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Side by Side Diff: webrtc/video/loopback.cc

Issue 1308403003: Refactoring full stack and loopback tests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Disabling a dependency for android only. Rebase master. Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include <stdio.h> 11 #include <stdio.h>
12 12
13 #include <map> 13 #include <map>
14 14
15 #include "webrtc/video/loopback.h" 15 #include "webrtc/video/loopback.h"
16 16
17 #include "testing/gtest/include/gtest/gtest.h" 17 #include "testing/gtest/include/gtest/gtest.h"
18 18
19 #include "webrtc/base/checks.h" 19 #include "webrtc/base/checks.h"
20 #include "webrtc/base/scoped_ptr.h" 20 #include "webrtc/base/scoped_ptr.h"
21 #include "webrtc/call.h" 21 #include "webrtc/call.h"
22 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" 25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
26 #include "webrtc/system_wrappers/interface/clock.h" 26 #include "webrtc/system_wrappers/interface/clock.h"
27 #include "webrtc/test/encoder_settings.h" 27 #include "webrtc/test/encoder_settings.h"
28 #include "webrtc/test/fake_encoder.h" 28 #include "webrtc/test/fake_encoder.h"
29 #include "webrtc/test/layer_filtering_transport.h" 29 #include "webrtc/test/layer_filtering_transport.h"
30 #include "webrtc/test/frame_generator.h"
30 #include "webrtc/test/run_loop.h" 31 #include "webrtc/test/run_loop.h"
31 #include "webrtc/test/testsupport/trace_to_stderr.h"
32 #include "webrtc/test/video_capturer.h" 32 #include "webrtc/test/video_capturer.h"
33 #include "webrtc/test/video_renderer.h" 33 #include "webrtc/test/video_renderer.h"
34 #include "webrtc/typedefs.h" 34 #include "webrtc/typedefs.h"
35 35
36 namespace webrtc { 36 namespace webrtc {
37 namespace test { 37 namespace test {
38 38
39 static const int kAbsSendTimeExtensionId = 7;
40
41 static const uint32_t kSendSsrc = 0x654321;
42 static const uint32_t kSendRtxSsrc = 0x654322;
43 static const uint32_t kReceiverLocalSsrc = 0x123456;
44
45 static const uint8_t kRtxVideoPayloadType = 96; 39 static const uint8_t kRtxVideoPayloadType = 96;
46 static const uint8_t kVideoPayloadTypeVP8 = 124; 40 static const uint8_t kVideoPayloadTypeVP8 = 124;
47 static const uint8_t kVideoPayloadTypeVP9 = 125; 41 static const uint8_t kVideoPayloadTypeVP9 = 125;
48 42
49 Loopback::Loopback(const Config& config) 43 Loopback::Loopback(const Config& config)
50 : config_(config), clock_(Clock::GetRealTimeClock()) { 44 : config_(config), clock_(Clock::GetRealTimeClock()) {
51 } 45 }
52 46
53 Loopback::~Loopback() { 47 void Loopback::TestBody() {
48 // Empty. Use Run() instead.
sprang_webrtc 2015/09/11 08:01:34 Should this ever be reached?
ivica 2015/09/14 17:12:55 Whoops, rebase error. This file should've been rem
54 } 49 }
55 50
56 void Loopback::Run() { 51 void Loopback::Run() {
57 rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_; 52 rtc::scoped_ptr<test::VideoRenderer> local_preview(
58 if (config_.logs) 53 test::VideoRenderer::Create("Local Preview", config_.shared.width,
59 trace_to_stderr_.reset(new test::TraceToStderr); 54 config_.shared.height));
55 rtc::scoped_ptr<test::VideoRenderer> loopback_video(
56 test::VideoRenderer::Create("Loopback Video", config_.shared.width,
57 config_.shared.height));
60 58
61 rtc::scoped_ptr<test::VideoRenderer> local_preview( 59 test::DirectTransport transport(config_.shared.pipe);
62 test::VideoRenderer::Create("Local Preview", config_.width, 60 Call::Config call_config(&transport);
63 config_.height));
64 rtc::scoped_ptr<test::VideoRenderer> loopback_video(
65 test::VideoRenderer::Create("Loopback Video", config_.width,
66 config_.height));
67 61
68 Call::Config call_config;
69 call_config.bitrate_config.min_bitrate_bps = 62 call_config.bitrate_config.min_bitrate_bps =
70 static_cast<int>(config_.min_bitrate_kbps) * 1000; 63 static_cast<int>(config_.shared.min_bitrate_bps);
71 call_config.bitrate_config.start_bitrate_bps = 64 call_config.bitrate_config.start_bitrate_bps =
72 static_cast<int>(config_.start_bitrate_kbps) * 1000; 65 static_cast<int>(config_.start_bitrate_bps);
73 call_config.bitrate_config.max_bitrate_bps = 66 call_config.bitrate_config.max_bitrate_bps =
74 static_cast<int>(config_.max_bitrate_kbps) * 1000; 67 static_cast<int>(config_.shared.max_bitrate_bps);
75 rtc::scoped_ptr<Call> call(Call::Create(call_config)); 68 rtc::scoped_ptr<Call> call(Call::Create(call_config));
76 69
77 FakeNetworkPipe::Config pipe_config; 70 FakeNetworkPipe::Config pipe_config;
78 pipe_config.loss_percent = config_.loss_percent; 71 pipe_config.loss_percent = config_.loss_percent;
79 pipe_config.link_capacity_kbps = config_.link_capacity_kbps; 72 pipe_config.link_capacity_kbps = config_.link_capacity_kbps;
80 pipe_config.queue_length_packets = config_.queue_size; 73 pipe_config.queue_length_packets = config_.queue_size;
81 pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms; 74 pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms;
82 pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms; 75 pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms;
83 LayerFilteringTransport send_transport( 76 LayerFilteringTransport send_transport(
84 pipe_config, kVideoPayloadTypeVP8, kVideoPayloadTypeVP9, 77 pipe_config, kVideoPayloadTypeVP8, kVideoPayloadTypeVP9,
85 static_cast<uint8_t>(config_.tl_discard_threshold), 78 static_cast<uint8_t>(config_.tl_discard_threshold),
86 static_cast<uint8_t>(config_.sl_discard_threshold)); 79 static_cast<uint8_t>(config_.sl_discard_threshold));
87 80
88 // Loopback, call sends to itself. 81 // Loopback, call sends to itself.
89 send_transport.SetReceiver(call->Receiver()); 82 send_transport.SetReceiver(call->Receiver());
90 83
91 VideoSendStream::Config send_config(&send_transport); 84 rtc::scoped_ptr<VideoEncoder> encoder;
92 send_config.rtp.ssrcs.push_back(kSendSsrc); 85 SendConfigDefaultValues(&config_.shared);
93 send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc); 86 SetupFromSharedConfig(config_.shared, &encoder);
94 send_config.rtp.rtx.payload_type = kRtxVideoPayloadType; 87 send_config_.local_renderer = local_preview.get();
95 send_config.rtp.nack.rtp_history_ms = 1000; 88 send_config_.encoder_settings.payload_type = kVideoPayloadType;
96 send_config.rtp.extensions.push_back( 89 // VideoSendStream::Config send_config;
97 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); 90 // send_config.rtp.ssrcs.push_back(kSendSsrc); // (!)
91 // send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc); // (!)
92 send_config_.rtp.rtx.payload_type = kRtxVideoPayloadType; // (!)
sprang_webrtc 2015/09/11 08:01:34 What's with the comments here?
ivica 2015/09/14 17:12:55 Same as above, file deleted.
98 93
99 send_config.local_renderer = local_preview.get(); 94 rtc::scoped_ptr<test::FrameGenerator> frame_generator;
100 rtc::scoped_ptr<VideoEncoder> encoder; 95 SetupScreenshare(config_.shared, &codec_settings_, &frame_generator);
sprang_webrtc 2015/09/11 08:01:34 Is this only used with screenshare?
ivica 2015/09/14 17:12:55 Same as above, file deleted.
101 if (config_.codec == "VP8") {
102 encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
103 } else if (config_.codec == "VP9") {
104 encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
105 } else {
106 // Codec not supported.
107 RTC_NOTREACHED() << "Codec not supported!";
108 return;
109 }
110 const int payload_type =
111 config_.codec == "VP8" ? kVideoPayloadTypeVP8 : kVideoPayloadTypeVP9;
112 send_config.encoder_settings.encoder = encoder.get();
113 send_config.encoder_settings.payload_name = config_.codec;
114 send_config.encoder_settings.payload_type = payload_type;
115
116 VideoEncoderConfig encoder_config(CreateEncoderConfig());
117 96
118 VideoSendStream* send_stream = 97 VideoSendStream* send_stream =
119 call->CreateVideoSendStream(send_config, encoder_config); 98 call->CreateVideoSendStream(send_config_, encoder_config_);
120 99
121 rtc::scoped_ptr<test::VideoCapturer> capturer(CreateCapturer(send_stream)); 100 test::FrameGeneratorCapturer* capturer = new test::FrameGeneratorCapturer(
101 clock_, send_stream->Input(), frame_generator.release(),
102 config_.shared.fps);
103 EXPECT_TRUE(capturer->Init());
122 104
123 VideoReceiveStream::Config receive_config(&send_transport); 105 receive_configs_[0].rtp.rtx[kVideoPayloadType].ssrc = kSendRtxSsrcs[0];
124 receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; 106 receive_configs_[0].rtp.rtx[kVideoPayloadType].payload_type =
125 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; 107 kRtxVideoPayloadType;
126 receive_config.rtp.nack.rtp_history_ms = 1000; 108 receive_configs_[0].rtp.extensions.push_back(
127 receive_config.rtp.remb = true;
128 receive_config.rtp.rtx[payload_type].ssrc = kSendRtxSsrc;
129 receive_config.rtp.rtx[payload_type].payload_type = kRtxVideoPayloadType;
130 receive_config.rtp.extensions.push_back(
131 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); 109 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
132 receive_config.renderer = loopback_video.get(); 110 receive_configs_[0].renderer = loopback_video.get();
133 VideoReceiveStream::Decoder decoder =
134 test::CreateMatchingDecoder(send_config.encoder_settings);
135 receive_config.decoders.push_back(decoder);
136 111
137 VideoReceiveStream* receive_stream = 112 VideoReceiveStream* receive_stream =
138 call->CreateVideoReceiveStream(receive_config); 113 call->CreateVideoReceiveStream(receive_configs_[0]);
139 114
140 receive_stream->Start(); 115 receive_stream->Start();
141 send_stream->Start(); 116 send_stream->Start();
142 capturer->Start(); 117 capturer->Start();
143 118
144 test::PressEnterToContinue(); 119 test::PressEnterToContinue();
145 120
146 capturer->Stop(); 121 capturer->Stop();
147 send_stream->Stop(); 122 send_stream->Stop();
148 receive_stream->Stop(); 123 receive_stream->Stop();
149 124
150 call->DestroyVideoReceiveStream(receive_stream); 125 call->DestroyVideoReceiveStream(receive_stream);
151 call->DestroyVideoSendStream(send_stream); 126 call->DestroyVideoSendStream(send_stream);
152 127
153 delete decoder.decoder; 128 // delete decoder.decoder;
sprang_webrtc 2015/09/11 08:01:34 Remove
ivica 2015/09/14 17:12:55 Same as above, file deleted.
154 129
155 send_transport.StopSending(); 130 send_transport.StopSending();
156 } 131 }
157 132
158 VideoEncoderConfig Loopback::CreateEncoderConfig() {
159 VideoEncoderConfig encoder_config;
160 encoder_config.streams = test::CreateVideoStreams(1);
161 VideoStream* stream = &encoder_config.streams[0];
162 stream->width = config_.width;
163 stream->height = config_.height;
164 stream->min_bitrate_bps = static_cast<int>(config_.min_bitrate_kbps) * 1000;
165 stream->max_bitrate_bps = static_cast<int>(config_.max_bitrate_kbps) * 1000;
166 stream->target_bitrate_bps =
167 static_cast<int>(config_.max_bitrate_kbps) * 1000;
168 stream->max_framerate = config_.fps;
169 stream->max_qp = 56;
170 if (config_.num_temporal_layers != 0) {
171 stream->temporal_layer_thresholds_bps.resize(config_.num_temporal_layers -
172 1);
173 }
174 return encoder_config;
175 }
176
177 test::VideoCapturer* Loopback::CreateCapturer(VideoSendStream* send_stream) {
178 return test::VideoCapturer::Create(send_stream->Input(), config_.width,
179 config_.height, config_.fps, clock_);
180 }
181
182 } // namespace test 133 } // namespace test
183 } // namespace webrtc 134 } // namespace webrtc
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