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audio_coding: remove "main" directory

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This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}
This commit is contained in:
kjellander
2015-11-26 04:44:54 -08:00
committed by Commit bot
parent a3c20bb9a0
commit 3e6db2321c
85 changed files with 1276 additions and 498 deletions
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57
webrtc/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 "webrtc/base/scoped_ptr.h"
#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, int channels, int bitrate, int frame_length,
int percent_loss = 0);
void OpenOutFile(int test_number);
rtc::scoped_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_;
int 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_