudioCoder();
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int32_t SetEncodeCodec(const CodecInst& codec_inst);
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int32_t SetDecodeCodec(const CodecInst& codec_inst);
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int32_t Decode(AudioFrame* decoded_audio,
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uint32_t samp_freq_hz,
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const int8_t* incoming_payload,
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size_t payload_length);
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int32_t PlayoutData(AudioFrame* decoded_audio, uint16_t samp_freq_hz);
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int32_t Encode(const AudioFrame& audio,
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int8_t* encoded_data,
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size_t* encoded_length_in_bytes);
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protected:
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int32_t SendData(FrameType frame_type,
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uint8_t payload_type,
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uint32_t time_stamp,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation) override;
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private:
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std::unique_ptr<AudioCodingModule> acm_;
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acm2::CodecManager codec_manager_;
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acm2::RentACodec rent_a_codec_;
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CodecInst receive_codec_;
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uint32_t encode_timestamp_;
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int8_t* encoded_data_;
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size_t encoded_length_in_bytes_;
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uint32_t decode_timestamp_;
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};
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} // namespace webrtc
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#endif // VOICE_ENGINE_CODER_H_
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