Enable audio streams to send padding.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 18 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 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.
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr int kBitrateStatisticsWindowMs = 1000;
 
 constexpr size_t kMinFlexfecPacketsToStoreForPacing = 50;
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
@@ skipping 425 matching lines @@
       break;
     bytes_left -= payload_size;
   }
   return bytes_to_send - bytes_left;
 }
 
 size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
   // Always send full padding packets. This is accounted for by the
   // RtpPacketSender, which will make sure we don't send too much padding even
   // 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();
     uint32_t ssrc;
     uint32_t timestamp;
     int64_t capture_time_ms;
     uint16_t sequence_number;
     int payload_type;
     bool over_rtx;
     {
       rtc::CritScope lock(&send_critsect_);
       if (!sending_media_)
         break;
       timestamp = last_rtp_timestamp_;
       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_;
         payload_type = payload_type_;
         over_rtx = false;
       } else {
         // Without abs-send-time or transport sequence number a media packet
         // must be sent before padding so that the timestamps used for
         // estimation are correct.
         if (!media_has_been_sent_ &&
@@ skipping 271 matching lines @@
 
   // RED+ULPFEC.
   int pt_red;
   int pt_fec;
   video_->GetUlpfecConfig(&pt_red, &pt_fec);
   return static_cast<int>(packet.PayloadType()) == pt_red &&
          static_cast<int>(packet.payload()[0]) == pt_fec;
 }
 
 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)
     bytes_sent += SendPadData(bytes - bytes_sent, probe_cluster_id);
   return bytes_sent;
 }
 
 bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
                               StorageType storage,
                               RtpPacketSender::Priority priority) {
@@ skipping 469 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc