webrtc_m130/video/end_to_end_tests/multi_stream_tester.cc
Niels Möller 259a497632 Reland "Reland "Move rtp-specific config out of EncoderSettings.""
This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e.

Reason for revert: Intend to investigate and fix perf problems.

Original change's description:
> Revert "Reland "Move rtp-specific config out of EncoderSettings.""
> 
> This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
> 
> Reason for revert: Regression in ramp up perf tests.
> 
> Original change's description:
> > Reland "Move rtp-specific config out of EncoderSettings."
> >
> > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
> >
> > Original change's description:
> > > Move rtp-specific config out of EncoderSettings.
> > >
> > > In VideoSendStream::Config, move payload_name and payload_type from
> > > EncoderSettings to Rtp.
> > >
> > > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > > and should perhaps be renamed in a follow up cl. It's no longer
> > > passed as an argument to VideoCodecInitializer::SetupCodec.
> > >
> > > The latter then needs a different way to know the codec type,
> > > which is provided by a new codec_type member in VideoEncoderConfig.
> > >
> > > Bug: webrtc:8830
> > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22532}
> >
> > Bug: webrtc:8830
> > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> > Reviewed-on: https://webrtc-review.googlesource.com/63721
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22595}
> 
> TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
> 
> Bug: webrtc:8830,chromium:827080
> Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
> Reviewed-on: https://webrtc-review.googlesource.com/65520
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22677}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8830, chromium:827080
Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f
Reviewed-on: https://webrtc-review.googlesource.com/66862
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22751}
2018-04-05 14:30:09 +00:00

153 lines
5.3 KiB
C++

/*
* Copyright 2018 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 "video/end_to_end_tests/multi_stream_tester.h"
#include <memory>
#include <utility>
#include <vector>
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "test/call_test.h"
#include "test/encoder_settings.h"
namespace webrtc {
MultiStreamTester::MultiStreamTester(
test::SingleThreadedTaskQueueForTesting* task_queue)
: task_queue_(task_queue) {
// TODO(sprang): Cleanup when msvc supports explicit initializers for array.
codec_settings[0] = {1, 640, 480};
codec_settings[1] = {2, 320, 240};
codec_settings[2] = {3, 240, 160};
class multi_stream_test {
public:
multi_stream_test();
};
}
MultiStreamTester::~MultiStreamTester() {}
void MultiStreamTester::RunTest() {
webrtc::RtcEventLogNullImpl event_log;
Call::Config config(&event_log);
std::unique_ptr<Call> sender_call;
std::unique_ptr<Call> receiver_call;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
VideoSendStream* send_streams[kNumStreams];
VideoReceiveStream* receive_streams[kNumStreams];
test::FrameGeneratorCapturer* frame_generators[kNumStreams];
std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders;
std::unique_ptr<VideoEncoder> encoders[kNumStreams];
task_queue_->SendTask([&]() {
sender_call = rtc::WrapUnique(Call::Create(config));
receiver_call = rtc::WrapUnique(Call::Create(config));
sender_transport =
rtc::WrapUnique(CreateSendTransport(task_queue_, sender_call.get()));
receiver_transport = rtc::WrapUnique(
CreateReceiveTransport(task_queue_, receiver_call.get()));
sender_transport->SetReceiver(receiver_call->Receiver());
receiver_transport->SetReceiver(sender_call->Receiver());
for (size_t i = 0; i < kNumStreams; ++i)
encoders[i] = VP8Encoder::Create();
for (size_t i = 0; i < kNumStreams; ++i) {
uint32_t ssrc = codec_settings[i].ssrc;
int width = codec_settings[i].width;
int height = codec_settings[i].height;
VideoSendStream::Config send_config(sender_transport.get());
send_config.rtp.ssrcs.push_back(ssrc);
send_config.encoder_settings.encoder = encoders[i].get();
send_config.rtp.payload_name = "VP8";
send_config.rtp.payload_type = kVideoPayloadType;
VideoEncoderConfig encoder_config;
test::FillEncoderConfiguration(kVideoCodecVP8, 1, &encoder_config);
encoder_config.max_bitrate_bps = 100000;
UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
send_streams[i] = sender_call->CreateVideoSendStream(
send_config.Copy(), encoder_config.Copy());
send_streams[i]->Start();
VideoReceiveStream::Config receive_config(receiver_transport.get());
receive_config.rtp.remote_ssrc = ssrc;
receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config);
allocated_decoders.push_back(
std::unique_ptr<VideoDecoder>(decoder.decoder));
receive_config.decoders.push_back(decoder);
UpdateReceiveConfig(i, &receive_config);
receive_streams[i] =
receiver_call->CreateVideoReceiveStream(std::move(receive_config));
receive_streams[i]->Start();
frame_generators[i] = test::FrameGeneratorCapturer::Create(
width, height, rtc::nullopt, rtc::nullopt, 30,
Clock::GetRealTimeClock());
send_streams[i]->SetSource(
frame_generators[i],
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_generators[i]->Start();
}
});
Wait();
task_queue_->SendTask([&]() {
for (size_t i = 0; i < kNumStreams; ++i) {
frame_generators[i]->Stop();
sender_call->DestroyVideoSendStream(send_streams[i]);
receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
delete frame_generators[i];
}
sender_transport.reset();
receiver_transport.reset();
sender_call.reset();
receiver_call.reset();
});
}
void MultiStreamTester::UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) {}
void MultiStreamTester::UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) {}
test::DirectTransport* MultiStreamTester::CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) {
return new test::DirectTransport(task_queue, sender_call, payload_type_map_);
}
test::DirectTransport* MultiStreamTester::CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* receiver_call) {
return new test::DirectTransport(task_queue, receiver_call,
payload_type_map_);
}
} // namespace webrtc