Refactor how padding is calculated.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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 324 matching lines @@
   }
   if (video_codec.maxBitrate < kEncoderMinBitrateKbps)
     video_codec.maxBitrate = kEncoderMinBitrateKbps;
 
   RTC_DCHECK_GT(streams[0].max_framerate, 0);
   video_codec.maxFramerate = streams[0].max_framerate;
 
   return video_codec;
 }
 
+int CalulcateMaxPadBitrateBps(const VideoEncoderConfig& config,
+                              bool pad_to_min_bitrate) {
+  int pad_up_to_bitrate_bps = 0;
+  // Calulate max padding bitrate for a multi layer codec.
+  if (config.streams.size() > 1) {
+    // Pad to min bitrate of the highest layer.
+    pad_up_to_bitrate_bps =
+        config.streams[config.streams.size() - 1].min_bitrate_bps;
+    // + target_bitrate_bps of the lower layers.
+    for (size_t i = 0; i < config.streams.size() - 1; ++i) {
+      pad_up_to_bitrate_bps += config.streams[i].target_bitrate_bps;
+    }
+  } else if (pad_to_min_bitrate) {
+    pad_up_to_bitrate_bps = config.streams[0].min_bitrate_bps;
+  }
+
+  pad_up_to_bitrate_bps =
+      std::max(pad_up_to_bitrate_bps, config.min_transmit_bitrate_bps);
+
+  return pad_up_to_bitrate_bps;
+}
+
 }  // namespace
 
 namespace internal {
 VideoSendStream::VideoSendStream(
     int num_cpu_cores,
     ProcessThread* module_process_thread,
     CallStats* call_stats,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
     SendDelayStats* send_delay_stats,
@@ skipping 171 matching lines @@
   return false;
 }
 
 void VideoSendStream::EncoderProcess() {
   RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
                       config_.encoder_settings.encoder,
                       config_.encoder_settings.payload_type,
                       config_.encoder_settings.internal_source));
 
   while (true) {
-    encoder_wakeup_event_.Wait(rtc::Event::kForever);
+    const int kEncodeCheckForActiviyPeriodMs = 1000;
+    encoder_wakeup_event_.Wait(kEncodeCheckForActiviyPeriodMs);
     if (rtc::AtomicOps::AcquireLoad(&stop_encoder_thread_))
       break;
     rtc::Optional<EncoderSettings> encoder_settings;
     {
       rtc::CritScope lock(&encoder_settings_crit_);
       if (pending_encoder_settings_) {
         encoder_settings = pending_encoder_settings_;
         pending_encoder_settings_ = rtc::Optional<EncoderSettings>();
       }
     }
     if (encoder_settings) {
       encoder_settings->video_codec.startBitrate =
           bitrate_allocator_->AddObserver(
               this, encoder_settings->video_codec.minBitrate * 1000,
               encoder_settings->video_codec.maxBitrate * 1000,
+              CalulcateMaxPadBitrateBps(encoder_settings->config,
+                                        config_.suspend_below_min_bitrate),
               !config_.suspend_below_min_bitrate) /
           1000;
 
-      payload_router_.SetSendStreams(encoder_settings->streams);
+      payload_router_.SetSendStreams(encoder_settings->config.streams);
       vie_encoder_.SetEncoder(encoder_settings->video_codec,
-                              encoder_settings->min_transmit_bitrate_bps,
                               payload_router_.MaxPayloadLength());
 
       // vie_encoder_.SetEncoder must be called before this.
       if (config_.suspend_below_min_bitrate)
         video_sender_->SuspendBelowMinBitrate();
 
       // Clear stats for disabled layers.
-      for (size_t i = encoder_settings->streams.size();
+      for (size_t i = encoder_settings->config.streams.size();
            i < config_.rtp.ssrcs.size(); ++i) {
         stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
       }
 
       // We might've gotten new settings while configuring the encoder settings,
       // restart from the top to see if that's the case before trying to encode
       // a frame (which might correspond to the last frame size).
       encoder_wakeup_event_.Set();
       continue;
     }
 
     VideoFrame frame;
     if (input_.GetVideoFrame(&frame)) {
       // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should
       // register as a sink to the VideoSource instead.
       if (config_.pre_encode_callback) {
         config_.pre_encode_callback->OnFrame(frame);
       }
       vie_encoder_.EncodeVideoFrame(frame);
     }
+
+    // TODO(perkj): This is temporary. Activity should be notified by ViEncoder
+    // once it owns its own thread.
+    vie_encoder_.CheckForActivity();
   }
   vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
 }
 
 void VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size());
   VideoCodec video_codec = VideoEncoderConfigToVideoCodec(
       config, config_.encoder_settings.payload_name,
       config_.encoder_settings.payload_type);
   {
     rtc::CritScope lock(&encoder_settings_crit_);
-    pending_encoder_settings_ = rtc::Optional<EncoderSettings>(
-        {video_codec, config.min_transmit_bitrate_bps, config.streams});
+    pending_encoder_settings_ =
+        rtc::Optional<EncoderSettings>({video_codec, config});
   }
   encoder_wakeup_event_.Set();
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::OveruseDetected() {
   if (config_.overuse_callback)
@@ skipping 33 matching lines @@
     }
     if (file_writer) {
       bool ok = file_writer->WriteFrame(encoded_image);
       RTC_DCHECK(ok);
     }
   }
 
   return return_value;
 }
 
+void VideoSendStream::OnEncoderActivityChanged(bool active) {
+  // Notify the |bitrate_allocator_| if the encoder have timed out so it can
+  // stop sending padding if applicable.
+  bitrate_allocator_->NotifyObserverInactive(this, !active);
+}
+
 void VideoSendStream::ConfigureProtection() {
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // Payload types without picture ID cannot determine that a stream is complete
   // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
   // a waste of bandwidth since FEC packets still have to be transmitted. Note
   // that this is not the case with FLEXFEC.
   if (enable_protection_nack &&
       !PayloadTypeSupportsSkippingFecPackets(
@@ skipping 101 matching lines @@
   return rtp_states;
 }
 
 void VideoSendStream::SignalNetworkState(NetworkState state) {
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                 : RtcpMode::kOff);
   }
 }
 
-int VideoSendStream::GetPaddingNeededBps() const {
-  return vie_encoder_.GetPaddingNeededBps();
-}
-
 void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                        uint8_t fraction_loss,
                                        int64_t rtt) {
   payload_router_.SetTargetSendBitrate(bitrate_bps);
   vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
 }
 
 int VideoSendStream::ProtectionRequest(const FecProtectionParams* delta_params,
                                        const FecProtectionParams* key_params,
                                        uint32_t* sent_video_rate_bps,
@@ skipping 12 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc