Remove SendPacer from ViEEncoder

webrtc/modules/pacing/paced_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 228 matching lines @@
 
   int target_rate_kbps_;
   int bytes_remaining_;
 };
 }  // namespace paced_sender
 
 const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;
 
 PacedSender::PacedSender(Clock* clock,
-                         Callback* callback,
-                         int bitrate_kbps,
-                         int max_bitrate_kbps,
-                         int min_bitrate_kbps)
+                         SenderDelegate* delegate,
+                         int estimated_bitrate_bps)
     : clock_(clock),
-      callback_(callback),
+      delegate_(delegate),
       critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       paused_(false),
       probing_enabled_(true),
-      media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)),
-      padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)),
+      media_budget_(new paced_sender::IntervalBudget(
+          estimated_bitrate_bps / 1000 * kDefaultPaceMultiplier)),

[stefan-webrtc, 2016/04/29 10:48:38]

I'm slightly hesitant to moving kDefaultPaceMultiplier in here. It removes the
possibility of setting different multipliers for different applications / source
types. For instance if we know we're only operating on a local wifi (not going
over the open internet) we may want to have a higher multiplier. Do you think
there's a big benefit to having it here?

[perkj_webrtc, 2016/05/02 11:29:10]

I think its very weird that it has to be set outside the pacer. But if you ever
want to implement what you suggest it should be easy to add an API to the
PacedSender to configure it. 

+      padding_budget_(new paced_sender::IntervalBudget(0)),
       prober_(new BitrateProber()),
-      bitrate_bps_(1000 * bitrate_kbps),
-      max_bitrate_kbps_(max_bitrate_kbps),
+      estimated_bitrate_bps_(estimated_bitrate_bps),
+      min_send_bitrate_kbps_(0u),
+      target_bitrate_kbps_(estimated_bitrate_bps / 1000 *

[stefan-webrtc, 2016/04/29 10:48:38]

pacing_bitrate_kbps_, perhaps?

[perkj_webrtc, 2016/05/02 11:29:10]

Done.

+                           kDefaultPaceMultiplier),
       time_last_update_us_(clock->TimeInMicroseconds()),
       packets_(new paced_sender::PacketQueue(clock)),
       packet_counter_(0) {
   UpdateBytesPerInterval(kMinPacketLimitMs);
 }
 
 PacedSender::~PacedSender() {}
 
 void PacedSender::Pause() {
   CriticalSectionScoped cs(critsect_.get());
   paused_ = true;
 }
 
 void PacedSender::Resume() {
   CriticalSectionScoped cs(critsect_.get());
   paused_ = false;
 }
 
 void PacedSender::SetProbingEnabled(bool enabled) {
   RTC_CHECK_EQ(0u, packet_counter_);
   probing_enabled_ = enabled;
 }
 
-void PacedSender::UpdateBitrate(int bitrate_kbps,
-                                int max_bitrate_kbps,
-                                int min_bitrate_kbps) {
+void PacedSender::SetMinimumSendBitrate(int bitrate_bps) {
   CriticalSectionScoped cs(critsect_.get());
-  // Don't set media bitrate here as it may be boosted in order to meet max
-  // queue time constraint. Just update max_bitrate_kbps_ and let media_budget_
-  // be updated in Process().
-  padding_budget_->set_target_rate_kbps(min_bitrate_kbps);
-  bitrate_bps_ = 1000 * bitrate_kbps;
-  max_bitrate_kbps_ = max_bitrate_kbps;
+  min_send_bitrate_kbps_ = bitrate_bps / 1000;
+  target_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
+      kDefaultPaceMultiplier;
+}
+
+void PacedSender::SetNetWorkEstimateTargetBitrate(uint32_t bitrate_bps) {
+  LOG(LS_INFO) << "SetNetWorkEstimateTargetBitrate, bitrate " << bitrate_bps;
+
+  CriticalSectionScoped cs(critsect_.get());
+  estimated_bitrate_bps_ = bitrate_bps;
+  target_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
+      kDefaultPaceMultiplier;
+}
+
+void PacedSender::SetPaddingBitrate(int bitrate_bps) {
+  CriticalSectionScoped cs(critsect_.get());
+  padding_budget_->set_target_rate_kbps(bitrate_bps / 1000);
 }
 
 void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                uint32_t ssrc,
                                uint16_t sequence_number,
                                int64_t capture_time_ms,
                                size_t bytes,
                                bool retransmission) {
   CriticalSectionScoped cs(critsect_.get());
 
   if (probing_enabled_ && !prober_->IsProbing())
     prober_->SetEnabled(true);
   int64_t now_ms = clock_->TimeInMilliseconds();
-  prober_->OnIncomingPacket(bitrate_bps_, bytes, now_ms);
+  prober_->OnIncomingPacket(estimated_bitrate_bps_, bytes, now_ms);
 
   if (capture_time_ms < 0)
     capture_time_ms = now_ms;
 
   packets_->Push(paced_sender::Packet(priority, ssrc, sequence_number,
                                       capture_time_ms, now_ms, bytes,
                                       retransmission, packet_counter_++));
 }
 
 int64_t PacedSender::ExpectedQueueTimeMs() const {
   CriticalSectionScoped cs(critsect_.get());
-  RTC_DCHECK_GT(max_bitrate_kbps_, 0);
-  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / max_bitrate_kbps_);
+  RTC_DCHECK_GT(target_bitrate_kbps_, 0u);
+  return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
+                              target_bitrate_kbps_);
 }
 
 size_t PacedSender::QueueSizePackets() const {
   CriticalSectionScoped cs(critsect_.get());
   return packets_->SizeInPackets();
 }
 
 int64_t PacedSender::QueueInMs() const {
   CriticalSectionScoped cs(critsect_.get());
 
@@ skipping 20 matching lines @@
   int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_us_;
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
   return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
 }
 
 void PacedSender::Process() {
   int64_t now_us = clock_->TimeInMicroseconds();
   CriticalSectionScoped cs(critsect_.get());
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
-  int target_bitrate_kbps = max_bitrate_kbps_;
+  int target_bitrate_kbps = target_bitrate_kbps_;
   // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
   if (!paused_ && elapsed_time_ms > 0) {
     size_t queue_size_bytes = packets_->SizeInBytes();
     if (queue_size_bytes > 0) {
       // Assuming equal size packets and input/output rate, the average packet
       // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
       // time constraint shall be met. Determine bitrate needed for that.
       packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
       int64_t avg_time_left_ms = std::max<int64_t>(
           1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs());
@@ skipping 44 matching lines @@
     SendPadding(static_cast<size_t>(padding_needed));
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
   // TODO(holmer): Because of this bug issue 5307 we have to send audio
   // packets even when the pacer is paused. Here we assume audio packets are
   // always high priority and that they are the only high priority packets.
   if (paused_ && packet.priority != kHighPriority)
     return false;
   critsect_->Leave();
-  const bool success = callback_->TimeToSendPacket(packet.ssrc,
-                                                   packet.sequence_number,
-                                                   packet.capture_time_ms,
-                                                   packet.retransmission);
+  const bool success = delegate_->TimeToSendPacket(
+      packet.ssrc, packet.sequence_number, packet.capture_time_ms,
+      packet.retransmission);
   critsect_->Enter();
 
   if (success) {
     prober_->PacketSent(clock_->TimeInMilliseconds(), packet.bytes);
     // TODO(holmer): High priority packets should only be accounted for if we
     // are allocating bandwidth for audio.
     if (packet.priority != kHighPriority) {
       // Update media bytes sent.
       media_budget_->UseBudget(packet.bytes);
       padding_budget_->UseBudget(packet.bytes);
     }
   }
 
   return success;
 }
 
 void PacedSender::SendPadding(size_t padding_needed) {
   critsect_->Leave();
-  size_t bytes_sent = callback_->TimeToSendPadding(padding_needed);
+  size_t bytes_sent = delegate_->TimeToSendPadding(padding_needed);
   critsect_->Enter();
 
   if (bytes_sent > 0) {
     prober_->PacketSent(clock_->TimeInMilliseconds(), bytes_sent);
     media_budget_->UseBudget(bytes_sent);
     padding_budget_->UseBudget(bytes_sent);
   }
 }
 
 void PacedSender::UpdateBytesPerInterval(int64_t delta_time_ms) {
   media_budget_->IncreaseBudget(delta_time_ms);
   padding_budget_->IncreaseBudget(delta_time_ms);
 }
 }  // namespace webrtc