Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(81)

Unified Diff: webrtc/test/call_test.cc

Issue 2730073002: Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (Closed)
Patch Set: Moving all of thumbnails out of call test Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 68d6857e29d7208d85a8af114227b5fb951b25f9..fbb94e126f654e9f033e8489d7a8ccc1e8d72d4c 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -226,28 +226,34 @@ void CallTest::CreateSendConfig(size_t num_video_streams,
}
}
+void CallTest::CreateMatchingVideoReceiveConfigs(
+ const VideoSendStream::Config& video_send_config,
+ Transport* rtcp_send_transport) {
+ 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());
+ }
+}
+
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
RTC_DCHECK(video_receive_configs_.empty());
RTC_DCHECK(allocated_decoders_.empty());
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());
- }
+ CreateMatchingVideoReceiveConfigs(video_send_config_, rtcp_send_transport);
}
RTC_DCHECK_GE(1, num_audio_streams_);
@@ -347,6 +353,7 @@ void CallTest::DestroyStreams() {
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);

Powered by Google App Engine
This is Rietveld 408576698