Refactor RtpSender and SSRCDatabase a bit.

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 107 matching lines @@
     bool audio,
     Clock* clock,
     Transport* transport,
     RtpAudioFeedback* audio_feedback,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log)
+    RtcEventLog* event_log,
+    SSRCDatabase* ssrc_database)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       random_(clock_->TimeInMicroseconds()),
       bitrates_(new BitrateAggregator(bitrate_callback)),
       total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
-      send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       packet_over_head_(28),
       payload_type_(-1),
       payload_type_map_(),
       rtp_header_extension_map_(),
       transmission_time_offset_(0),
       absolute_send_time_(0),
       rotation_(kVideoRotation_0),
       cvo_mode_(kCVONone),
       transport_sequence_number_(0),
       // NACK.
       nack_byte_count_times_(),
       nack_byte_count_(),
       nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()),
       packet_history_(clock),
       // Statistics
       statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
-      ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
+      ssrc_db_(ssrc_database),
       remote_ssrc_(0),
       sequence_number_forced_(false),
       ssrc_forced_(false),
       timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
       media_has_been_sent_(false),
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
       rtx_payload_type_(-1),
       target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       target_bitrate_(0) {
   memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
   memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
   // We need to seed the random generator.
+  // TODO(holmer,tommi): Why?^^^ (SSRCDatabase does its own seeding).
+  // TODO(holmer,tommi): Note also that TimeInMilliseconds might return 0 on Mac
+  // early on in the process.

[stefan-webrtc, 2016/01/23 17:38:39]

I think it's because timestamps and sequence numbers should start randomly.

[tommi, 2016/01/24 11:51:49]

I might be misunderstanding, so please bear with me :)

The timestamp or tick_util.* implementation that's supposed to return the
current tick count, actually always returns a very low value on first call and
subsequent calls will return a value relative to when the tick count was first
fetched.  It's actually not that random and doesn't reflect the tick count as it
does on other platforms.  Maybe it's simply a bug (pbos knows more).

The call to srand() here, doesn't affect the Random() class from what I can tell
and is also not related to timestamps.  So, I don't see why it is needed or why
calling it should be done in a particular instance of RtpSender.  If the
requirement comes from SSRCDatabase, then I'd move it to the ctor of
SSRCDatabase.

[stefan-webrtc, 2016/01/25 10:09:32]

I didn't actually read the code, but apparently sequence_number_ and timestamp_
are set using random_ nowadays too.

However, on line 607 below rand() is still used.

   srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
-  ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
-  ssrc_rtx_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+  ssrc_ = ssrc_db_->CreateSSRC();  // Can't be 0.
+  ssrc_rtx_ = ssrc_db_->CreateSSRC();  // Can't be 0.
+  // TODO(holmer, tommi): Should we DCHECK ^^^ on the 'Can't be 0' part?

[stefan-webrtc, 2016/01/23 17:38:39]

Feel free to add. According to SsrcDatabase that should be fine.

[tommi, 2016/01/24 11:51:49]

Done.

   bitrates_->set_ssrc(ssrc_);
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
 }
 
 RTPSender::~RTPSender() {
+  // TODO(tommi,holmer: We don't grab locks in the dtor before accessing member

[stefan-webrtc, 2016/01/23 17:38:39]

TODO(tommi,holmer)


I think the only known model for this class is that the user needs to guarantee
that all methods have finished when the destructor is called. Agree that it
would be good to make it more restricted and document that.

[tommi, 2016/01/24 11:51:49]

Done
OK, I added a thread checker to the dtor to start with, which makes sure that
the sender instance is deleted on the same thread it was created on.  We can
build out from there for other methods.

+  // variables but we grab them in all other methods. (what's the design?)
+  // Start documenting what thread we're on in what method so that it's easier
+  // to understand performance attributes and possibly remove locks.
   if (remote_ssrc_ != 0) {
-    ssrc_db_.ReturnSSRC(remote_ssrc_);
+    ssrc_db_->ReturnSSRC(remote_ssrc_);
   }
-  ssrc_db_.ReturnSSRC(ssrc_);
+  ssrc_db_->ReturnSSRC(ssrc_);
 
-  SSRCDatabase::ReturnSSRCDatabase();
   while (!payload_type_map_.empty()) {
     std::map<int8_t, RtpUtility::Payload*>::iterator it =
         payload_type_map_.begin();
     delete it->second;
     payload_type_map_.erase(it);
   }
 }
 
 void RTPSender::SetTargetBitrate(uint32_t bitrate) {
   CriticalSectionScoped cs(target_bitrate_critsect_.get());
@@ skipping 25 matching lines @@
 
 uint32_t RTPSender::NackOverheadRate() const {
   return nack_bitrate_.BitrateLast();
 }
 
 int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
   if (transmission_time_offset > (0x800000 - 1) ||
       transmission_time_offset < -(0x800000 - 1)) {  // Word24.
     return -1;
   }
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   transmission_time_offset_ = transmission_time_offset;
   return 0;
 }
 
 int32_t RTPSender::SetAbsoluteSendTime(uint32_t absolute_send_time) {
   if (absolute_send_time > 0xffffff) {  // UWord24.
     return -1;
   }
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   absolute_send_time_ = absolute_send_time;
   return 0;
 }
 
 void RTPSender::SetVideoRotation(VideoRotation rotation) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   rotation_ = rotation;
 }
 
 int32_t RTPSender::SetTransportSequenceNumber(uint16_t sequence_number) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   transport_sequence_number_ = sequence_number;
   return 0;
 }
 
 int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
                                               uint8_t id) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   if (type == kRtpExtensionVideoRotation) {
     cvo_mode_ = kCVOInactive;
     return rtp_header_extension_map_.RegisterInactive(type, id);
   }
   return rtp_header_extension_map_.Register(type, id);
 }
 
 bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return rtp_header_extension_map_.IsRegistered(type);
 }
 
 int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return rtp_header_extension_map_.Deregister(type);
 }
 
 size_t RTPSender::RtpHeaderExtensionTotalLength() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return rtp_header_extension_map_.GetTotalLengthInBytes();
 }
 
 int32_t RTPSender::RegisterPayload(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     int8_t payload_number,
     uint32_t frequency,
     size_t channels,
     uint32_t rate) {
   assert(payload_name);
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_number);
 
   if (payload_type_map_.end() != it) {
     // We already use this payload type.
     RtpUtility::Payload* payload = it->second;
     assert(payload);
 
     // Check if it's the same as we already have.
@@ skipping 22 matching lines @@
   } else {
     payload = video_->CreateVideoPayload(payload_name, payload_number, rate);
   }
   if (payload) {
     payload_type_map_[payload_number] = payload;
   }
   return ret_val;
 }
 
 int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
 
   if (payload_type_map_.end() == it) {
     return -1;
   }
   RtpUtility::Payload* payload = it->second;
   delete payload;
   payload_type_map_.erase(it);
   return 0;
 }
 
 void RTPSender::SetSendPayloadType(int8_t payload_type) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   payload_type_ = payload_type;
 }
 
 int8_t RTPSender::SendPayloadType() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return payload_type_;
 }
 
 int RTPSender::SendPayloadFrequency() const {
   return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
 }
 
 int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
                                        uint16_t packet_over_head) {
   // Sanity check.
   RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
       << "Invalid max payload length: " << max_payload_length;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   max_payload_length_ = max_payload_length;
   packet_over_head_ = packet_over_head;
   return 0;
 }
 
 size_t RTPSender::MaxDataPayloadLength() const {
   int rtx;
   {
-    CriticalSectionScoped rtx_lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
   if (audio_configured_) {
     return max_payload_length_ - RTPHeaderLength();
   } else {
     return max_payload_length_ - RTPHeaderLength()  // RTP overhead.
            - video_->FECPacketOverhead()            // FEC/ULP/RED overhead.
            - ((rtx) ? 2 : 0);                       // RTX overhead.
   }
 }
 
 size_t RTPSender::MaxPayloadLength() const {
   return max_payload_length_;
 }
 
 uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
 
 void RTPSender::SetRtxStatus(int mode) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   rtx_ = mode;
 }
 
 int RTPSender::RtxStatus() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return rtx_;
 }
 
 void RTPSender::SetRtxSsrc(uint32_t ssrc) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   ssrc_rtx_ = ssrc;
 }
 
 uint32_t RTPSender::RtxSsrc() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return ssrc_rtx_;
 }
 
 void RTPSender::SetRtxPayloadType(int payload_type,
                                   int associated_payload_type) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   RTC_DCHECK_LE(payload_type, 127);
   RTC_DCHECK_LE(associated_payload_type, 127);
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
     return;
   }
 
   rtx_payload_type_map_[associated_payload_type] = payload_type;
   rtx_payload_type_ = payload_type;
 }
 
 std::pair<int, int> RTPSender::RtxPayloadType() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   for (const auto& kv : rtx_payload_type_map_) {
     if (kv.second == rtx_payload_type_) {
       return std::make_pair(rtx_payload_type_, kv.first);
     }
   }
   return std::make_pair(-1, -1);
 }
 
 int32_t RTPSender::CheckPayloadType(int8_t payload_type,
                                     RtpVideoCodecTypes* video_type) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
     return -1;
   }
   if (audio_configured_) {
     int8_t red_pl_type = -1;
     if (audio_->RED(&red_pl_type) == 0) {
       // We have configured RED.
       if (red_pl_type == payload_type) {
@@ skipping 21 matching lines @@
   if (!payload->audio && !audio_configured_) {
     video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
     *video_type = payload->typeSpecific.Video.videoCodecType;
     video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
   }
   return 0;
 }
 
 RTPSenderInterface::CVOMode RTPSender::ActivateCVORtpHeaderExtension() {
   if (cvo_mode_ == kCVOInactive) {
-    CriticalSectionScoped cs(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) {
       cvo_mode_ = kCVOActivated;
     }
   }
   return cvo_mode_;
 }
 
 int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                                     int8_t payload_type,
                                     uint32_t capture_timestamp,
                                     int64_t capture_time_ms,
                                     const uint8_t* payload_data,
                                     size_t payload_size,
                                     const RTPFragmentationHeader* fragmentation,
                                     const RTPVideoHeader* rtp_hdr) {
   uint32_t ssrc;
   {
     // Drop this packet if we're not sending media packets.
-    CriticalSectionScoped cs(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     ssrc = ssrc_;
     if (!sending_media_) {
       return 0;
     }
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
     LOG(LS_ERROR) << "Don't send data with unknown payload type: "
                   << static_cast<int>(payload_type) << ".";
     return -1;
@@ skipping 31 matching lines @@
   }
   if (frame_count_observer_) {
     frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
   }
 
   return ret_val;
 }
 
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
   {
-    CriticalSectionScoped cs(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     if ((rtx_ & kRtxRedundantPayloads) == 0)
       return 0;
   }
 
   uint8_t buffer[IP_PACKET_SIZE];
   int bytes_left = static_cast<int>(bytes_to_send);
   while (bytes_left > 0) {
     size_t length = bytes_left;
     int64_t capture_time_ms;
     if (!packet_history_.GetBestFittingPacket(buffer, &length,
@@ skipping 41 matching lines @@
                              transport_sequence_number_allocator_;
   for (; bytes > 0; bytes -= padding_bytes_in_packet) {
     if (bytes < padding_bytes_in_packet)
       bytes = padding_bytes_in_packet;
 
     uint32_t ssrc;
     uint16_t sequence_number;
     int payload_type;
     bool over_rtx;
     {
-      CriticalSectionScoped cs(send_critsect_.get());
+      rtc::CritScope lock(&send_critsect_);
       if (!timestamp_provided) {
         timestamp = 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_)
           return 0;
         ssrc = ssrc_;
         sequence_number = sequence_number_;
@@ skipping 93 matching lines @@
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(
         RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
         corrected_capture_tims_ms, length - header.headerLength, true);
 
     return length;
   }
   int rtx = kRtxOff;
   {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
   if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
                             (rtx & kRtxRetransmitted) > 0, true)) {
     return -1;
   }
   return static_cast<int32_t>(length);
 }
 
 bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
@@ skipping 77 matching lines @@
     UpdateNACKBitRate(bytes_re_sent, now);
   }
 }
 
 bool RTPSender::ProcessNACKBitRate(uint32_t now) {
   uint32_t num = 0;
   size_t byte_count = 0;
   const uint32_t kAvgIntervalMs = 1000;
   uint32_t target_bitrate = GetTargetBitrate();
 
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   if (target_bitrate == 0) {
     return true;
   }
   for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
     if ((now - nack_byte_count_times_[num]) > kAvgIntervalMs) {
       // Don't use data older than 1sec.
       break;
     } else {
       byte_count += nack_byte_count_[num];
     }
   }
   uint32_t time_interval = kAvgIntervalMs;
   if (num == NACK_BYTECOUNT_SIZE) {
     // More than NACK_BYTECOUNT_SIZE nack messages has been received
     // during the last msg_interval.
     if (nack_byte_count_times_[num - 1] <= now) {
       time_interval = now - nack_byte_count_times_[num - 1];
     }
   }
   return (byte_count * 8) < (target_bitrate / 1000 * time_interval);
 }
 
 void RTPSender::UpdateNACKBitRate(uint32_t bytes, int64_t now) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   if (bytes == 0)
     return;
   nack_bitrate_.Update(bytes);
   // Save bitrate statistics.
   // Shift all but first time.
   for (int i = NACK_BYTECOUNT_SIZE - 2; i >= 0; i--) {
     nack_byte_count_[i + 1] = nack_byte_count_[i];
     nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
   }
   nack_byte_count_[0] = bytes;
@@ skipping 15 matching lines @@
                                                &length,
                                                &stored_time_ms)) {
     // Packet cannot be found. Allow sending to continue.
     return true;
   }
   if (!retransmission && capture_time_ms > 0) {
     UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
   }
   int rtx;
   {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
   return PrepareAndSendPacket(data_buffer,
                               length,
                               capture_time_ms,
                               retransmission && (rtx & kRtxRetransmitted) > 0,
                               retransmission);
 }
 
 bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
@@ skipping 37 matching lines @@
     options.packet_id =
         UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
   }
 
   if (using_transport_seq && transport_feedback_observer_) {
     transport_feedback_observer_->AddPacket(options.packet_id, length, true);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
                  is_retransmit);
   return ret;
 }
 
 void RTPSender::UpdateRtpStats(const uint8_t* buffer,
                                size_t packet_length,
                                const RTPHeader& header,
@@ skipping 39 matching lines @@
   video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
   if (audio_configured_ || bytes == 0)
     return 0;
   {
-    CriticalSectionScoped cs(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
       return 0;
   }
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);
   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
@@ skipping 51 matching lines @@
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
   // ready for a retransmission.
   packet_history_.SetSent(rtp_header.sequenceNumber);
 
   if (!sent)
     return -1;
 
   {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer, length, rtp_header, false, false);
   return 0;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_)
     return;
 
   uint32_t ssrc;
   int avg_delay_ms = 0;
   int max_delay_ms = 0;
   {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     ssrc = ssrc_;
   }
   {
     CriticalSectionScoped cs(statistics_crit_.get());
     // TODO(holmer): Compute this iteratively instead.
     send_delays_[now_ms] = now_ms - capture_time_ms;
     send_delays_.erase(send_delays_.begin(),
                        send_delays_.lower_bound(now_ms -
                        kSendSideDelayWindowMs));
     int num_delays = 0;
     for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
          it != send_delays_.end(); ++it) {
       max_delay_ms = std::max(max_delay_ms, it->second);
       avg_delay_ms += it->second;
       ++num_delays;
     }
     if (num_delays == 0)
       return;
     avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
   }
   send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
                                                   ssrc);
 }
 
 void RTPSender::ProcessBitrate() {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   total_bitrate_sent_.Process();
   nack_bitrate_.Process();
   if (audio_configured_) {
     return;
   }
   video_->ProcessBitrate();
 }
 
 size_t RTPSender::RTPHeaderLength() const {
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   size_t rtp_header_length = kRtpHeaderLength;
   rtp_header_length += sizeof(uint32_t) * csrcs_.size();
   rtp_header_length += RtpHeaderExtensionTotalLength();
   return rtp_header_length;
 }
 
 uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   uint16_t first_allocated_sequence_number = sequence_number_;
   sequence_number_ += packets_to_send;
   return first_allocated_sequence_number;
 }
 
 void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
                                 StreamDataCounters* rtx_stats) const {
   CriticalSectionScoped lock(statistics_crit_.get());
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
@@ skipping 38 matching lines @@
 }
 
 int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer,
                                   int8_t payload_type,
                                   bool marker_bit,
                                   uint32_t capture_timestamp,
                                   int64_t capture_time_ms,
                                   bool timestamp_provided,
                                   bool inc_sequence_number) {
   assert(payload_type >= 0);
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   if (timestamp_provided) {
     timestamp_ = start_timestamp_ + capture_timestamp;
   } else {
     // Make a unique time stamp.
     // We can't inc by the actual time, since then we increase the risk of back
     // timing.
     timestamp_++;
   }
   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
@@ skipping 286 matching lines @@
 
   *extension_offset = block_pos;
   return ExtensionStatus::kOk;
 }
 
 void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
                                              size_t rtp_packet_length,
                                              const RTPHeader& rtp_header,
                                              int64_t time_diff_ms) const {
   size_t offset;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   switch (VerifyExtension(kRtpExtensionTransmissionTimeOffset, rtp_packet,
                           rtp_packet_length, rtp_header,
                           kTransmissionTimeOffsetLength, &offset)) {
     case ExtensionStatus::kNotRegistered:
       return;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update transmission time offset.";
       return;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   // Update transmission offset field (converting to a 90 kHz timestamp).
   ByteWriter<int32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
                                          time_diff_ms * 90);  // RTP timestamp.
 }
 
 bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet,
                                  size_t rtp_packet_length,
                                  const RTPHeader& rtp_header,
                                  bool is_voiced,
                                  uint8_t dBov) const {
   size_t offset;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   switch (VerifyExtension(kRtpExtensionAudioLevel, rtp_packet,
                           rtp_packet_length, rtp_header, kAudioLevelLength,
                           &offset)) {
     case ExtensionStatus::kNotRegistered:
       return false;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update audio level.";
       return false;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   rtp_packet[offset + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
   return true;
 }
 
 bool RTPSender::UpdateVideoRotation(uint8_t* rtp_packet,
                                     size_t rtp_packet_length,
                                     const RTPHeader& rtp_header,
                                     VideoRotation rotation) const {
   size_t offset;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   switch (VerifyExtension(kRtpExtensionVideoRotation, rtp_packet,
                           rtp_packet_length, rtp_header, kVideoRotationLength,
                           &offset)) {
     case ExtensionStatus::kNotRegistered:
       return false;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update CVO.";
       return false;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   rtp_packet[offset + 1] = ConvertVideoRotationToCVOByte(rotation);
   return true;
 }
 
 void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                                        size_t rtp_packet_length,
                                        const RTPHeader& rtp_header,
                                        int64_t now_ms) const {
   size_t offset;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   switch (VerifyExtension(kRtpExtensionAbsoluteSendTime, rtp_packet,
                           rtp_packet_length, rtp_header,
                           kAbsoluteSendTimeLength, &offset)) {
     case ExtensionStatus::kNotRegistered:
       return;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update absolute send time";
       return;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
   // fractional part).
   ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
                                           ConvertMsTo24Bits(now_ms));
 }
 
 uint16_t RTPSender::UpdateTransportSequenceNumber(
     uint8_t* rtp_packet,
     size_t rtp_packet_length,
     const RTPHeader& rtp_header) const {
   size_t offset;
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet,
                           rtp_packet_length, rtp_header,
                           kTransportSequenceNumberLength, &offset)) {
     case ExtensionStatus::kNotRegistered:
       return 0;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update transport sequence number";
       return 0;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
   BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
   return seq;
 }
 
 void RTPSender::SetSendingStatus(bool enabled) {
   if (enabled) {
     uint32_t frequency_hz = SendPayloadFrequency();
     uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
 
     // Will be ignored if it's already configured via API.
     SetStartTimestamp(RTPtime, false);
   } else {
-    CriticalSectionScoped lock(send_critsect_.get());
+    rtc::CritScope lock(&send_critsect_);
     if (!ssrc_forced_) {
       // Generate a new SSRC.
-      ssrc_db_.ReturnSSRC(ssrc_);
-      ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+      ssrc_db_->ReturnSSRC(ssrc_);
+      ssrc_ = ssrc_db_->CreateSSRC();  // Can't be 0.
       bitrates_->set_ssrc(ssrc_);
     }
     // Don't initialize seq number if SSRC passed externally.
     if (!sequence_number_forced_ && !ssrc_forced_) {
       // Generate a new sequence number.
       sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
     }
   }
 }
 
 void RTPSender::SetSendingMediaStatus(bool enabled) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   sending_media_ = enabled;
 }
 
 bool RTPSender::SendingMedia() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return sending_media_;
 }
 
 uint32_t RTPSender::Timestamp() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return timestamp_;
 }
 
 void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   if (force) {
     start_timestamp_forced_ = true;
     start_timestamp_ = timestamp;
   } else {
     if (!start_timestamp_forced_) {
       start_timestamp_ = timestamp;
     }
   }
 }
 
 uint32_t RTPSender::StartTimestamp() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return start_timestamp_;
 }
 
 uint32_t RTPSender::GenerateNewSSRC() {
   // If configured via API, return 0.
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   if (ssrc_forced_) {
     return 0;
   }
-  ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
+  ssrc_ = ssrc_db_->CreateSSRC();  // Can't be 0.
   bitrates_->set_ssrc(ssrc_);
   return ssrc_;
 }
 
 void RTPSender::SetSSRC(uint32_t ssrc) {
   // This is configured via the API.
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   if (ssrc_ == ssrc && ssrc_forced_) {
     return;  // Since it's same ssrc, don't reset anything.
   }
   ssrc_forced_ = true;
-  ssrc_db_.ReturnSSRC(ssrc_);
-  ssrc_db_.RegisterSSRC(ssrc);
+  ssrc_db_->ReturnSSRC(ssrc_);
+  ssrc_db_->RegisterSSRC(ssrc);
   ssrc_ = ssrc;
   bitrates_->set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
 uint32_t RTPSender::SSRC() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return ssrc_;
 }
 
 void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   csrcs_ = csrcs;
 }
 
 void RTPSender::SetSequenceNumber(uint16_t seq) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   sequence_number_forced_ = true;
   sequence_number_ = seq;
 }
 
 uint16_t RTPSender::SequenceNumber() const {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   return sequence_number_;
 }
 
 // Audio.
 int32_t RTPSender::SendTelephoneEvent(uint8_t key,
                                       uint16_t time_ms,
                                       uint8_t level) {
   if (!audio_configured_) {
     return -1;
   }
@@ skipping 56 matching lines @@
     const FecProtectionParams *key_params) {
   if (audio_configured_) {
     return -1;
   }
   video_->SetFecParameters(delta_params, key_params);
   return 0;
 }
 
 void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
                                uint8_t* buffer_rtx) {
-  CriticalSectionScoped cs(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   uint8_t* data_buffer_rtx = buffer_rtx;
   // Add RTX header.
   RtpUtility::RtpHeaderParser rtp_parser(
       reinterpret_cast<const uint8_t*>(buffer), *length);
 
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
 
   // Add original RTP header.
   memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
@@ skipping 34 matching lines @@
   CriticalSectionScoped cs(statistics_crit_.get());
   return rtp_stats_callback_;
 }
 
 uint32_t RTPSender::BitrateSent() const {
   return total_bitrate_sent_.BitrateLast();
 }
 
 void RTPSender::SetRtpState(const RtpState& rtp_state) {
   SetStartTimestamp(rtp_state.start_timestamp, true);
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   sequence_number_ = rtp_state.sequence_number;
   sequence_number_forced_ = true;
   timestamp_ = rtp_state.timestamp;
   capture_time_ms_ = rtp_state.capture_time_ms;
   last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
   media_has_been_sent_ = rtp_state.media_has_been_sent;
 }
 
 RtpState RTPSender::GetRtpState() const {
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_;
   state.start_timestamp = start_timestamp_;
   state.timestamp = timestamp_;
   state.capture_time_ms = capture_time_ms_;
   state.last_timestamp_time_ms = last_timestamp_time_ms_;
   state.media_has_been_sent = media_has_been_sent_;
 
   return state;
 }
 
 void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
   sequence_number_rtx_ = rtp_state.sequence_number;
 }
 
 RtpState RTPSender::GetRtxRtpState() const {
-  CriticalSectionScoped lock(send_critsect_.get());
+  rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc