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Adds support to change transport routes in Scenario tests.

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This CL makes it possible to change transport routes while running
a scenario based test.

To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.

Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
This commit is contained in:
Sebastian Jansson
2018-10-22 11:49:03 +02:00
committed by Commit Bot
parent 8d33c0c104
commit 800e121dca
15 changed files with 300 additions and 266 deletions
Download Patch File Download Diff File Expand all files Collapse all files

49
test/scenario/audio_stream.cc
View File

@ -106,9 +106,7 @@ SendAudioStream::SendAudioStream(
}
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
TimeDelta min_frame_length = config.encoder.initial_frame_length;
;
TimeDelta max_frame_length = config.encoder.initial_frame_length;
;
if (field_trial::IsEnabled("WebRTC-Audio-FrameLengthAdaptation") &&
!config.adapt.frame.min_rate_for_20_ms.IsZero()) {
if (!config.adapt.frame.min_rate_for_60_ms.IsZero()) {
@ -118,7 +116,8 @@ SendAudioStream::SendAudioStream(
}
}
DataSize rtp_overhead = DataSize::bytes(12);
DataSize total_overhead = config.stream.packet_overhead + rtp_overhead;
DataSize total_overhead =
sender_->transport_.packet_overhead() + rtp_overhead;
min_rate += total_overhead / max_frame_length;
max_rate += total_overhead / min_frame_length;
}
@ -138,7 +137,7 @@ SendAudioStream::SendAudioStream(
send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
sender->call_->OnAudioTransportOverheadChanged(
config.stream.packet_overhead.bytes());
sender_->transport_.packet_overhead().bytes());
}
}
@ -150,15 +149,6 @@ void SendAudioStream::Start() {
send_stream_->Start();
}
bool SendAudioStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
uint64_t receiver,
Timestamp at_time) {
// Removes added overhead before delivering RTCP packet to sender.
RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
sender_->DeliverPacket(MediaType::AUDIO, packet, at_time);
return true;
}
ReceiveAudioStream::ReceiveAudioStream(
CallClient* receiver,
AudioStreamConfig config,
@ -170,6 +160,7 @@ ReceiveAudioStream::ReceiveAudioStream(
recv_config.rtp.local_ssrc = CallTest::kReceiverLocalAudioSsrc;
recv_config.rtcp_send_transport = feedback_transport;
recv_config.rtp.remote_ssrc = send_stream->ssrc_;
receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
if (config.stream.in_bandwidth_estimation) {
recv_config.rtp.transport_cc = true;
recv_config.rtp.extensions = {
@ -185,49 +176,21 @@ ReceiveAudioStream::~ReceiveAudioStream() {
receiver_->call_->DestroyAudioReceiveStream(receive_stream_);
}
bool ReceiveAudioStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
uint64_t receiver,
Timestamp at_time) {
RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
receiver_->DeliverPacket(MediaType::AUDIO, packet, at_time);
return true;
}
AudioStreamPair::~AudioStreamPair() = default;
AudioStreamPair::AudioStreamPair(
CallClient* sender,
std::vector<NetworkNode*> send_link,
uint64_t send_receiver_id,
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
CallClient* receiver,
std::vector<NetworkNode*> return_link,
uint64_t return_receiver_id,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
AudioStreamConfig config)
: config_(config),
send_link_(send_link),
return_link_(return_link),
send_transport_(sender,
send_link.front(),
send_receiver_id,
config.stream.packet_overhead),
return_transport_(receiver,
return_link.front(),
return_receiver_id,
config.stream.packet_overhead),
send_stream_(sender, config, encoder_factory, &send_transport_),
send_stream_(sender, config, encoder_factory, &sender->transport_),
receive_stream_(receiver,
config,
&send_stream_,
decoder_factory,
&return_transport_) {
NetworkNode::Route(send_transport_.ReceiverId(), send_link_,
&receive_stream_);
NetworkNode::Route(return_transport_.ReceiverId(), return_link_,
&send_stream_);
}
&receiver->transport_) {}
} // namespace test
} // namespace webrtc