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Unified Diff: webrtc/call/call_perf_tests.cc

Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 months ago
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Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 12cafd9582faff91ce53c4db013d9655579c98fe..81fbdb7d49f9a5ec28d50b60382a73191de9098d 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -672,7 +672,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
encoder_config->streams[0].target_bitrate_bps =
encoder_config->streams[0].max_bitrate_bps = 2000000;
- encoder_config_ = *encoder_config;
+ encoder_config_ = encoder_config->Copy();
}
void OnVideoStreamsCreated(
@@ -686,7 +686,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
<< "Timed out before receiving an initial high bitrate.";
encoder_config_.streams[0].width *= 2;
encoder_config_.streams[0].height *= 2;
- send_stream_->ReconfigureVideoEncoder(encoder_config_);
+ send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy());
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a couple of high bitrate estimates "
"after reconfiguring the send stream.";
« webrtc/audio/audio_send_stream.h ('K') | « webrtc/call/call.cc ('k') | webrtc/config.h » ('j') | no next file with comments »

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