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}
153 lines
5.3 KiB
C++
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
|