e7edea9759
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}
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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