9d7eb13c40
Reason for revert: Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now. Original issue's description: > Move FilePlayer and FileRecorder to Voice Engine > > Because Voice Engine was the only user. > > (This has been landed twice before, as > https://codereview.webrtc.org/2037623002 and > https://codereview.webrtc.org/2240163002. Third time's a charm!) > > NOPRESUBMIT=True > TBR=kjellander@webrtc.org > > Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef > Cr-Commit-Position: refs/heads/master@{#13777} TBR= # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2245413002 Cr-Commit-Position: refs/heads/master@{#13779}
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 WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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#include <memory>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
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#include "webrtc/modules/audio_coding/acm2/rent_a_codec.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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public:
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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 // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
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