Remove bwe_rtp_play and add rtp_to_text to the build.
This CL also switches rtp_to_text to ABSL_FLAG.
Bug: webrtc:10616
Change-Id: I6a2ce921e4c622a9fe08e7de724b8c7ed06f3597
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144630
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28502}
diff --git a/BUILD.gn b/BUILD.gn
index c889955..418ff1b 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -52,6 +52,7 @@
"modules:modules_unittests",
"modules/audio_coding:audio_coding_tests",
"modules/audio_processing:audio_processing_tests",
+ "modules/remote_bitrate_estimator:rtp_to_text",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
"pc:peerconnection_unittests",