2018-02-02 16:24:16 +01:00
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/*
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* Copyright 2018 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 "video/end_to_end_tests/multi_stream_tester.h"
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#include <memory>
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#include <utility>
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#include <vector>
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2019-01-07 10:21:47 -08:00
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#include "absl/memory/memory.h"
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2019-08-07 12:24:53 +02:00
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#include "api/rtc_event_log/rtc_event_log.h"
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2019-04-18 14:34:16 +02:00
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#include "api/task_queue/default_task_queue_factory.h"
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2019-09-30 04:16:28 +02:00
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#include "api/task_queue/task_queue_base.h"
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2018-08-20 13:27:45 +02:00
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#include "api/test/simulated_network.h"
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2018-10-26 15:57:48 +02:00
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#include "api/test/video/function_video_encoder_factory.h"
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2018-11-08 10:02:56 -08:00
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#include "api/video/builtin_video_bitrate_allocator_factory.h"
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2018-08-20 13:27:45 +02:00
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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2019-01-11 09:11:00 -08:00
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#include "media/engine/internal_decoder_factory.h"
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2018-02-02 16:24:16 +01:00
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#include "modules/video_coding/codecs/vp8/include/vp8.h"
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2019-10-21 09:24:27 +02:00
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#include "rtc_base/task_queue_for_test.h"
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2018-02-02 16:24:16 +01:00
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#include "test/call_test.h"
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#include "test/encoder_settings.h"
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2019-11-08 16:17:41 +01:00
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#include "test/single_threaded_task_queue.h"
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2018-02-02 16:24:16 +01:00
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namespace webrtc {
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2019-11-08 16:17:41 +01:00
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MultiStreamTester::MultiStreamTester() {
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2018-02-02 16:24:16 +01:00
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// TODO(sprang): Cleanup when msvc supports explicit initializers for array.
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codec_settings[0] = {1, 640, 480};
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codec_settings[1] = {2, 320, 240};
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codec_settings[2] = {3, 240, 160};
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}
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2019-11-08 16:17:41 +01:00
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MultiStreamTester::~MultiStreamTester() = default;
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2018-02-02 16:24:16 +01:00
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void MultiStreamTester::RunTest() {
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2019-08-07 12:24:53 +02:00
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webrtc::RtcEventLogNull event_log;
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2019-04-18 14:34:16 +02:00
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auto task_queue_factory = CreateDefaultTaskQueueFactory();
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2019-11-08 16:17:41 +01:00
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// TODO(bugs.webrtc.org/10933): Use production task queue implementation.
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auto task_queue =
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std::make_unique<test::DEPRECATED_SingleThreadedTaskQueueForTesting>(
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"TaskQueue");
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2018-02-02 16:24:16 +01:00
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Call::Config config(&event_log);
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2019-07-03 14:56:33 +02:00
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config.task_queue_factory = task_queue_factory.get();
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2018-02-02 16:24:16 +01:00
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std::unique_ptr<Call> sender_call;
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std::unique_ptr<Call> receiver_call;
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std::unique_ptr<test::DirectTransport> sender_transport;
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std::unique_ptr<test::DirectTransport> receiver_transport;
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VideoSendStream* send_streams[kNumStreams];
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VideoReceiveStream* receive_streams[kNumStreams];
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test::FrameGeneratorCapturer* frame_generators[kNumStreams];
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2018-04-19 09:04:13 +02:00
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test::FunctionVideoEncoderFactory encoder_factory(
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[]() { return VP8Encoder::Create(); });
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2018-11-08 10:02:56 -08:00
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std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory =
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CreateBuiltinVideoBitrateAllocatorFactory();
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2018-09-28 09:07:24 +02:00
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InternalDecoderFactory decoder_factory;
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2018-02-02 16:24:16 +01:00
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2019-11-08 16:17:41 +01:00
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SendTask(RTC_FROM_HERE, task_queue.get(), [&]() {
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Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
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sender_call = absl::WrapUnique(Call::Create(config));
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receiver_call = absl::WrapUnique(Call::Create(config));
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2019-11-08 16:17:41 +01:00
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sender_transport = CreateSendTransport(task_queue.get(), sender_call.get());
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2019-09-30 04:16:28 +02:00
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receiver_transport =
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2019-11-08 16:17:41 +01:00
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CreateReceiveTransport(task_queue.get(), receiver_call.get());
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2018-02-02 16:24:16 +01:00
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sender_transport->SetReceiver(receiver_call->Receiver());
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receiver_transport->SetReceiver(sender_call->Receiver());
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for (size_t i = 0; i < kNumStreams; ++i) {
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uint32_t ssrc = codec_settings[i].ssrc;
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int width = codec_settings[i].width;
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int height = codec_settings[i].height;
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VideoSendStream::Config send_config(sender_transport.get());
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send_config.rtp.ssrcs.push_back(ssrc);
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2018-04-19 09:04:13 +02:00
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send_config.encoder_settings.encoder_factory = &encoder_factory;
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2018-11-08 10:02:56 -08:00
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send_config.encoder_settings.bitrate_allocator_factory =
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bitrate_allocator_factory.get();
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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 15:36:51 +02:00
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send_config.rtp.payload_name = "VP8";
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send_config.rtp.payload_type = kVideoPayloadType;
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2018-02-02 16:24:16 +01:00
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VideoEncoderConfig encoder_config;
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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 15:36:51 +02:00
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test::FillEncoderConfiguration(kVideoCodecVP8, 1, &encoder_config);
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2018-02-02 16:24:16 +01:00
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encoder_config.max_bitrate_bps = 100000;
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UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
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send_streams[i] = sender_call->CreateVideoSendStream(
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send_config.Copy(), encoder_config.Copy());
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send_streams[i]->Start();
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VideoReceiveStream::Config receive_config(receiver_transport.get());
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receive_config.rtp.remote_ssrc = ssrc;
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receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
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VideoReceiveStream::Decoder decoder =
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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 15:36:51 +02:00
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test::CreateMatchingDecoder(send_config);
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2018-09-28 09:07:24 +02:00
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decoder.decoder_factory = &decoder_factory;
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2018-02-02 16:24:16 +01:00
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receive_config.decoders.push_back(decoder);
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UpdateReceiveConfig(i, &receive_config);
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receive_streams[i] =
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receiver_call->CreateVideoReceiveStream(std::move(receive_config));
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receive_streams[i]->Start();
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2019-04-18 14:34:16 +02:00
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auto* frame_generator = new test::FrameGeneratorCapturer(
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Clock::GetRealTimeClock(),
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test::FrameGenerator::CreateSquareGenerator(
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width, height, absl::nullopt, absl::nullopt),
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30, *task_queue_factory);
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frame_generators[i] = frame_generator;
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send_streams[i]->SetSource(frame_generator,
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2018-05-16 14:20:41 -07:00
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DegradationPreference::MAINTAIN_FRAMERATE);
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2019-04-18 14:34:16 +02:00
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frame_generator->Init();
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frame_generator->Start();
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2018-02-02 16:24:16 +01:00
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}
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});
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Wait();
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2019-11-08 16:17:41 +01:00
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SendTask(RTC_FROM_HERE, task_queue.get(), [&]() {
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2018-02-02 16:24:16 +01:00
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for (size_t i = 0; i < kNumStreams; ++i) {
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frame_generators[i]->Stop();
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sender_call->DestroyVideoSendStream(send_streams[i]);
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receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
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delete frame_generators[i];
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}
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sender_transport.reset();
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receiver_transport.reset();
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sender_call.reset();
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|
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|
receiver_call.reset();
|
|
|
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|
});
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
void MultiStreamTester::UpdateSendConfig(
|
|
|
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|
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) {}
|
|
|
|
|
|
2019-09-30 04:16:28 +02:00
|
|
|
std::unique_ptr<test::DirectTransport> MultiStreamTester::CreateSendTransport(
|
|
|
|
|
TaskQueueBase* task_queue,
|
2018-02-02 16:24:16 +01:00
|
|
|
Call* sender_call) {
|
2019-09-30 04:16:28 +02:00
|
|
|
return std::make_unique<test::DirectTransport>(
|
2018-08-20 13:27:45 +02:00
|
|
|
task_queue,
|
2019-09-17 17:06:18 +02:00
|
|
|
std::make_unique<FakeNetworkPipe>(
|
2018-10-08 12:28:56 +02:00
|
|
|
Clock::GetRealTimeClock(),
|
2019-09-17 17:06:18 +02:00
|
|
|
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
|
2018-08-20 13:27:45 +02:00
|
|
|
sender_call, payload_type_map_);
|
2018-02-02 16:24:16 +01:00
|
|
|
}
|
|
|
|
|
|
2019-09-30 04:16:28 +02:00
|
|
|
std::unique_ptr<test::DirectTransport>
|
|
|
|
|
MultiStreamTester::CreateReceiveTransport(TaskQueueBase* task_queue,
|
|
|
|
|
Call* receiver_call) {
|
|
|
|
|
return std::make_unique<test::DirectTransport>(
|
2018-08-20 13:27:45 +02:00
|
|
|
task_queue,
|
2019-09-17 17:06:18 +02:00
|
|
|
std::make_unique<FakeNetworkPipe>(
|
2018-10-08 12:28:56 +02:00
|
|
|
Clock::GetRealTimeClock(),
|
2019-09-17 17:06:18 +02:00
|
|
|
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
|
2018-08-20 13:27:45 +02:00
|
|
|
receiver_call, payload_type_map_);
|
2018-02-02 16:24:16 +01:00
|
|
|
}
|
|
|
|
|
} // namespace webrtc
|