webrtc_m130/webrtc/video/loopback.cc
Stefan Holmer e590416722 Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
2015-03-26 10:11:22 +00:00

169 lines
5.8 KiB
C++

/*
* 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 "webrtc/video/loopback.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/system_wrappers/interface/clock.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/testsupport/trace_to_stderr.h"
#include "webrtc/test/video_capturer.h"
#include "webrtc/test/video_renderer.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace test {
static const int kAbsSendTimeExtensionId = 7;
static const uint32_t kSendSsrc = 0x654321;
static const uint32_t kSendRtxSsrc = 0x654322;
static const uint32_t kReceiverLocalSsrc = 0x123456;
static const uint8_t kRtxPayloadType = 96;
Loopback::Loopback(const Config& config)
: config_(config), clock_(Clock::GetRealTimeClock()) {
}
Loopback::~Loopback() {
}
void Loopback::Run() {
rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
if (config_.logs)
trace_to_stderr_.reset(new test::TraceToStderr);
rtc::scoped_ptr<test::VideoRenderer> local_preview(
test::VideoRenderer::Create("Local Preview", config_.width,
config_.height));
rtc::scoped_ptr<test::VideoRenderer> loopback_video(
test::VideoRenderer::Create("Loopback Video", config_.width,
config_.height));
FakeNetworkPipe::Config pipe_config;
pipe_config.loss_percent = config_.loss_percent;
pipe_config.link_capacity_kbps = config_.link_capacity_kbps;
pipe_config.queue_length_packets = config_.queue_size;
pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms;
pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms;
test::DirectTransport transport(pipe_config);
Call::Config call_config(&transport);
call_config.bitrate_config.min_bitrate_bps =
static_cast<int>(config_.min_bitrate_kbps) * 1000;
call_config.bitrate_config.start_bitrate_bps =
static_cast<int>(config_.start_bitrate_kbps) * 1000;
call_config.bitrate_config.max_bitrate_bps =
static_cast<int>(config_.max_bitrate_kbps) * 1000;
rtc::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.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
send_config.local_renderer = local_preview.get();
rtc::scoped_ptr<VideoEncoder> encoder;
if (config_.codec == "VP8") {
encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
} else if (config_.codec == "VP9") {
encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
} else {
// Codec not supported.
RTC_NOTREACHED() << "Codec not supported!";
return;
}
send_config.encoder_settings.encoder = encoder.get();
send_config.encoder_settings.payload_name = config_.codec;
send_config.encoder_settings.payload_type = 124;
VideoEncoderConfig encoder_config(CreateEncoderConfig());
VideoSendStream* send_stream =
call->CreateVideoSendStream(send_config, encoder_config);
rtc::scoped_ptr<test::VideoCapturer> capturer(CreateCapturer(send_stream));
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.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
receive_config.renderer = loopback_video.get();
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config.encoder_settings);
receive_config.decoders.push_back(decoder);
VideoReceiveStream* receive_stream =
call->CreateVideoReceiveStream(receive_config);
receive_stream->Start();
send_stream->Start();
capturer->Start();
test::PressEnterToContinue();
capturer->Stop();
send_stream->Stop();
receive_stream->Stop();
call->DestroyVideoReceiveStream(receive_stream);
call->DestroyVideoSendStream(send_stream);
delete decoder.decoder;
transport.StopSending();
}
VideoEncoderConfig Loopback::CreateEncoderConfig() {
VideoEncoderConfig encoder_config;
encoder_config.streams = test::CreateVideoStreams(1);
VideoStream* stream = &encoder_config.streams[0];
stream->width = config_.width;
stream->height = config_.height;
stream->min_bitrate_bps = static_cast<int>(config_.min_bitrate_kbps) * 1000;
stream->max_bitrate_bps = static_cast<int>(config_.max_bitrate_kbps) * 1000;
stream->target_bitrate_bps =
static_cast<int>(config_.max_bitrate_kbps) * 1000;
stream->max_framerate = config_.fps;
stream->max_qp = 56;
return encoder_config;
}
test::VideoCapturer* Loopback::CreateCapturer(VideoSendStream* send_stream) {
return test::VideoCapturer::Create(send_stream->Input(), config_.width,
config_.height, config_.fps, clock_);
}
} // namespace test
} // namespace webrtc