Simplifies audio priority rate config in scenario tests.
Bug: webrtc:9510
Change-Id: Iecd2caa8d4353c64ec351969f999c8ed59c3a07d
Reviewed-on: https://webrtc-review.googlesource.com/c/110614
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25606}diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index 521bcb7..dc8e02c 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -9,6 +9,7 @@
*/
#include "test/scenario/audio_stream.h"
+#include "rtc_base/bitrateallocationstrategy.h"
#include "test/call_test.h"
#if WEBRTC_ENABLE_PROTOBUF
@@ -131,8 +132,12 @@
{RtpExtension::kTransportSequenceNumberUri, 8}};
}
- if (config.stream.rate_allocation_priority) {
+ if (config.encoder.priority_rate) {
send_config.track_id = sender->GetNextPriorityId();
+ sender_->call_->SetBitrateAllocationStrategy(
+ absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
+ send_config.track_id,
+ config.encoder.priority_rate->bps<uint32_t>()));
}
send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
