Reasons for revert: 1. glaznev discovered potentially related problems using the Android AppRTCDemo. 2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky. > Support associated payload type when registering Rtx payload type. > > Major changes include, > - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. > - Receiver: Restore RTP packets by the new RTX-APT map. > - Sender: Send RTP packets by checking RTX-APT map. > - Add RTX payload type for RED in the default codec list. > > BUG=4024 > R=pbos@webrtc.org, stefan@webrtc.org > TBR=mflodman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26259004 > > Patch from Changbin Shao <changbin.shao@intel.com>. TBR=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
225 lines
7.8 KiB
C++
225 lines
7.8 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include <map>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/call.h"
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#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
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#include "webrtc/system_wrappers/interface/clock.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/test/direct_transport.h"
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#include "webrtc/test/encoder_settings.h"
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#include "webrtc/test/fake_encoder.h"
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#include "webrtc/test/field_trial.h"
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#include "webrtc/test/run_loop.h"
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#include "webrtc/test/run_test.h"
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#include "webrtc/test/testsupport/trace_to_stderr.h"
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#include "webrtc/test/video_capturer.h"
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#include "webrtc/test/video_renderer.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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static const int kAbsSendTimeExtensionId = 7;
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namespace flags {
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DEFINE_int32(width, 640, "Video width.");
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size_t Width() { return static_cast<size_t>(FLAGS_width); }
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DEFINE_int32(height, 480, "Video height.");
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size_t Height() { return static_cast<size_t>(FLAGS_height); }
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DEFINE_int32(fps, 30, "Frames per second.");
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int Fps() { return static_cast<int>(FLAGS_fps); }
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DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
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size_t MinBitrate() { return static_cast<size_t>(FLAGS_min_bitrate); }
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DEFINE_int32(start_bitrate, 300, "Video starting bitrate.");
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size_t StartBitrate() { return static_cast<size_t>(FLAGS_start_bitrate); }
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DEFINE_int32(max_bitrate, 800, "Maximum video bitrate.");
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size_t MaxBitrate() { return static_cast<size_t>(FLAGS_max_bitrate); }
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DEFINE_string(codec, "VP8", "Video codec to use.");
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std::string Codec() { return static_cast<std::string>(FLAGS_codec); }
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DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
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int LossPercent() {
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return static_cast<int>(FLAGS_loss_percent);
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}
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DEFINE_int32(link_capacity,
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0,
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"Capacity (kbps) of the fake link. 0 means infinite.");
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int LinkCapacity() {
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return static_cast<int>(FLAGS_link_capacity);
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}
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DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets.");
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int QueueSize() {
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return static_cast<int>(FLAGS_queue_size);
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}
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DEFINE_int32(avg_propagation_delay_ms,
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0,
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"Average link propagation delay in ms.");
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int AvgPropagationDelayMs() {
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return static_cast<int>(FLAGS_avg_propagation_delay_ms);
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}
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DEFINE_int32(std_propagation_delay_ms,
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0,
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"Link propagation delay standard deviation in ms.");
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int StdPropagationDelayMs() {
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return static_cast<int>(FLAGS_std_propagation_delay_ms);
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}
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DEFINE_bool(logs, false, "print logs to stderr");
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DEFINE_string(
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force_fieldtrials,
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"",
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"Field trials control experimental feature code which can be forced. "
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"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
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" will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
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"trials are separated by \"/\"");
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} // namespace flags
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static const uint32_t kSendSsrc = 0x654321;
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static const uint32_t kSendRtxSsrc = 0x654322;
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static const uint32_t kReceiverLocalSsrc = 0x123456;
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static const uint8_t kRtxPayloadType = 96;
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void Loopback() {
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scoped_ptr<test::TraceToStderr> trace_to_stderr_;
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if (webrtc::flags::FLAGS_logs)
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trace_to_stderr_.reset(new test::TraceToStderr);
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scoped_ptr<test::VideoRenderer> local_preview(test::VideoRenderer::Create(
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"Local Preview", flags::Width(), flags::Height()));
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scoped_ptr<test::VideoRenderer> loopback_video(test::VideoRenderer::Create(
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"Loopback Video", flags::Width(), flags::Height()));
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FakeNetworkPipe::Config pipe_config;
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pipe_config.loss_percent = flags::LossPercent();
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pipe_config.link_capacity_kbps = flags::LinkCapacity();
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pipe_config.queue_length_packets = flags::QueueSize();
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pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
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pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
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test::DirectTransport transport(pipe_config);
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Call::Config call_config(&transport);
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call_config.stream_bitrates.min_bitrate_bps =
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static_cast<int>(flags::MinBitrate()) * 1000;
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call_config.stream_bitrates.start_bitrate_bps =
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static_cast<int>(flags::StartBitrate()) * 1000;
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call_config.stream_bitrates.max_bitrate_bps =
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static_cast<int>(flags::MaxBitrate()) * 1000;
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scoped_ptr<Call> call(Call::Create(call_config));
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// Loopback, call sends to itself.
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transport.SetReceiver(call->Receiver());
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VideoSendStream::Config send_config;
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send_config.rtp.ssrcs.push_back(kSendSsrc);
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send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);
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send_config.rtp.rtx.payload_type = kRtxPayloadType;
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send_config.rtp.nack.rtp_history_ms = 1000;
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send_config.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
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send_config.local_renderer = local_preview.get();
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scoped_ptr<VideoEncoder> encoder;
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if (flags::Codec() == "VP8") {
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encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
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} else if (flags::Codec() == "VP9") {
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encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
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} else {
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// Codec not supported.
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assert(false && "Codec not supported!");
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return;
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}
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send_config.encoder_settings.encoder = encoder.get();
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send_config.encoder_settings.payload_name = flags::Codec();
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send_config.encoder_settings.payload_type = 124;
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VideoEncoderConfig encoder_config;
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encoder_config.streams = test::CreateVideoStreams(1);
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VideoStream* stream = &encoder_config.streams[0];
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stream->width = flags::Width();
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stream->height = flags::Height();
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stream->min_bitrate_bps = call_config.stream_bitrates.min_bitrate_bps;
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stream->target_bitrate_bps = call_config.stream_bitrates.max_bitrate_bps;
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stream->max_bitrate_bps = call_config.stream_bitrates.max_bitrate_bps;
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stream->max_framerate = 30;
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stream->max_qp = 56;
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VideoSendStream* send_stream =
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call->CreateVideoSendStream(send_config, encoder_config);
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Clock* test_clock = Clock::GetRealTimeClock();
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scoped_ptr<test::VideoCapturer> camera(
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test::VideoCapturer::Create(send_stream->Input(),
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flags::Width(),
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flags::Height(),
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flags::Fps(),
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test_clock));
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VideoReceiveStream::Config receive_config;
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receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
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receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
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receive_config.rtp.nack.rtp_history_ms = 1000;
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receive_config.rtp.rtx[kRtxPayloadType].ssrc = kSendRtxSsrc;
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receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType;
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receive_config.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
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receive_config.renderer = loopback_video.get();
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VideoReceiveStream::Decoder decoder =
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test::CreateMatchingDecoder(send_config.encoder_settings);
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receive_config.decoders.push_back(decoder);
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VideoReceiveStream* receive_stream =
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call->CreateVideoReceiveStream(receive_config);
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receive_stream->Start();
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send_stream->Start();
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camera->Start();
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test::PressEnterToContinue();
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camera->Stop();
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send_stream->Stop();
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receive_stream->Stop();
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call->DestroyVideoReceiveStream(receive_stream);
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call->DestroyVideoSendStream(send_stream);
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delete decoder.decoder;
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transport.StopSending();
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}
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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::testing::InitGoogleTest(&argc, argv);
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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::test::InitFieldTrialsFromString(
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webrtc::flags::FLAGS_force_fieldtrials);
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webrtc::test::RunTest(webrtc::Loopback);
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return 0;
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}
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