commit d703cd022f64ddb6225b08719c2e40d1e1aa997a
author Minyue Li <minyue@webrtc.org> Tue May 21 14:13:40 2019
committer Commit Bot <commit-bot@chromium.org> Tue May 21 14:13:52 2019
tree 0f711e09b87efffd7a91e7bcddbdff393da15b0f
parent 19da5ced24220caef35d71b641ecc0133de475e9

Revert "Avoid encrypting empty audio packet."

This reverts commit b0ac94307e1787f83de2b9a2dc3b58309ea8654b.

Reason for revert: failing upstream tests

Original change's description:
> 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}

TBR=brandtr@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org

Change-Id: I856436ad78bcc5310810283bb5547070781d0684
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/132861665
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137518
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28009}

audio/channel_send.cc

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index d4266c2..e8360cb 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -495,15 +495,6 @@
                                   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 168f206..86026ff 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -221,6 +221,14 @@
     }
     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));