Add support for transport wide sequence numbers

webrtc/modules/pacing/packet_router.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/modules/pacing/include/packet_router.h"
 
+#include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
 
-PacketRouter::PacketRouter()
-    : crit_(CriticalSectionWrapper::CreateCriticalSection()) {
+PacketRouter::PacketRouter() : transport_seq_(0) {
 }
 
 PacketRouter::~PacketRouter() {
+  DCHECK(rtp_modules_.empty());
 }
 
 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&modules_lock_);
   DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
          rtp_modules_.end());
   rtp_modules_.push_back(rtp_module);
 }
 
 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
-  CriticalSectionScoped cs(crit_.get());
-  rtp_modules_.remove(rtp_module);
+  rtc::CritScope cs(&modules_lock_);
+  auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
+  DCHECK(it != rtp_modules_.end());
+  rtp_modules_.erase(it);

[stefan-webrtc, 2015/07/29 12:19:52]

rtp_modules_.remove(rtp_module); as before?

[sprang_webrtc, 2015/07/29 12:27:27]

erase() will directly remove that element from the list, while remove() will
iterate over the entire list and remove all instances where value matches the
argument. Since we have already asserted that there are no duplicates in the
list this is not necessary.

 }
 
 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
                                     uint16_t sequence_number,
                                     int64_t capture_timestamp,
                                     bool retransmission) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&modules_lock_);
   for (auto* rtp_module : rtp_modules_) {
     if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
       return rtp_module->TimeToSendPacket(ssrc, sequence_number,
                                           capture_timestamp, retransmission);
     }
   }
   return true;
 }
 
-size_t PacketRouter::TimeToSendPadding(size_t bytes) {
-  CriticalSectionScoped cs(crit_.get());
-  for (auto* rtp_module : rtp_modules_) {
-    if (rtp_module->SendingMedia())
-      return rtp_module->TimeToSendPadding(bytes);
+size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
+  size_t total_bytes_sent = 0;
+  rtc::CritScope cs(&modules_lock_);
+  for (RtpRtcp* module : rtp_modules_) {
+    if (module->SendingMedia()) {
+      size_t bytes_sent =
+          module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
+      total_bytes_sent += bytes_sent;
+      if (total_bytes_sent >= bytes_to_send)
+        break;
+    }
   }
-  return 0;
+  return total_bytes_sent;
 }
+
+void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {
+  rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number);
+}
+
+uint16_t PacketRouter::AllocateSequenceNumber() {
+  int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_);
+  int desired_prev_seq;
+  int new_seq;
+  do {
+    desired_prev_seq = prev_seq;
+    new_seq = (desired_prev_seq + 1) & 0xFFFF;
+    // Note: CompareAndSwap returns the actual value of transport_seq at the
+    // time the CAS operation was executed. Thus, if prev_seq is returned, the
+    // operation was successful - otherwise we need to retry. Saving the
+    // return value saves us a load on retry.
+    prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq,
+                                              new_seq);
+  } while (prev_seq != desired_prev_seq);
+
+  return new_seq;
+}
+
 }  // namespace webrtc