Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index ae491209b7f22edf1ff3d063a66ded37dfeff200..23bb2e2db1b0a24b6c70e72ef599066d0b5b7198 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -36,6 +36,7 @@ namespace webrtc {
 namespace {
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 constexpr size_t kMaxPaddingLength = 224;
+constexpr size_t kMinAudioPaddingLength = 50;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr size_t kRtpHeaderLength = 12;
 constexpr uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
@@ -483,6 +484,11 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
   // if a single packet is larger than requested.
   size_t padding_bytes_in_packet =

[mflodman, 2017/01/26 14:03:37]

Can we do this if (!audio_configured_) {} else {} instead? I think that would be
easier to read.

[stefan-webrtc, 2017/01/27 12:53:01]

Didn't become that much better, but it's hopefully a bit more readable.

       std::min(MaxPayloadSize(), kMaxPaddingLength);
+  if (audio_configured_) {
+    // Allow smaller padding packets for audio.
+    padding_bytes_in_packet = std::max(std::min(bytes, padding_bytes_in_packet),
+                                       kMinAudioPaddingLength);
+  }
   size_t bytes_sent = 0;
   while (bytes_sent < bytes) {
     int64_t now_ms = clock_->TimeInMilliseconds();
@@ -500,8 +506,9 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
       capture_time_ms = capture_time_ms_;
       if (rtx_ == kRtxOff) {
         // Without RTX we can't send padding in the middle of frames.
-        if (!last_packet_marker_bit_)
+        if (!audio_configured_ && !last_packet_marker_bit_) {

[stefan-webrtc, 2017/01/24 15:34:02]

Audio streams typically don't use the marker bit, and therefore it's not set.

[mflodman, 2017/01/26 14:03:37]

Marker bits are used in the beginning of a talk spurt, so can you add something
to the comment above why we don't wait for a marker bit.

[stefan-webrtc, 2017/01/27 12:53:01]

Done.

[mflodman, 2017/01/27 13:03:39]

Thanks!

           break;
+        }
         ssrc = ssrc_;
         sequence_number = sequence_number_;
         ++sequence_number_;
@@ -793,7 +800,7 @@ bool RTPSender::IsFecPacket(const RtpPacketToSend& packet) const {
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
-  if (audio_configured_ || bytes == 0)
+  if (bytes == 0)
     return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)