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Delete unused code.

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* Unused audio_coding and video_coding test code.
* Obsolete voice_engine android test app.
* Left-over placeholder files for remoteaudiotrack and
  portallocatorfactory.

In addition, change modules.gyp dependency from rtc_base to
rtc_base_approved.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2065353002 .

Cr-Commit-Position: refs/heads/master@{#13166}
This commit is contained in:
Niels Möller
2016-06-16 15:51:31 +02:00
parent 2d014be554
commit fc3a8ee47b
22 changed files with 1 additions and 3351 deletions
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1
webrtc/modules/video_coding/test/receiver_tests.h
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@ -19,7 +19,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
#include "webrtc/modules/video_coding/test/test_util.h"
#include "webrtc/modules/video_coding/test/video_source.h"
#include "webrtc/typedefs.h"
class RtpDataCallback : public webrtc::NullRtpData {

85
webrtc/modules/video_coding/test/video_source.h
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@ -1,85 +0,0 @@
/*
* 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_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
#include <string>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/typedefs.h"
enum VideoSize {
kUndefined,
kSQCIF, // 128*96 = 12 288
kQQVGA, // 160*120 = 19 200
kQCIF, // 176*144 = 25 344
kCGA, // 320*200 = 64 000
kQVGA, // 320*240 = 76 800
kSIF, // 352*240 = 84 480
kWQVGA, // 400*240 = 96 000
kCIF, // 352*288 = 101 376
kW288p, // 512*288 = 147 456 (WCIF)
k448p, // 576*448 = 281 088
kVGA, // 640*480 = 307 200
k432p, // 720*432 = 311 040
kW432p, // 768*432 = 331 776
k4SIF, // 704*480 = 337 920
kW448p, // 768*448 = 344 064
kNTSC, // 720*480 = 345 600
kFW448p, // 800*448 = 358 400
kWVGA, // 800*480 = 384 000
k4CIF, // 704*576 = 405 504
kSVGA, // 800*600 = 480 000
kW544p, // 960*544 = 522 240
kW576p, // 1024*576 = 589 824 (W4CIF)
kHD, // 960*720 = 691 200
kXGA, // 1024*768 = 786 432
kWHD, // 1280*720 = 921 600
kFullHD, // 1440*1080 = 1 555 200
kWFullHD, // 1920*1080 = 2 073 600
kNumberOfVideoSizes
};
class VideoSource {
public:
VideoSource();
VideoSource(std::string fileName,
VideoSize size,
float frameRate,
webrtc::VideoType type = webrtc::kI420);
VideoSource(std::string fileName,
uint16_t width,
uint16_t height,
float frameRate = 30,
webrtc::VideoType type = webrtc::kI420);
std::string GetFileName() const { return _fileName; }
uint16_t GetWidth() const { return _width; }
uint16_t GetHeight() const { return _height; }
webrtc::VideoType GetType() const { return _type; }
float GetFrameRate() const { return _frameRate; }
int GetWidthHeight(VideoSize size);
// Returns the filename with the path (including the leading slash) removed.
std::string GetName() const;
size_t GetFrameLength() const;
private:
std::string _fileName;
uint16_t _width;
uint16_t _height;
webrtc::VideoType _type;
float _frameRate;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_