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Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )

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Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
This commit is contained in:
kwiberg
2016-06-22 16:29:55 -07:00
committed by Commit bot
parent 079a7a197f
commit e7edea9759
18 changed files with 47 additions and 53 deletions
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68
webrtc/modules/utility/source/coder.h Normal file
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/*
* 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 <memory>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
class AudioCoder : public AudioPacketizationCallback {
public:
AudioCoder(uint32_t instance_id);
~AudioCoder();
int32_t SetEncodeCodec(const CodecInst& codec_inst);
int32_t SetDecodeCodec(const CodecInst& codec_inst);
int32_t Decode(AudioFrame& decoded_audio,
uint32_t samp_freq_hz,
const int8_t* incoming_payload,
size_t payload_length);
int32_t PlayoutData(AudioFrame& decoded_audio, uint16_t& samp_freq_hz);
int32_t Encode(const AudioFrame& audio,
int8_t* encoded_data,
size_t& encoded_length_in_bytes);
protected:
int32_t SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t time_stamp,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
private:
std::unique_ptr<AudioCodingModule> acm_;
acm2::CodecManager codec_manager_;
acm2::RentACodec rent_a_codec_;
CodecInst receive_codec_;
uint32_t encode_timestamp_;
int8_t* encoded_data_;
size_t encoded_length_in_bytes_;
uint32_t decode_timestamp_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_