webrtc_m130/test/call_test.h

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/*
* Copyright (c) 2014 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.
*/
#ifndef TEST_CALL_TEST_H_
#define TEST_CALL_TEST_H_
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <vector>
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
#include "api/array_view.h"
#include "api/audio/audio_device.h"
#include "api/environment/environment.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/units/time_delta.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_videorenderer.h"
#include "test/fake_vp8_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/rtp_rtcp_observer.h"
#include "test/run_loop.h"
#include "test/scoped_key_value_config.h"
#include "test/test_video_capturer.h"
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
#include "test/video_test_constants.h"
namespace webrtc {
namespace test {
class BaseTest;
class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
public:
CallTest();
virtual ~CallTest();
static const std::map<uint8_t, MediaType> payload_type_map_;
protected:
const Environment& env() const { return env_; }
void SetSendEventLog(std::unique_ptr<RtcEventLog> event_log);
void SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log);
Reland "Delete test/constants.h" This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6. Reason for revert: Failing tests fixed. Original change's description: > Revert "Delete test/constants.h" > > This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de. > > Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate > > Original change's description: > > Delete test/constants.h > > > > It's not possible to use constants.h for all RTP extensions > > after the number of extensions exceeds 14, which is the maximum > > number of one-byte RTP extensions. This is because some extensions > > would have to be assigned a number greater than 14, even if the > > test only involves 14 extensions or less. > > > > For uniformity's sake, this CL also edits some files to use an > > enum as the files involved in this CL, rather than free-floating > > const-ints. > > > > Bug: webrtc:10288 > > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123048 > > Commit-Queue: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26728} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org > > Bug: webrtc:10288, chromium:933127 > Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/123381 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26744} TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954 Bug: webrtc:10288, chromium:933127 Reviewed-on: https://webrtc-review.googlesource.com/c/123384 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-18 23:45:57 +01:00
void RegisterRtpExtension(const RtpExtension& extension);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
// Returns header extensions that can be parsed by the transport.
rtc::ArrayView<const RtpExtension> GetRegisteredExtensions() {
return rtp_extensions_;
}
Reland "Delete test/constants.h" This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6. Reason for revert: Failing tests fixed. Original change's description: > Revert "Delete test/constants.h" > > This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de. > > Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate > > Original change's description: > > Delete test/constants.h > > > > It's not possible to use constants.h for all RTP extensions > > after the number of extensions exceeds 14, which is the maximum > > number of one-byte RTP extensions. This is because some extensions > > would have to be assigned a number greater than 14, even if the > > test only involves 14 extensions or less. > > > > For uniformity's sake, this CL also edits some files to use an > > enum as the files involved in this CL, rather than free-floating > > const-ints. > > > > Bug: webrtc:10288 > > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123048 > > Commit-Queue: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26728} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org > > Bug: webrtc:10288, chromium:933127 > Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/123381 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26744} TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954 Bug: webrtc:10288, chromium:933127 Reviewed-on: https://webrtc-review.googlesource.com/c/123384 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-18 23:45:57 +01:00
// RunBaseTest overwrites the audio_state of the send and receive Call configs
// to simplify test code.
void RunBaseTest(BaseTest* test);
CallConfig SendCallConfig() const;
CallConfig RecvCallConfig() const;
void CreateCalls();
void CreateCalls(CallConfig sender_config, CallConfig receiver_config);
void CreateSenderCall();
void CreateSenderCall(CallConfig config);
void CreateReceiverCall(CallConfig config);
void DestroyCalls();
void CreateVideoSendConfig(VideoSendStream::Config* video_config,
size_t num_video_streams,
size_t num_used_ssrcs,
Transport* send_transport);
void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport);
void SetAudioConfig(const AudioSendStream::Config& config);
void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
void SetReceiveUlpFecConfig(
VideoReceiveStreamInterface::Config* receive_config);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
void CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams) {
CreateSendConfig(num_video_streams, num_audio_streams, num_flexfec_streams,
send_transport_.get());
}
void CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
void CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config) {
CreateMatchingVideoReceiveConfigs(video_send_config,
receive_transport_.get());
}
void CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport);
void CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
void AddMatchingVideoReceiveConfigs(
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
static AudioReceiveStreamInterface::Config CreateMatchingAudioConfig(
const AudioSendStream::Config& send_config,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
Transport* transport,
std::string sync_group);
void CreateMatchingFecConfig(
Transport* transport,
const VideoSendStream::Config& video_send_config);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
void CreateMatchingReceiveConfigs() {
CreateMatchingReceiveConfigs(receive_transport_.get());
}
void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
float speed,
int framerate,
int width,
int height);
void CreateFrameGeneratorCapturer(int framerate, int width, int height);
void CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
void CreateVideoStreams();
void CreateVideoSendStreams();
void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
void CreateAudioStreams();
void CreateFlexfecStreams();
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
// Receiver call must be created before calling CreateSendTransport in order
// to set a receiver.
// Rtp header extensions must be registered (RegisterRtpExtension(..)) before
// the transport is created in order for the receiving call object receive RTP
// packets with extensions.
void CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer);
void CreateReceiveTransport(const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer);
void ConnectVideoSourcesToStreams();
void Start();
void StartVideoSources();
void StartVideoStreams();
void Stop();
void StopVideoStreams();
void DestroyStreams();
void DestroyVideoSendStreams();
void SetFakeVideoCaptureRotation(VideoRotation rotation);
void SetVideoDegradation(DegradationPreference preference);
VideoSendStream::Config* GetVideoSendConfig();
void SetVideoSendConfig(const VideoSendStream::Config& config);
VideoEncoderConfig* GetVideoEncoderConfig();
void SetVideoEncoderConfig(const VideoEncoderConfig& config);
VideoSendStream* GetVideoSendStream();
FlexfecReceiveStream::Config* GetFlexFecConfig();
TaskQueueBase* task_queue() { return task_queue_.get(); }
// RtpPacketSinkInterface implementation.
void OnRtpPacket(const RtpPacketReceived& packet) override;
test::RunLoop loop_;
test::ScopedKeyValueConfig field_trials_;
Environment env_;
Environment send_env_;
Environment recv_env_;
std::unique_ptr<Call> sender_call_;
std::unique_ptr<PacketTransport> send_transport_;
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
SimulatedNetworkInterface* send_simulated_network_ = nullptr;
std::vector<VideoSendStream::Config> video_send_configs_;
std::vector<VideoEncoderConfig> video_encoder_configs_;
std::vector<VideoSendStream*> video_send_streams_;
AudioSendStream::Config audio_send_config_;
AudioSendStream* audio_send_stream_;
std::unique_ptr<Call> receiver_call_;
std::unique_ptr<PacketTransport> receive_transport_;
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
SimulatedNetworkInterface* receive_simulated_network_ = nullptr;
std::vector<VideoReceiveStreamInterface::Config> video_receive_configs_;
std::vector<VideoReceiveStreamInterface*> video_receive_streams_;
std::vector<AudioReceiveStreamInterface::Config> audio_receive_configs_;
std::vector<AudioReceiveStreamInterface*> audio_receive_streams_;
std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
test::FrameGeneratorCapturer* frame_generator_capturer_;
std::vector<std::unique_ptr<TestVideoCapturer>> video_sources_;
DegradationPreference degradation_preference_ =
DegradationPreference::MAINTAIN_FRAMERATE;
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
std::unique_ptr<NetworkStatePredictorFactoryInterface>
network_state_predictor_factory_;
std::unique_ptr<NetworkControllerFactoryInterface>
network_controller_factory_;
test::FunctionVideoEncoderFactory fake_encoder_factory_;
int fake_encoder_max_bitrate_ = -1;
test::FunctionVideoDecoderFactory fake_decoder_factory_;
std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
// Number of simulcast substreams.
size_t num_video_streams_;
size_t num_audio_streams_;
size_t num_flexfec_streams_;
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
test::FakeVideoRenderer fake_renderer_;
private:
std::optional<RtpExtension> GetRtpExtensionByUri(
Reland "Delete test/constants.h" This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6. Reason for revert: Failing tests fixed. Original change's description: > Revert "Delete test/constants.h" > > This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de. > > Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate > > Original change's description: > > Delete test/constants.h > > > > It's not possible to use constants.h for all RTP extensions > > after the number of extensions exceeds 14, which is the maximum > > number of one-byte RTP extensions. This is because some extensions > > would have to be assigned a number greater than 14, even if the > > test only involves 14 extensions or less. > > > > For uniformity's sake, this CL also edits some files to use an > > enum as the files involved in this CL, rather than free-floating > > const-ints. > > > > Bug: webrtc:10288 > > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123048 > > Commit-Queue: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26728} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org > > Bug: webrtc:10288, chromium:933127 > Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/123381 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26744} TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954 Bug: webrtc:10288, chromium:933127 Reviewed-on: https://webrtc-review.googlesource.com/c/123384 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-18 23:45:57 +01:00
const std::string& uri) const;
void AddRtpExtensionByUri(const std::string& uri,
std::vector<RtpExtension>* extensions) const;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
Reland "Delete test/constants.h" This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6. Reason for revert: Failing tests fixed. Original change's description: > Revert "Delete test/constants.h" > > This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de. > > Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate > > Original change's description: > > Delete test/constants.h > > > > It's not possible to use constants.h for all RTP extensions > > after the number of extensions exceeds 14, which is the maximum > > number of one-byte RTP extensions. This is because some extensions > > would have to be assigned a number greater than 14, even if the > > test only involves 14 extensions or less. > > > > For uniformity's sake, this CL also edits some files to use an > > enum as the files involved in this CL, rather than free-floating > > const-ints. > > > > Bug: webrtc:10288 > > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123048 > > Commit-Queue: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26728} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org > > Bug: webrtc:10288, chromium:933127 > Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/123381 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26744} TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954 Bug: webrtc:10288, chromium:933127 Reviewed-on: https://webrtc-review.googlesource.com/c/123384 Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-18 23:45:57 +01:00
std::vector<RtpExtension> rtp_extensions_;
rtc::scoped_refptr<AudioProcessing> apm_send_;
rtc::scoped_refptr<AudioProcessing> apm_recv_;
rtc::scoped_refptr<AudioDeviceModule> fake_send_audio_device_;
rtc::scoped_refptr<AudioDeviceModule> fake_recv_audio_device_;
};
class BaseTest : public RtpRtcpObserver {
public:
BaseTest();
explicit BaseTest(TimeDelta timeout);
virtual ~BaseTest();
virtual void PerformTest() = 0;
virtual bool ShouldCreateReceivers() const = 0;
virtual size_t GetNumVideoStreams() const;
virtual size_t GetNumAudioStreams() const;
virtual size_t GetNumFlexfecStreams() const;
virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
virtual void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device);
virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
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virtual void OnTransportCreated(PacketTransport* to_receiver,
SimulatedNetworkInterface* sender_network,
PacketTransport* to_sender,
SimulatedNetworkInterface* receiver_network);
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
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virtual BuiltInNetworkBehaviorConfig GetSendTransportConfig() const;
virtual BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const;
virtual void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config);
virtual void ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate);
virtual void ModifyVideoDegradationPreference(
DegradationPreference* degradation_preference);
virtual void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>& receive_streams);
virtual void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* receive_configs);
virtual void OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>& receive_streams);
virtual void ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs);
virtual void OnFlexfecStreamsCreated(
const std::vector<FlexfecReceiveStream*>& receive_streams);
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
virtual void OnStreamsStopped();
};
class SendTest : public BaseTest {
public:
explicit SendTest(TimeDelta timeout);
bool ShouldCreateReceivers() const override;
};
class EndToEndTest : public BaseTest {
public:
EndToEndTest();
explicit EndToEndTest(TimeDelta timeout);
bool ShouldCreateReceivers() const override;
};
} // namespace test
} // namespace webrtc
#endif // TEST_CALL_TEST_H_