webrtc_m130/webrtc/test/call_test.cc

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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.
*/
#include "webrtc/test/call_test.h"
#include <algorithm>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
namespace test {
namespace {
const int kVideoRotationRtpExtensionId = 4;
}
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d > > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057 TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-29 23:57:43 -07:00
void CallTest::PayloadDemuxer::SetReceiver(PacketReceiver* receiver) {
receiver_ = receiver;
}
PacketReceiver::DeliveryStatus CallTest::PayloadDemuxer::DeliverPacket(
MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
if (media_type == MediaType::ANY) {
// This simplistic demux logic will not make much sense for RTCP
// packets, but it seems that doesn't matter.
RTC_CHECK_GE(length, 2);
uint8_t pt = packet[1] & 0x7f;
if (pt == kFakeVideoSendPayloadType || pt == kFlexfecPayloadType) {
media_type = MediaType::VIDEO;
} else {
media_type = MediaType::AUDIO;
}
}
return receiver_->DeliverPacket(media_type, packet, length, packet_time);
}
CallTest::CallTest()
: clock_(Clock::GetRealTimeClock()),
video_send_config_(nullptr),
video_send_stream_(nullptr),
audio_send_config_(nullptr),
audio_send_stream_(nullptr),
fake_encoder_(clock_),
num_video_streams_(1),
num_audio_streams_(0),
num_flexfec_streams_(0),
decoder_factory_(CreateBuiltinAudioDecoderFactory()),
fake_send_audio_device_(nullptr),
fake_recv_audio_device_(nullptr) {}
CallTest::~CallTest() {
}
void CallTest::RunBaseTest(BaseTest* test) {
num_video_streams_ = test->GetNumVideoStreams();
num_audio_streams_ = test->GetNumAudioStreams();
num_flexfec_streams_ = test->GetNumFlexfecStreams();
RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
Call::Config send_config(test->GetSenderCallConfig());
if (num_audio_streams_ > 0) {
CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
fake_recv_audio_device_.get());
CreateVoiceEngines();
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe_send_.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
send_config.audio_state = AudioState::Create(audio_state_config);
}
CreateSenderCall(send_config);
if (test->ShouldCreateReceivers()) {
Call::Config recv_config(test->GetReceiverCallConfig());
if (num_audio_streams_ > 0) {
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe_recv_.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
recv_config.audio_state = AudioState::Create(audio_state_config);
}
CreateReceiverCall(recv_config);
}
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
receive_transport_.reset(test->CreateReceiveTransport());
send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
if (test->ShouldCreateReceivers()) {
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d > > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057 TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-29 23:57:43 -07:00
// For tests using only video or only audio, we rely on each test
// configuring the underlying FakeNetworkPipe with the right media
// type. But for tests sending both video and audio over the same
// FakeNetworkPipe, we need to "demux", i.e., setting the
// MediaType based on RTP payload type.
if (num_video_streams_ > 0 && num_audio_streams_ > 0) {
receive_demuxer_.SetReceiver(receiver_call_->Receiver());
send_transport_->SetReceiver(&receive_demuxer_);
send_demuxer_.SetReceiver(sender_call_->Receiver());
receive_transport_->SetReceiver(&send_demuxer_);
} else {
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_->SetReceiver(sender_call_->Receiver());
}
if (num_video_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
if (num_audio_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
} else {
// Sender-only call delivers to itself.
send_transport_->SetReceiver(sender_call_->Receiver());
receive_transport_->SetReceiver(nullptr);
}
CreateSendConfig(num_video_streams_, num_audio_streams_, num_flexfec_streams_,
send_transport_.get());
if (test->ShouldCreateReceivers()) {
CreateMatchingReceiveConfigs(receive_transport_.get());
}
if (num_video_streams_ > 0) {
test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
&video_encoder_config_);
}
if (num_audio_streams_ > 0) {
test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
CreateFlexfecStreams();
test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
}
if (num_video_streams_ > 0) {
CreateVideoStreams();
test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
}
if (num_audio_streams_ > 0) {
CreateAudioStreams();
test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
}
if (num_video_streams_ > 0) {
int width = kDefaultWidth;
int height = kDefaultHeight;
int frame_rate = kDefaultFramerate;
test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
CreateFrameGeneratorCapturer(frame_rate, width, height);
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
}
Start();
test->PerformTest();
send_transport_->StopSending();
receive_transport_->StopSending();
Stop();
DestroyStreams();
DestroyCalls();
if (num_audio_streams_ > 0)
DestroyVoiceEngines();
test->OnTestFinished();
}
void CallTest::Start() {
if (video_send_stream_)
video_send_stream_->Start();
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
video_recv_stream->Start();
if (audio_send_stream_) {
audio_send_stream_->Start();
}
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Start();
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
flexfec_recv_stream->Start();
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
}
void CallTest::Stop() {
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Stop();
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
flexfec_recv_stream->Stop();
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Stop();
if (audio_send_stream_) {
audio_send_stream_->Stop();
}
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
video_recv_stream->Stop();
if (video_send_stream_)
video_send_stream_->Stop();
}
void CallTest::CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config) {
CreateSenderCall(sender_config);
CreateReceiverCall(receiver_config);
}
void CallTest::CreateSenderCall(const Call::Config& config) {
sender_call_.reset(Call::Create(config));
}
void CallTest::CreateReceiverCall(const Call::Config& config) {
receiver_call_.reset(Call::Create(config));
}
void CallTest::DestroyCalls() {
sender_call_.reset();
receiver_call_.reset();
}
void CallTest::CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport) {
RTC_DCHECK(num_video_streams <= kNumSsrcs);
RTC_DCHECK_LE(num_audio_streams, 1);
RTC_DCHECK_LE(num_flexfec_streams, 1);
RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
if (num_video_streams > 0) {
video_send_config_ = VideoSendStream::Config(send_transport);
video_send_config_.encoder_settings.encoder = &fake_encoder_;
video_send_config_.encoder_settings.payload_name = "FAKE";
video_send_config_.encoder_settings.payload_type =
kFakeVideoSendPayloadType;
video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
for (size_t i = 0; i < num_video_streams; ++i)
video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
video_send_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
}
if (num_audio_streams > 0) {
audio_send_config_ = AudioSendStream::Config(send_transport);
audio_send_config_.voe_channel_id = voe_send_.channel_id;
audio_send_config_.rtp.ssrc = kAudioSendSsrc;
audio_send_config_.send_codec_spec.codec_inst =
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
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CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
}
// TODO(brandtr): Update this when we support multistream protection.
if (num_flexfec_streams > 0) {
video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
}
}
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
video_receive_configs_.clear();
allocated_decoders_.clear();
if (num_video_streams_ > 0) {
RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
VideoReceiveStream::Config video_config(rtcp_send_transport);
video_config.rtp.remb = false;
video_config.rtp.transport_cc = true;
video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : video_send_config_.rtp.extensions)
video_config.rtp.extensions.push_back(extension);
video_config.renderer = &fake_renderer_;
for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(video_send_config_.encoder_settings);
allocated_decoders_.push_back(
std::unique_ptr<VideoDecoder>(decoder.decoder));
video_config.decoders.clear();
video_config.decoders.push_back(decoder);
video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
video_receive_configs_.push_back(video_config.Copy());
}
}
RTC_DCHECK_GE(1, num_audio_streams_);
if (num_audio_streams_ == 1) {
RTC_DCHECK_LE(0, voe_send_.channel_id);
AudioReceiveStream::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = rtcp_send_transport;
audio_config.voe_channel_id = voe_recv_.channel_id;
audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
audio_config.decoder_factory = decoder_factory_;
audio_config.decoder_map = {{120, {"opus", 48000, 2}}};
audio_receive_configs_.push_back(audio_config);
}
// TODO(brandtr): Update this when we support multistream protection.
RTC_DCHECK(num_flexfec_streams_ <= 1);
if (num_flexfec_streams_ == 1) {
FlexfecReceiveStream::Config config(rtcp_send_transport);
config.payload_type = kFlexfecPayloadType;
config.remote_ssrc = kFlexfecSendSsrc;
config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
config.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : video_send_config_.rtp.extensions)
config.rtp_header_extensions.push_back(extension);
flexfec_receive_configs_.push_back(config);
}
}
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
float speed,
int framerate,
int width,
int height) {
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
width, height, framerate * speed, clock));
video_send_stream_->SetSource(
frame_generator_capturer_.get(),
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) Reason for revert: This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 Original issue's description: > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > Reason for revert: > Found issue with test case, will add fix to reland cl. > > Original issue's description: > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > Reason for revert: > > Breaks perf tests: > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > Original issue's description: > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > current degradation preference is maintain-resolution rather than > > > balanced. > > > > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > Cr-Commit-Position: refs/heads/master@{#17331} > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2781433002 > Cr-Commit-Position: refs/heads/master@{#17474} > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2783183003 Cr-Commit-Position: refs/heads/master@{#17477}
2017-03-30 10:44:38 -07:00
VideoSendStream::DegradationPreference::kBalanced);
}
void CallTest::CreateFrameGeneratorCapturer(int framerate,
int width,
int height) {
frame_generator_capturer_.reset(
test::FrameGeneratorCapturer::Create(width, height, framerate, clock_));
video_send_stream_->SetSource(
frame_generator_capturer_.get(),
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) Reason for revert: This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780 Original issue's description: > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) > > Reason for revert: > Found issue with test case, will add fix to reland cl. > > Original issue's description: > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) > > > > Reason for revert: > > Breaks perf tests: > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679 > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325 > > > > Original issue's description: > > > Add framerate to VideoSinkWants and ability to signal on overuse > > > > > > In ViEEncoder, try to reduce framerate instead of resolution if the > > > current degradation preference is maintain-resolution rather than > > > balanced. > > > > > > BUG=webrtc:4172 > > > > > > Review-Url: https://codereview.webrtc.org/2716643002 > > > Cr-Commit-Position: refs/heads/master@{#17327} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/72acf2526177bb4dbb5103cd6e165eb4361a5ae6 > > > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > # Skipping CQ checks because original CL landed less than 1 days ago. > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > BUG=webrtc:4172 > > > > Review-Url: https://codereview.webrtc.org/2764133002 > > Cr-Commit-Position: refs/heads/master@{#17331} > > Committed: https://chromium.googlesource.com/external/webrtc/+/8b45b11144c968b4173215c76f78c710c9a2ed0b > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:4172 > > Review-Url: https://codereview.webrtc.org/2781433002 > Cr-Commit-Position: refs/heads/master@{#17474} > Committed: https://chromium.googlesource.com/external/webrtc/+/3ea3c77e93121b1ab9d5e46641e6764f2cca0d51 TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2783183003 Cr-Commit-Position: refs/heads/master@{#17477}
2017-03-30 10:44:38 -07:00
VideoSendStream::DegradationPreference::kBalanced);
}
void CallTest::CreateFakeAudioDevices(
std::unique_ptr<FakeAudioDevice::Capturer> capturer,
std::unique_ptr<FakeAudioDevice::Renderer> renderer) {
fake_send_audio_device_.reset(new FakeAudioDevice(
std::move(capturer), nullptr, 1.f));
fake_recv_audio_device_.reset(new FakeAudioDevice(
nullptr, std::move(renderer), 1.f));
}
void CallTest::CreateVideoStreams() {
RTC_DCHECK(video_send_stream_ == nullptr);
RTC_DCHECK(video_receive_streams_.empty());
RTC_DCHECK(audio_send_stream_ == nullptr);
RTC_DCHECK(audio_receive_streams_.empty());
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), video_encoder_config_.Copy());
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
video_receive_configs_[i].Copy()));
}
}
void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
frame_generator_capturer_->SetFakeRotation(rotation);
}
void CallTest::CreateAudioStreams() {
audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
audio_receive_streams_.push_back(
receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
}
}
void CallTest::CreateFlexfecStreams() {
for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
flexfec_receive_streams_.push_back(
receiver_call_->CreateFlexfecReceiveStream(
flexfec_receive_configs_[i]));
}
}
void CallTest::DestroyStreams() {
if (audio_send_stream_)
sender_call_->DestroyAudioSendStream(audio_send_stream_);
audio_send_stream_ = nullptr;
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
if (video_send_stream_)
sender_call_->DestroyVideoSendStream(video_send_stream_);
video_send_stream_ = nullptr;
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
video_receive_streams_.clear();
allocated_decoders_.clear();
}
void CallTest::CreateVoiceEngines() {
voe_send_.voice_engine = VoiceEngine::Create();
voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
decoder_factory_));
VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
voe_send_.channel_id = voe_send_.base->CreateChannel(config);
EXPECT_GE(voe_send_.channel_id, 0);
voe_recv_.voice_engine = VoiceEngine::Create();
voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
decoder_factory_));
voe_recv_.channel_id = voe_recv_.base->CreateChannel();
EXPECT_GE(voe_recv_.channel_id, 0);
}
void CallTest::DestroyVoiceEngines() {
voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
voe_recv_.channel_id = -1;
voe_recv_.base->Release();
voe_recv_.base = nullptr;
voe_send_.base->DeleteChannel(voe_send_.channel_id);
voe_send_.channel_id = -1;
voe_send_.base->Release();
voe_send_.base = nullptr;
VoiceEngine::Delete(voe_send_.voice_engine);
voe_send_.voice_engine = nullptr;
VoiceEngine::Delete(voe_recv_.voice_engine);
voe_recv_.voice_engine = nullptr;
}
const int CallTest::kDefaultWidth;
const int CallTest::kDefaultHeight;
const int CallTest::kDefaultFramerate;
const int CallTest::kDefaultTimeoutMs = 30 * 1000;
const int CallTest::kLongTimeoutMs = 120 * 1000;
const uint8_t CallTest::kVideoSendPayloadType = 100;
const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
const uint8_t CallTest::kSendRtxPayloadType = 98;
const uint8_t CallTest::kRedPayloadType = 118;
const uint8_t CallTest::kRtxRedPayloadType = 99;
const uint8_t CallTest::kUlpfecPayloadType = 119;
const uint8_t CallTest::kFlexfecPayloadType = 120;
const uint8_t CallTest::kAudioSendPayloadType = 103;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
0xBADCAFF};
const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
0xC0FFEF};
const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
const int CallTest::kNackRtpHistoryMs = 1000;
BaseTest::BaseTest() {}
BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
}
BaseTest::~BaseTest() {
}
std::unique_ptr<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() {
return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000);
}
std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() {
return FakeAudioDevice::CreateDiscardRenderer(48000);
}
void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
FakeAudioDevice* recv_audio_device) {
}
Call::Config BaseTest::GetSenderCallConfig() {
return Call::Config(&event_log_);
}
Call::Config BaseTest::GetReceiverCallConfig() {
return Call::Config(&event_log_);
}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
}
test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d > > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057 TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-29 23:57:43 -07:00
MediaType::VIDEO,
FakeNetworkPipe::Config());
}
test::PacketTransport* BaseTest::CreateReceiveTransport() {
return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) > > Reason for revert: > Reverting since this seems to break multiple WebRTC Perf buildbots > > Original issue's description: > > Don't hardcode MediaType::ANY in FakeNetworkPipe. > > > > Instead let each test set the appropriate media type. This simplifies > > demuxing in Call and later in RtpTransportController. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2774463003 > > Cr-Commit-Position: refs/heads/master@{#17418} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9c47b00e24da2941eb095df5a4459c6d98a8a88d > > TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2784543002 > Cr-Commit-Position: refs/heads/master@{#17427} > Committed: https://chromium.googlesource.com/external/webrtc/+/3a3bd5061089da5327fc549337a8430054d66057 TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2783853002 Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-29 23:57:43 -07:00
MediaType::VIDEO,
FakeNetworkPipe::Config());
}
size_t BaseTest::GetNumVideoStreams() const {
return 1;
}
size_t BaseTest::GetNumAudioStreams() const {
return 0;
}
size_t BaseTest::GetNumFlexfecStreams() const {
return 0;
}
void BaseTest::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {}
void BaseTest::ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate) {}
void BaseTest::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) {}
void BaseTest::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) {}
void BaseTest::OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStream*>& receive_streams) {}
void BaseTest::ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
void BaseTest::OnFlexfecStreamsCreated(
const std::vector<FlexfecReceiveStream*>& receive_streams) {}
void BaseTest::OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer) {
}
void BaseTest::OnTestFinished() {
}
SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
}
bool SendTest::ShouldCreateReceivers() const {
return false;
}
EndToEndTest::EndToEndTest() {}
EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
}
bool EndToEndTest::ShouldCreateReceivers() const {
return true;
}
} // namespace test
} // namespace webrtc