| Index: webrtc/video/end_to_end_tests.cc
|
| diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
|
| index 8e3105f7411d9af37012cb7ea6635ef7edf4b79a..5363071e1715a793092c1425b12ecdb1983374bd 100644
|
| --- a/webrtc/video/end_to_end_tests.cc
|
| +++ b/webrtc/video/end_to_end_tests.cc
|
| @@ -1204,8 +1204,8 @@ class MultiStreamTest {
|
|
|
| UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
|
|
|
| - send_streams[i] =
|
| - sender_call->CreateVideoSendStream(send_config, encoder_config);
|
| + send_streams[i] = sender_call->CreateVideoSendStream(
|
| + send_config.Copy(), encoder_config.Copy());
|
| send_streams[i]->Start();
|
|
|
| VideoReceiveStream::Config receive_config(receiver_transport.get());
|
| @@ -2395,7 +2395,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
|
| }
|
| }
|
|
|
| - video_encoder_config_all_streams_ = *encoder_config;
|
| + video_encoder_config_all_streams_ = encoder_config->Copy();
|
| if (send_single_ssrc_first_)
|
| encoder_config->streams.resize(1);
|
| }
|
| @@ -2414,7 +2414,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
|
| if (send_single_ssrc_first_) {
|
| // Set full simulcast and continue with the rest of the SSRCs.
|
| send_stream_->ReconfigureVideoEncoder(
|
| - video_encoder_config_all_streams_);
|
| + std::move(video_encoder_config_all_streams_));
|
| EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
|
| }
|
| }
|
| @@ -3055,7 +3055,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
|
|
|
| // Use the same total bitrates when sending a single stream to avoid lowering
|
| // the bitrate estimate and requiring a subsequent rampup.
|
| - VideoEncoderConfig one_stream = video_encoder_config_;
|
| + VideoEncoderConfig one_stream = video_encoder_config_.Copy();
|
| one_stream.streams.resize(1);
|
| for (size_t i = 1; i < video_encoder_config_.streams.size(); ++i) {
|
| one_stream.streams.front().min_bitrate_bps +=
|
| @@ -3082,8 +3082,8 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
|
| sender_call_->DestroyVideoSendStream(video_send_stream_);
|
|
|
| // Re-create VideoSendStream with only one stream.
|
| - video_send_stream_ =
|
| - sender_call_->CreateVideoSendStream(video_send_config_, one_stream);
|
| + video_send_stream_ = sender_call_->CreateVideoSendStream(
|
| + video_send_config_.Copy(), one_stream.Copy());
|
| video_send_stream_->Start();
|
| CreateFrameGeneratorCapturer();
|
| frame_generator_capturer_->Start();
|
| @@ -3092,18 +3092,18 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
|
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
|
|
|
| // Reconfigure back to use all streams.
|
| - video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
|
| + video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
|
| observer.ResetExpectedSsrcs(kNumSsrcs);
|
| EXPECT_TRUE(observer.Wait())
|
| << "Timed out waiting for all SSRCs to send packets.";
|
|
|
| // Reconfigure down to one stream.
|
| - video_send_stream_->ReconfigureVideoEncoder(one_stream);
|
| + video_send_stream_->ReconfigureVideoEncoder(one_stream.Copy());
|
| observer.ResetExpectedSsrcs(1);
|
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
|
|
|
| // Reconfigure back to use all streams.
|
| - video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
|
| + video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
|
| observer.ResetExpectedSsrcs(kNumSsrcs);
|
| EXPECT_TRUE(observer.Wait())
|
| << "Timed out waiting for all SSRCs to send packets.";
|
|
|