commit 70efddeceda987c2e598bf8e0d0225d578f566d7
author Erik Språng <sprang@webrtc.org> Wed Aug 21 11:36:20 2019
committer Commit Bot <commit-bot@chromium.org> Wed Aug 21 12:44:09 2019
tree 86225111f51d35a002876ab759f43c0f22669b90
parent 21e99dac24b99dcae540fa1e09a7362f2cd5994d

Set local ssrc at construction of Rtp module

The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.

Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}

audio/audio_receive_stream.cc

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index c4abea0..c093342 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -78,8 +78,9 @@
   return voe::CreateChannelReceive(
       clock, module_process_thread, internal_audio_state->audio_device_module(),
       config.media_transport_config, config.rtcp_send_transport, event_log,
-      config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
-      config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
+      config.rtp.local_ssrc, config.rtp.remote_ssrc,
+      config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
+      config.jitter_buffer_min_delay_ms,
       config.jitter_buffer_enable_rtx_handling, config.decoder_factory,
       config.codec_pair_id, config.frame_decryptor, config.crypto_options);
 }
@@ -381,12 +382,9 @@
   RTC_DCHECK(first_time ||
              old_config.decoder_factory == new_config.decoder_factory);
 
-  if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
-    channel_receive->SetLocalSSRC(new_config.rtp.local_ssrc);
-  }
-
   if (!first_time) {
-    // Remote ssrc can't be changed mid-stream.
+    // SSRC can't be changed mid-stream.
+    RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc);
     RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
   }