commit b0ac94307e1787f83de2b9a2dc3b58309ea8654b
author Minyue Li <minyue@webrtc.org> Tue May 21 09:30:21 2019
committer Commit Bot <commit-bot@chromium.org> Tue May 21 11:14:10 2019
tree 6fa130e8e9c3ac09fd62fe79e640b186a5c09d87
parent 4d29ef063ca917d2b43176e456fb3a127a915373

Avoid encrypting empty audio packet.

Bug: b/132861665
Change-Id: I161ba8697ae88857927f27fa6d3914b7201fdeab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137049
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28006}

audio/channel_send.cc

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index e8360cb..d4266c2 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -495,6 +495,15 @@
                                   uint8_t payloadType,
                                   uint32_t timeStamp,
                                   rtc::ArrayView<const uint8_t> payload) {
+  if (payload.empty()) {
+    if (frameType == AudioFrameType::kEmptyFrame) {
+      // we don't send empty audio RTP packets
+      // no error since we use it to drive DTMF when we use VAD
+      return 0;
+    }
+    return -1;
+  }
+
   if (_includeAudioLevelIndication) {
     // Store current audio level in the RTP sender.
     // The level will be used in combination with voice-activity state

modules/rtp_rtcp/source/rtp_sender_audio.cc

diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 86026ff..168f206 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -221,14 +221,6 @@
     }
     return true;
   }
-  if (payload_size == 0 || payload_data == NULL) {
-    if (frame_type == AudioFrameType::kEmptyFrame) {
-      // we don't send empty audio RTP packets
-      // no error since we use it to drive DTMF when we use VAD
-      return true;
-    }
-    return false;
-  }
 
   std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket();
   packet->SetMarker(MarkerBit(frame_type, payload_type));