Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData
It appears unused everywhere. It will be deleted in a followup cl.
Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 55f7af0..38b2362 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -316,7 +316,7 @@
// Send data to the channel. "channel" will handle the loss simulation.
channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
- rtp_timestamp_, bitstream, bitstream_len_byte, NULL);
+ rtp_timestamp_, bitstream, bitstream_len_byte);
if (first_packet) {
first_packet = false;
start_time_stamp = rtp_timestamp_;