Initialize VideoSendStream members in constructor.

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.
  */
 
 #include "webrtc/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/video/call_stats.h"
-#include "webrtc/video/encoder_state_feedback.h"
 #include "webrtc/video/video_capture_input.h"
-#include "webrtc/video/vie_channel.h"
-#include "webrtc/video/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class PacedSender;
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
 
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
@@ skipping 97 matching lines @@
       suspended_ssrcs_(suspended_ssrcs),
       module_process_thread_(module_process_thread),
       call_stats_(call_stats),
       congestion_controller_(congestion_controller),
       overuse_detector_(
           Clock::GetRealTimeClock(),
           GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
           this,
           config.post_encode_callback,
           &stats_proxy_),
-      encoder_feedback_(new EncoderStateFeedback()),
-      use_config_bitrate_(true) {
+      vie_encoder_(num_cpu_cores,
+                   module_process_thread_,
+                   &stats_proxy_,
+                   config.pre_encode_callback,
+                   &overuse_detector_,
+                   congestion_controller_->pacer(),
+                   &payload_router_,
+                   bitrate_allocator),
+      vcm_(vie_encoder_.vcm()),
+      vie_channel_(config.send_transport,
+                   module_process_thread_,
+                   &payload_router_,
+                   nullptr,
+                   encoder_feedback_.GetRtcpIntraFrameObserver(),
+                   congestion_controller_->GetBitrateController()
+                       ->CreateRtcpBandwidthObserver(),
+                   GetTransportFeedbackObserver(),

[stefan-webrtc, 2016/02/11 10:15:40]

I'd prefer if this is done as in audio_send_stream.cc instead. There we always
pass in the transport feedback observer.

[pbos-webrtc, 2016/02/11 11:23:33]

Done.

+                   congestion_controller_->GetRemoteBitrateEstimator(false),

[stefan-webrtc, 2016/02/11 10:15:40]

I think we should be able to pass in nullptr here.

[pbos-webrtc, 2016/02/11 11:23:34]

Done.

+                   call_stats_->rtcp_rtt_stats(),
+                   congestion_controller_->pacer(),
+                   congestion_controller_->packet_router(),
+                   config_.rtp.ssrcs.size(),
+                   true),
+      input_(&vie_encoder_,
+             config_.local_renderer,
+             &stats_proxy_,
+             &overuse_detector_) {
   LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
 
-  // Set up Call-wide sequence numbers, if configured for this send stream.
-  TransportFeedbackObserver* transport_feedback_observer = nullptr;
-  for (const RtpExtension& extension : config.rtp.extensions) {
-    if (extension.name == RtpExtension::kTransportSequenceNumber) {
-      transport_feedback_observer =
-          congestion_controller_->GetTransportFeedbackObserver();
-      break;
-    }
-  }
+  RTC_CHECK(vie_encoder_.Init());
+  RTC_CHECK(vie_channel_.Init() == 0);
 
-  const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
+  vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
 
-  vie_encoder_.reset(new ViEEncoder(
-      num_cpu_cores, module_process_thread_, &stats_proxy_,
-      config.pre_encode_callback, &overuse_detector_,
-      congestion_controller_->pacer(), &payload_router_, bitrate_allocator));
-  vcm_ = vie_encoder_->vcm();
-  RTC_CHECK(vie_encoder_->Init());
+  call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
-  vie_channel_.reset(new ViEChannel(
-      config.send_transport, module_process_thread_, &payload_router_, nullptr,
-      encoder_feedback_->GetRtcpIntraFrameObserver(),
-      congestion_controller_->GetBitrateController()
-          ->CreateRtcpBandwidthObserver(),
-      transport_feedback_observer,
-      congestion_controller_->GetRemoteBitrateEstimator(false),
-      call_stats_->rtcp_rtt_stats(), congestion_controller_->pacer(),
-      congestion_controller_->packet_router(), ssrcs.size(), true));
-  RTC_CHECK(vie_channel_->Init() == 0);
-
-  vcm_->RegisterProtectionCallback(vie_channel_->vcm_protection_callback());
-
-  call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());
-
-  std::vector<uint32_t> first_ssrc(1, ssrcs[0]);
-  vie_encoder_->SetSsrcs(first_ssrc);
+  vie_encoder_.SetSsrcs(std::vector<uint32_t>(1, config_.rtp.ssrcs[0]));
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
-      RTC_CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id));
+      RTC_CHECK_EQ(0, vie_channel_.SetSendTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
-      RTC_CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id));
+      RTC_CHECK_EQ(0, vie_channel_.SetSendAbsoluteSendTimeStatus(true, id));
     } else if (extension == RtpExtension::kVideoRotation) {
-      RTC_CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id));
+      RTC_CHECK_EQ(0, vie_channel_.SetSendVideoRotationStatus(true, id));
     } else if (extension == RtpExtension::kTransportSequenceNumber) {
-      RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id));
+      RTC_CHECK_EQ(0, vie_channel_.SetSendTransportSequenceNumber(true, id));
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 
   congestion_controller_->SetChannelRembStatus(true, false,
-                                               vie_channel_->rtp_rtcp());
+                                               vie_channel_.rtp_rtcp());
 
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
-  vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec,
+  vie_channel_.SetProtectionMode(enable_protection_nack, enable_protection_fec,
                                   config_.rtp.fec.red_payload_type,
                                   config_.rtp.fec.ulpfec_payload_type);
-  vie_encoder_->SetProtectionMethod(enable_protection_nack,
+  vie_encoder_.SetProtectionMethod(enable_protection_nack,
                                     enable_protection_fec);
 
   ConfigureSsrcs();
 
-  vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
-
-  input_.reset(new internal::VideoCaptureInput(
-      vie_encoder_.get(), config_.local_renderer, &stats_proxy_,
-      &overuse_detector_));
+  vie_channel_.SetRTCPCName(config_.rtp.c_name.c_str());
 
   // 28 to match packet overhead in ModuleRtpRtcpImpl.
   RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28));
-  vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
+  vie_channel_.SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
 
   RTC_DCHECK(config.encoder_settings.encoder != nullptr);
   RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
   RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
-  RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder(
+  RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
                       config.encoder_settings.encoder,
                       config.encoder_settings.payload_type,
                       config.encoder_settings.internal_source));
 
   RTC_CHECK(ReconfigureVideoEncoder(encoder_config));
 
-  vie_channel_->RegisterSendSideDelayObserver(&stats_proxy_);
+  vie_channel_.RegisterSendSideDelayObserver(&stats_proxy_);
 
   if (config_.post_encode_callback)
-    vie_encoder_->RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
+    vie_encoder_.RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
 
   if (config_.suspend_below_min_bitrate)
-    vie_encoder_->SuspendBelowMinBitrate();
+    vie_encoder_.SuspendBelowMinBitrate();
 
-  congestion_controller_->AddEncoder(vie_encoder_.get());
-  encoder_feedback_->AddEncoder(ssrcs, vie_encoder_.get());
+  congestion_controller_->AddEncoder(&vie_encoder_);
+  encoder_feedback_.AddEncoder(config_.rtp.ssrcs, &vie_encoder_);
 
-  vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
-  vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
-  vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
-  vie_channel_->RegisterSendBitrateObserver(&stats_proxy_);
-  vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_);
+  vie_channel_.RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
+  vie_channel_.RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
+  vie_channel_.RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
+  vie_channel_.RegisterSendBitrateObserver(&stats_proxy_);
+  vie_channel_.RegisterSendFrameCountObserver(&stats_proxy_);
 
   module_process_thread_->RegisterModule(&overuse_detector_);
 }
 
 VideoSendStream::~VideoSendStream() {
   LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
   module_process_thread_->DeRegisterModule(&overuse_detector_);
   // Remove vcm_protection_callback (part of vie_channel_) before destroying
   // ViEChannel. vcm_ is owned by ViEEncoder and the registered callback does
   // not outlive it.
   vcm_->RegisterProtectionCallback(nullptr);
-  vie_channel_->RegisterSendFrameCountObserver(nullptr);
-  vie_channel_->RegisterSendBitrateObserver(nullptr);
-  vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
-  vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr);
-  vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr);
+  vie_channel_.RegisterSendFrameCountObserver(nullptr);
+  vie_channel_.RegisterSendBitrateObserver(nullptr);
+  vie_channel_.RegisterRtcpPacketTypeCounterObserver(nullptr);
+  vie_channel_.RegisterSendChannelRtpStatisticsCallback(nullptr);
+  vie_channel_.RegisterSendChannelRtcpStatisticsCallback(nullptr);
 
-  // Remove capture input (thread) so that it's not running after the current
-  // channel is deleted.
-  input_.reset();
-
-  vie_encoder_->DeRegisterExternalEncoder(
+  vie_encoder_.DeRegisterExternalEncoder(
       config_.encoder_settings.payload_type);
 
-  call_stats_->DeregisterStatsObserver(vie_channel_->GetStatsObserver());
+  call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
   congestion_controller_->SetChannelRembStatus(false, false,
-                                               vie_channel_->rtp_rtcp());
+                                               vie_channel_.rtp_rtcp());
 
   // Remove the feedback, stop all encoding threads and processing. This must be
   // done before deleting the channel.
-  congestion_controller_->RemoveEncoder(vie_encoder_.get());
-  encoder_feedback_->RemoveEncoder(vie_encoder_.get());
+  congestion_controller_->RemoveEncoder(&vie_encoder_);
+  encoder_feedback_.RemoveEncoder(&vie_encoder_);
 
-  uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
+  uint32_t remote_ssrc = vie_channel_.GetRemoteSSRC();
   congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
       remote_ssrc);
 }
 
 VideoCaptureInput* VideoSendStream::Input() {
-  return input_.get();
+  return &input_;
 }
 
 void VideoSendStream::Start() {
   transport_adapter_.Enable();
-  vie_encoder_->Pause();
-  if (vie_channel_->StartSend() == 0) {
+  vie_encoder_.Pause();
+  if (vie_channel_.StartSend() == 0) {
     // Was not already started, trigger a keyframe.
-    vie_encoder_->SendKeyFrame();
+    vie_encoder_.SendKeyFrame();
   }
-  vie_encoder_->Restart();
-  vie_channel_->StartReceive();
+  vie_encoder_.Restart();
+  vie_channel_.StartReceive();
 }
 
 void VideoSendStream::Stop() {
   // TODO(pbos): Make sure the encoder stops here.
-  vie_channel_->StopSend();
-  vie_channel_->StopReceive();
+  vie_channel_.StopSend();
+  vie_channel_.StopReceive();
   transport_adapter_.Disable();
 }
 
 bool VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   const std::vector<VideoStream>& streams = config.streams;
   RTC_DCHECK(!streams.empty());
   RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
@@ skipping 129 matching lines @@
 
   // Clear stats for disabled layers.
   for (size_t i = video_codec.numberOfSimulcastStreams;
        i < config_.rtp.ssrcs.size(); ++i) {
     stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
   }
 
   stats_proxy_.SetContentType(config.content_type);
 
   RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
-  vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
+  vie_encoder_.SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
 
   encoder_config_ = config;
-  use_config_bitrate_ = false;
   return true;
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
+  return vie_channel_.ReceivedRTCPPacket(packet, length) == 0;
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::OveruseDetected() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
 }
 
 void VideoSendStream::NormalUsage() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
 }
 
 void VideoSendStream::ConfigureSsrcs() {
-  vie_channel_->SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
+  vie_channel_.SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    vie_channel_->SetSSRC(ssrc, kViEStreamTypeNormal,
+    vie_channel_.SetSSRC(ssrc, kViEStreamTypeNormal,
                           static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
+      vie_channel_.SetRtpStateForSsrc(ssrc, it->second);
   }
 
   if (config_.rtp.rtx.ssrcs.empty()) {
     return;
   }
 
   // Set up RTX.
   RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
-    vie_channel_->SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
+    vie_channel_.SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx,
                           static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
+      vie_channel_.SetRtpStateForSsrc(ssrc, it->second);
   }
 
   RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
-  vie_channel_->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
+  vie_channel_.SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
                                       config_.encoder_settings.payload_type);
   if (config_.rtp.fec.red_payload_type != -1 &&
       config_.rtp.fec.red_rtx_payload_type != -1) {
-    vie_channel_->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
+    vie_channel_.SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type,
                                         config_.rtp.fec.red_payload_type);
   }
 }
 
 std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
   std::map<uint32_t, RtpState> rtp_states;
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
+    rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
-    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
+    rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
   }
 
   return rtp_states;
 }
 
 void VideoSendStream::SignalNetworkState(NetworkState state) {
   // When network goes up, enable RTCP status before setting transmission state.
   // When it goes down, disable RTCP afterwards. This ensures that any packets
   // sent due to the network state changed will not be dropped.
   if (state == kNetworkUp)
-    vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
-  vie_encoder_->SetNetworkTransmissionState(state == kNetworkUp);
+    vie_channel_.SetRTCPMode(config_.rtp.rtcp_mode);
+  vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
   if (state == kNetworkDown)
-    vie_channel_->SetRTCPMode(RtcpMode::kOff);
+    vie_channel_.SetRTCPMode(RtcpMode::kOff);
 }
 
 int64_t VideoSendStream::GetRtt() const {
   webrtc::RtcpStatistics rtcp_stats;
   uint16_t frac_lost;
   uint32_t cumulative_lost;
   uint32_t extended_max_sequence_number;
   uint32_t jitter;
   int64_t rtt_ms;
-  if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
+  if (vie_channel_.GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
                                           &extended_max_sequence_number,
                                           &jitter, &rtt_ms) == 0) {
     return rtt_ms;
   }
   return -1;
 }
 
 int VideoSendStream::GetPaddingNeededBps() const {
-  return vie_encoder_->GetPaddingNeededBps();
+  return vie_encoder_.GetPaddingNeededBps();
 }
 
 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
   static const int kEncoderMinBitrate = 30;
   if (video_codec.maxBitrate == 0) {
     // Unset max bitrate -> cap to one bit per pixel.
     video_codec.maxBitrate =
         (video_codec.width * video_codec.height * video_codec.maxFramerate) /
         1000;
   }
 
   if (video_codec.minBitrate < kEncoderMinBitrate)
     video_codec.minBitrate = kEncoderMinBitrate;
   if (video_codec.maxBitrate < kEncoderMinBitrate)
     video_codec.maxBitrate = kEncoderMinBitrate;
 
   // Stop the media flow while reconfiguring.
-  vie_encoder_->Pause();
+  vie_encoder_.Pause();
 
-  if (vie_encoder_->SetEncoder(video_codec) != 0) {
+  if (vie_encoder_.SetEncoder(video_codec) != 0) {
     LOG(LS_ERROR) << "Failed to set encoder.";
     return false;
   }
 
-  if (vie_channel_->SetSendCodec(video_codec, false) != 0) {
+  if (vie_channel_.SetSendCodec(video_codec, false) != 0) {
     LOG(LS_ERROR) << "Failed to set send codec.";
     return false;
   }
 
   // Not all configured SSRCs have to be utilized (simulcast senders don't have
   // to send on all SSRCs at once etc.)
   std::vector<uint32_t> used_ssrcs = config_.rtp.ssrcs;
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
-  vie_encoder_->SetSsrcs(used_ssrcs);
+  vie_encoder_.SetSsrcs(used_ssrcs);
 
   // Restart the media flow
-  vie_encoder_->Restart();
+  vie_encoder_.Restart();
 
   return true;
 }
 
+TransportFeedbackObserver* VideoSendStream::GetTransportFeedbackObserver()
+    const {
+  // Set up Call-wide sequence numbers, if configured for this send stream.
+  for (const RtpExtension& extension : config_.rtp.extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      return congestion_controller_->GetTransportFeedbackObserver();
+    }
+  }
+  return nullptr;
+}
+
 }  // namespace internal
 }  // namespace webrtc