/* * Copyright (c) 2013 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. */ #include <stdio.h> #include <map> #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/call.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/direct_transport.h" #include "webrtc/test/encoder_settings.h" #include "webrtc/test/fake_encoder.h" #include "webrtc/test/run_loop.h" #include "webrtc/test/run_test.h" #include "webrtc/test/video_capturer.h" #include "webrtc/test/video_renderer.h" #include "webrtc/typedefs.h" namespace webrtc { namespace flags { DEFINE_int32(width, 640, "Video width."); size_t Width() { return static_cast<size_t>(FLAGS_width); } DEFINE_int32(height, 480, "Video height."); size_t Height() { return static_cast<size_t>(FLAGS_height); } DEFINE_int32(fps, 30, "Frames per second."); int Fps() { return static_cast<int>(FLAGS_fps); } DEFINE_int32(min_bitrate, 50, "Minimum video bitrate."); size_t MinBitrate() { return static_cast<size_t>(FLAGS_min_bitrate); } DEFINE_int32(start_bitrate, 300, "Video starting bitrate."); size_t StartBitrate() { return static_cast<size_t>(FLAGS_start_bitrate); } DEFINE_int32(max_bitrate, 800, "Maximum video bitrate."); size_t MaxBitrate() { return static_cast<size_t>(FLAGS_max_bitrate); } DEFINE_string(codec, "VP8", "Video codec to use."); std::string Codec() { return static_cast<std::string>(FLAGS_codec); } DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost."); int LossPercent() { return static_cast<int>(FLAGS_loss_percent); } DEFINE_int32(link_capacity, 0, "Capacity (kbps) of the fake link. 0 means infinite."); int LinkCapacity() { return static_cast<int>(FLAGS_link_capacity); } DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets."); int QueueSize() { return static_cast<int>(FLAGS_queue_size); } DEFINE_int32(avg_propagation_delay_ms, 0, "Average link propagation delay in ms."); int AvgPropagationDelayMs() { return static_cast<int>(FLAGS_avg_propagation_delay_ms); } DEFINE_int32(std_propagation_delay_ms, 0, "Link propagation delay standard deviation in ms."); int StdPropagationDelayMs() { return static_cast<int>(FLAGS_std_propagation_delay_ms); } } // namespace flags static const uint32_t kSendSsrc = 0x654321; static const uint32_t kSendRtxSsrc = 0x654322; static const uint32_t kReceiverLocalSsrc = 0x123456; static const uint8_t kRtxPayloadType = 96; void Loopback() { scoped_ptr<test::VideoRenderer> local_preview(test::VideoRenderer::Create( "Local Preview", flags::Width(), flags::Height())); scoped_ptr<test::VideoRenderer> loopback_video(test::VideoRenderer::Create( "Loopback Video", flags::Width(), flags::Height())); FakeNetworkPipe::Config pipe_config; pipe_config.loss_percent = flags::LossPercent(); pipe_config.link_capacity_kbps = flags::LinkCapacity(); pipe_config.queue_length_packets = flags::QueueSize(); pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs(); pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs(); test::DirectTransport transport(pipe_config); Call::Config call_config(&transport); call_config.start_bitrate_bps = static_cast<int>(flags::StartBitrate()) * 1000; scoped_ptr<Call> call(Call::Create(call_config)); // Loopback, call sends to itself. transport.SetReceiver(call->Receiver()); VideoSendStream::Config send_config; send_config.rtp.ssrcs.push_back(kSendSsrc); send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc); send_config.rtp.rtx.payload_type = kRtxPayloadType; send_config.rtp.nack.rtp_history_ms = 1000; send_config.local_renderer = local_preview.get(); scoped_ptr<VideoEncoder> encoder; if (flags::Codec() == "VP8") { encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8)); } else { // Codec not supported. assert(false && "Codec not supported!"); return; } send_config.encoder_settings.encoder = encoder.get(); send_config.encoder_settings.payload_name = flags::Codec(); send_config.encoder_settings.payload_type = 124; VideoEncoderConfig encoder_config; encoder_config.streams = test::CreateVideoStreams(1); VideoStream* stream = &encoder_config.streams[0]; stream->width = flags::Width(); stream->height = flags::Height(); stream->min_bitrate_bps = static_cast<int>(flags::MinBitrate()) * 1000; stream->target_bitrate_bps = static_cast<int>(flags::MaxBitrate()) * 1000; stream->max_bitrate_bps = static_cast<int>(flags::MaxBitrate()) * 1000; stream->max_framerate = 30; stream->max_qp = 56; VideoSendStream* send_stream = call->CreateVideoSendStream(send_config, encoder_config); Clock* test_clock = Clock::GetRealTimeClock(); scoped_ptr<test::VideoCapturer> camera( test::VideoCapturer::Create(send_stream->Input(), flags::Width(), flags::Height(), flags::Fps(), test_clock)); VideoReceiveStream::Config receive_config; receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; receive_config.rtp.local_ssrc = kReceiverLocalSsrc; receive_config.rtp.nack.rtp_history_ms = 1000; receive_config.rtp.rtx[kRtxPayloadType].ssrc = kSendRtxSsrc; receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType; receive_config.renderer = loopback_video.get(); VideoCodec codec = test::CreateDecoderVideoCodec(send_config.encoder_settings); receive_config.codecs.push_back(codec); VideoReceiveStream* receive_stream = call->CreateVideoReceiveStream(receive_config); receive_stream->Start(); send_stream->Start(); camera->Start(); test::PressEnterToContinue(); camera->Stop(); send_stream->Stop(); receive_stream->Stop(); call->DestroyVideoReceiveStream(receive_stream); call->DestroyVideoSendStream(send_stream); transport.StopSending(); } } // namespace webrtc int main(int argc, char* argv[]) { ::testing::InitGoogleTest(&argc, argv); google::ParseCommandLineFlags(&argc, &argv, true); webrtc::test::RunTest(webrtc::Loopback); return 0; }