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| 1 /* | |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 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 | |
| 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. | |
| 9 */ | |
| 10 | |
| 11 #include <stdio.h> | |
| 12 | |
| 13 #include <map> | |
| 14 | |
| 15 #include "webrtc/video/loopback.h" | |
| 16 | |
| 17 #include "testing/gtest/include/gtest/gtest.h" | |
| 18 | |
| 19 #include "webrtc/base/checks.h" | |
| 20 #include "webrtc/base/scoped_ptr.h" | |
| 21 #include "webrtc/call.h" | |
| 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | |
| 23 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | |
| 24 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | |
| 25 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" | |
| 26 #include "webrtc/system_wrappers/interface/clock.h" | |
| 27 #include "webrtc/test/encoder_settings.h" | |
| 28 #include "webrtc/test/fake_encoder.h" | |
| 29 #include "webrtc/test/layer_filtering_transport.h" | |
| 30 #include "webrtc/test/run_loop.h" | |
| 31 #include "webrtc/test/testsupport/trace_to_stderr.h" | |
| 32 #include "webrtc/test/video_capturer.h" | |
| 33 #include "webrtc/test/video_renderer.h" | |
| 34 #include "webrtc/typedefs.h" | |
| 35 | |
| 36 namespace webrtc { | |
| 37 namespace test { | |
| 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; | |
| 46 static const uint8_t kVideoPayloadTypeVP8 = 124; | |
| 47 static const uint8_t kVideoPayloadTypeVP9 = 125; | |
| 48 | |
| 49 Loopback::Loopback(const Config& config) | |
| 50 : config_(config), clock_(Clock::GetRealTimeClock()) { | |
| 51 } | |
| 52 | |
| 53 Loopback::~Loopback() { | |
| 54 } | |
| 55 | |
| 56 void Loopback::Run() { | |
| 57 rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_; | |
| 58 if (config_.logs) | |
| 59 trace_to_stderr_.reset(new test::TraceToStderr); | |
| 60 | |
| 61 rtc::scoped_ptr<test::VideoRenderer> local_preview( | |
| 62 test::VideoRenderer::Create("Local Preview", config_.width, | |
| 63 config_.height)); | |
| 64 rtc::scoped_ptr<test::VideoRenderer> loopback_video( | |
| 65 test::VideoRenderer::Create("Loopback Video", config_.width, | |
| 66 config_.height)); | |
| 67 | |
| 68 Call::Config call_config; | |
| 69 call_config.bitrate_config.min_bitrate_bps = | |
| 70 static_cast<int>(config_.min_bitrate_kbps) * 1000; | |
| 71 call_config.bitrate_config.start_bitrate_bps = | |
| 72 static_cast<int>(config_.start_bitrate_kbps) * 1000; | |
| 73 call_config.bitrate_config.max_bitrate_bps = | |
| 74 static_cast<int>(config_.max_bitrate_kbps) * 1000; | |
| 75 rtc::scoped_ptr<Call> call(Call::Create(call_config)); | |
| 76 | |
| 77 FakeNetworkPipe::Config pipe_config; | |
| 78 pipe_config.loss_percent = config_.loss_percent; | |
| 79 pipe_config.link_capacity_kbps = config_.link_capacity_kbps; | |
| 80 pipe_config.queue_length_packets = config_.queue_size; | |
| 81 pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms; | |
| 82 pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms; | |
| 83 LayerFilteringTransport send_transport( | |
| 84 pipe_config, kVideoPayloadTypeVP8, kVideoPayloadTypeVP9, | |
| 85 static_cast<uint8_t>(config_.tl_discard_threshold), | |
| 86 static_cast<uint8_t>(config_.sl_discard_threshold)); | |
| 87 | |
| 88 // Loopback, call sends to itself. | |
| 89 send_transport.SetReceiver(call->Receiver()); | |
| 90 | |
| 91 VideoSendStream::Config send_config(&send_transport); | |
| 92 send_config.rtp.ssrcs.push_back(kSendSsrc); | |
| 93 send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc); | |
| 94 send_config.rtp.rtx.payload_type = kRtxVideoPayloadType; | |
| 95 send_config.rtp.nack.rtp_history_ms = 1000; | |
| 96 send_config.rtp.extensions.push_back( | |
| 97 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); | |
| 98 | |
| 99 send_config.local_renderer = local_preview.get(); | |
| 100 rtc::scoped_ptr<VideoEncoder> encoder; | |
| 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 | |
| 118 VideoSendStream* send_stream = | |
| 119 call->CreateVideoSendStream(send_config, encoder_config); | |
| 120 | |
| 121 rtc::scoped_ptr<test::VideoCapturer> capturer(CreateCapturer(send_stream)); | |
| 122 | |
| 123 VideoReceiveStream::Config receive_config(&send_transport); | |
| 124 receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; | |
| 125 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; | |
| 126 receive_config.rtp.nack.rtp_history_ms = 1000; | |
| 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)); | |
| 132 receive_config.renderer = loopback_video.get(); | |
| 133 VideoReceiveStream::Decoder decoder = | |
| 134 test::CreateMatchingDecoder(send_config.encoder_settings); | |
| 135 receive_config.decoders.push_back(decoder); | |
| 136 | |
| 137 VideoReceiveStream* receive_stream = | |
| 138 call->CreateVideoReceiveStream(receive_config); | |
| 139 | |
| 140 receive_stream->Start(); | |
| 141 send_stream->Start(); | |
| 142 capturer->Start(); | |
| 143 | |
| 144 test::PressEnterToContinue(); | |
| 145 | |
| 146 capturer->Stop(); | |
| 147 send_stream->Stop(); | |
| 148 receive_stream->Stop(); | |
| 149 | |
| 150 call->DestroyVideoReceiveStream(receive_stream); | |
| 151 call->DestroyVideoSendStream(send_stream); | |
| 152 | |
| 153 delete decoder.decoder; | |
| 154 | |
| 155 send_transport.StopSending(); | |
| 156 } | |
| 157 | |
| 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 | |
| 183 } // namespace webrtc | |
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Chromium Code Reviews
Issue 1308403003:
Refactoring full stack and loopback tests (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master