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Moving src/webrtc into src/.

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In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
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61
modules/audio_coding/test/opus_test.h Normal file
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/*
* Copyright (c) 2012 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_AUDIO_CODING_TEST_OPUS_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
#include <math.h>
#include <memory>
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/test/ACMTest.h"
#include "webrtc/modules/audio_coding/test/Channel.h"
#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/modules/audio_coding/test/TestStereo.h"
namespace webrtc {
class OpusTest : public ACMTest {
public:
OpusTest();
~OpusTest();
void Perform();
private:
void Run(TestPackStereo* channel,
size_t channels,
int bitrate,
size_t frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
std::unique_ptr<AudioCodingModule> acm_receiver_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
PCMFile out_file_;
PCMFile out_file_standalone_;
int counter_;
uint8_t payload_type_;
uint32_t rtp_timestamp_;
acm2::ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;
WebRtcOpusDecInst* opus_stereo_decoder_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_