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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellander1afca732016-02-07 20:46:45 -08002 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellander1afca732016-02-07 20:46:45 -08004 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00009 */
10
kjellandera96e2d72016-02-04 23:52:28 -080011#ifndef WEBRTC_MEDIA_BASE_TESTUTILS_H_
12#define WEBRTC_MEDIA_BASE_TESTUTILS_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000013
14#include <string>
15#include <vector>
wu@webrtc.orgcadf9042013-08-30 21:24:16 +000016
henrike@webrtc.org28e20752013-07-10 00:45:36 +000017#include "libyuv/compare.h"
tfarina5237aaf2015-11-10 23:44:30 -080018#include "webrtc/base/arraysize.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000019#include "webrtc/base/basictypes.h"
20#include "webrtc/base/sigslot.h"
21#include "webrtc/base/window.h"
kjellandera96e2d72016-02-04 23:52:28 -080022#include "webrtc/media/base/mediachannel.h"
23#include "webrtc/media/base/videocapturer.h"
24#include "webrtc/media/base/videocommon.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000025
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000026namespace rtc {
jbauchf1f87202016-03-30 06:43:37 -070027class ByteBufferReader;
28class ByteBufferWriter;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000029class StreamInterface;
30}
31
nisseacd935b2016-11-11 03:55:13 -080032namespace webrtc {
33class VideoFrame;
34}
35
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036namespace cricket {
37
38// Returns size of 420 image with rounding on chroma for odd sizes.
39#define I420_SIZE(w, h) (w * h + (((w + 1) / 2) * ((h + 1) / 2)) * 2)
40// Returns size of ARGB image.
41#define ARGB_SIZE(w, h) (w * h * 4)
42
43template <class T> inline std::vector<T> MakeVector(const T a[], size_t s) {
44 return std::vector<T>(a, a + s);
45}
tfarina5237aaf2015-11-10 23:44:30 -080046#define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a))
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047
48struct RtpDumpPacket;
49class RtpDumpWriter;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050
51struct RawRtpPacket {
jbauchf1f87202016-03-30 06:43:37 -070052 void WriteToByteBuffer(uint32_t in_ssrc, rtc::ByteBufferWriter* buf) const;
53 bool ReadFromByteBuffer(rtc::ByteBufferReader* buf);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054 // Check if this packet is the same as the specified packet except the
55 // sequence number and timestamp, which should be the same as the specified
56 // parameters.
Peter Boström0c4e06b2015-10-07 12:23:21 +020057 bool SameExceptSeqNumTimestampSsrc(const RawRtpPacket& packet,
58 uint16_t seq,
59 uint32_t ts,
60 uint32_t ssc) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061 int size() const { return 28; }
62
Peter Boström0c4e06b2015-10-07 12:23:21 +020063 uint8_t ver_to_cc;
64 uint8_t m_to_pt;
65 uint16_t sequence_number;
66 uint32_t timestamp;
67 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068 char payload[16];
69};
70
71struct RawRtcpPacket {
jbauchf1f87202016-03-30 06:43:37 -070072 void WriteToByteBuffer(rtc::ByteBufferWriter* buf) const;
73 bool ReadFromByteBuffer(rtc::ByteBufferReader* buf);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000074 bool EqualsTo(const RawRtcpPacket& packet) const;
75
Peter Boström0c4e06b2015-10-07 12:23:21 +020076 uint8_t ver_to_count;
77 uint8_t type;
78 uint16_t length;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079 char payload[16];
80};
81
82class RtpTestUtility {
83 public:
84 static size_t GetTestPacketCount();
85
86 // Write the first count number of kTestRawRtcpPackets or kTestRawRtpPackets,
87 // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return
88 // true if successful.
Peter Boström0c4e06b2015-10-07 12:23:21 +020089 static bool WriteTestPackets(size_t count,
90 bool rtcp,
91 uint32_t rtp_ssrc,
92 RtpDumpWriter* writer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093
94 // Loop read the first count number of packets from the specified stream.
95 // Verify the elapsed time of the dump packets increase monotonically. If the
96 // stream is a RTP stream, verify the RTP sequence number, timestamp, and
97 // payload. If the stream is a RTCP stream, verify the RTCP header and
98 // payload.
Peter Boström0c4e06b2015-10-07 12:23:21 +020099 static bool VerifyTestPacketsFromStream(size_t count,
100 rtc::StreamInterface* stream,
101 uint32_t ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000102
103 // Verify the dump packet is the same as the raw RTP packet.
104 static bool VerifyPacket(const RtpDumpPacket* dump,
105 const RawRtpPacket* raw,
106 bool header_only);
107
Peter Boström0c4e06b2015-10-07 12:23:21 +0200108 static const uint32_t kDefaultSsrc = 1;
109 static const uint32_t kRtpTimestampIncrease = 90;
110 static const uint32_t kDefaultTimeIncrease = 30;
111 static const uint32_t kElapsedTimeInterval = 10;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112 static const RawRtpPacket kTestRawRtpPackets[];
113 static const RawRtcpPacket kTestRawRtcpPackets[];
114
115 private:
116 RtpTestUtility() {}
117};
118
119// Test helper for testing VideoCapturer implementations.
nisse79246972016-08-23 05:50:09 -0700120class VideoCapturerListener
121 : public sigslot::has_slots<>,
nisseacd935b2016-11-11 03:55:13 -0800122 public rtc::VideoSinkInterface<webrtc::VideoFrame> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123 public:
124 explicit VideoCapturerListener(VideoCapturer* cap);
nisse79246972016-08-23 05:50:09 -0700125 ~VideoCapturerListener();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000126
127 CaptureState last_capture_state() const { return last_capture_state_; }
128 int frame_count() const { return frame_count_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000129 int frame_width() const { return frame_width_; }
130 int frame_height() const { return frame_height_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000131 bool resolution_changed() const { return resolution_changed_; }
132
133 void OnStateChange(VideoCapturer* capturer, CaptureState state);
nisseacd935b2016-11-11 03:55:13 -0800134 void OnFrame(const webrtc::VideoFrame& frame) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000135
136 private:
nisse79246972016-08-23 05:50:09 -0700137 VideoCapturer* capturer_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000138 CaptureState last_capture_state_;
139 int frame_count_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000140 int frame_width_;
141 int frame_height_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142 bool resolution_changed_;
143};
144
145class ScreencastEventCatcher : public sigslot::has_slots<> {
146 public:
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000147 ScreencastEventCatcher() : ssrc_(0), ev_(rtc::WE_RESIZE) { }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200148 uint32_t ssrc() const { return ssrc_; }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000149 rtc::WindowEvent event() const { return ev_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200150 void OnEvent(uint32_t ssrc, rtc::WindowEvent ev) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151 ssrc_ = ssrc;
152 ev_ = ev;
153 }
154 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200155 uint32_t ssrc_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000156 rtc::WindowEvent ev_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000157};
158
159class VideoMediaErrorCatcher : public sigslot::has_slots<> {
160 public:
161 VideoMediaErrorCatcher() : ssrc_(0), error_(VideoMediaChannel::ERROR_NONE) { }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200162 uint32_t ssrc() const { return ssrc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163 VideoMediaChannel::Error error() const { return error_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200164 void OnError(uint32_t ssrc, VideoMediaChannel::Error error) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165 ssrc_ = ssrc;
166 error_ = error;
167 }
168 private:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200169 uint32_t ssrc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000170 VideoMediaChannel::Error error_;
171};
172
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173// Checks whether |codecs| contains |codec|; checks using Codec::Matches().
174template <class C>
175bool ContainsMatchingCodec(const std::vector<C>& codecs, const C& codec) {
176 typename std::vector<C>::const_iterator it;
177 for (it = codecs.begin(); it != codecs.end(); ++it) {
178 if (it->Matches(codec)) {
179 return true;
180 }
181 }
182 return false;
183}
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000184
185// Create Simulcast StreamParams with given |ssrcs| and |cname|.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200186cricket::StreamParams CreateSimStreamParams(const std::string& cname,
187 const std::vector<uint32_t>& ssrcs);
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000188// Create Simulcast stream with given |ssrcs| and |rtx_ssrcs|.
189// The number of |rtx_ssrcs| must match number of |ssrcs|.
190cricket::StreamParams CreateSimWithRtxStreamParams(
Peter Boström0c4e06b2015-10-07 12:23:21 +0200191 const std::string& cname,
192 const std::vector<uint32_t>& ssrcs,
193 const std::vector<uint32_t>& rtx_ssrcs);
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000194
brandtr9688e382016-11-22 00:59:48 -0800195// Create StreamParams with single primary SSRC and corresponding FlexFEC SSRC.
196cricket::StreamParams CreatePrimaryWithFecFrStreamParams(
197 const std::string& cname,
198 uint32_t primary_ssrc,
199 uint32_t flexfec_ssrc);
200
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201} // namespace cricket
202
kjellandera96e2d72016-02-04 23:52:28 -0800203#endif // WEBRTC_MEDIA_BASE_TESTUTILS_H_