Make sure that the audio stream is allocated with the correct overhead.
This fixes two cases when the allocation is not updated correctly:
- The frame length range is not updated when audio network adaptor is enabled.
- The per-packet overhead is not updated unless the bitrate observer has been reconfigured.
Bug: webrtc:11001
Change-Id: I2ee25f956741a4be08661f874556582dd60a3bd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169848
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30709}
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index e7c6540..de152cd 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -162,6 +162,7 @@
SetupDefaultChannelSend(audio_bwe_enabled);
SetupMockForSetupSendCodec(expect_set_encoder_call);
+ SetupMockForCallEncoder();
// Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
// calls from the default ctor behavior.
@@ -514,7 +515,6 @@
auto stream_config = helper.config();
stream_config.audio_network_adaptor_config = kAnaReconfigString;
- helper.SetupMockForCallEncoder();
send_stream->Reconfigure(stream_config);
}