Ensure CallTest derived tests per default set min/max audio bitrate.
This ensure BWE works as intended with transport sequence numbers on
audio.
Tested with webrtc_perf_tests --gtest_filter=CallPerfTest.Min_Bitrate_VideoAndAudio
and --gtest_filter=Rampup*
Bug: webrtc:14854, webrtc:7135, b/266786240
Change-Id: I3b7a743149c22035e582a2157b5f0a93747857cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291523
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39208}
diff --git a/test/call_test.cc b/test/call_test.cc
index 62d1839..7eccf78 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -292,6 +292,8 @@
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
+ audio_send_config.min_bitrate_bps = 6000;
+ audio_send_config.max_bitrate_bps = 60000;
audio_send_config.encoder_factory = audio_encoder_factory_;
SetAudioConfig(audio_send_config);
}