Removed vie_defines.h

webrtc/video_engine/vie_channel.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 16 matching lines @@
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/thread_wrapper.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/report_block_stats.h"
-#include "webrtc/video_engine/vie_defines.h"
 
 namespace webrtc {
 
 const int kMaxDecodeWaitTimeMs = 50;
 static const int kMaxTargetDelayMs = 10000;
 static const float kMaxIncompleteTimeMultiplier = 3.5f;
+const int kMinSendSidePacketHistorySize = 600;
+const int kMaxPacketAgeToNack = 450;
+const int kMaxNackListSize = 250;
 
 // Helper class receiving statistics callbacks.
 class ChannelStatsObserver : public CallStatsObserver {
  public:
   explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
   virtual ~ChannelStatsObserver() {}
 
   // Implements StatsObserver.
   virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
     owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
@@ skipping 48 matching lines @@
       vie_sync_(vcm_),
       stats_observer_(new ChannelStatsObserver(this)),
       receive_stats_callback_(nullptr),
       incoming_video_stream_(nullptr),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
       transport_feedback_observer_(transport_feedback_observer),
-      nack_history_size_sender_(kSendSidePacketHistorySize),
+      nack_history_size_sender_(kMinSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       report_block_stats_sender_(new ReportBlockStats()),
       time_of_first_rtt_ms_(-1),
       rtt_sum_ms_(0),
       last_rtt_ms_(0),
       num_rtts_(0),
       rtp_rtcp_modules_(
           CreateRtpRtcpModules(!sender,
                                vie_receiver_.GetReceiveStatistics(),
                                transport,
                                sender ? intra_frame_observer_ : nullptr,
                                sender ? bandwidth_observer_.get() : nullptr,
                                transport_feedback_observer_,
                                rtt_stats_,
                                &rtcp_packet_type_counter_observer_,
                                remote_bitrate_estimator,
                                paced_sender_,
                                packet_router_,
                                &send_bitrate_observer_,
                                &send_frame_count_observer_,
                                &send_side_delay_observer_,
                                max_rtp_streams)),
       num_active_rtp_rtcp_modules_(1) {
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]);
   vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
 }
 
 int32_t ViEChannel::Init() {
+  const int kDefaultRenderDelayMs = 10;

[pbos-webrtc, 2015/10/29 15:39:40]

static

[mflodman, 2015/11/06 11:38:36]

Done.

   module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
 
   // RTP/RTCP initialization.
   module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]);
 
   rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
   if (paced_sender_) {
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
       rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
   }
   packet_router_->AddRtpModule(rtp_rtcp_modules_[0]);
   if (sender_) {
     std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]);
     send_payload_router_->SetSendingRtpModules(send_rtp_modules);
     RTC_DCHECK(!send_payload_router_->active());
   }
   if (vcm_->RegisterReceiveCallback(this) != 0) {
     return -1;
   }
   vcm_->RegisterFrameTypeCallback(this);
   vcm_->RegisterReceiveStatisticsCallback(this);
   vcm_->RegisterDecoderTimingCallback(this);
-  vcm_->SetRenderDelay(kViEDefaultRenderDelayMs);
+  vcm_->SetRenderDelay(kDefaultRenderDelayMs);
 
   module_process_thread_->RegisterModule(vcm_);
   module_process_thread_->RegisterModule(&vie_sync_);
 
   return 0;
 }
 
 ViEChannel::~ViEChannel() {
   UpdateHistograms();
   // Make sure we don't get more callbacks from the RTP module.
@@ skipping 380 matching lines @@
   return false;
 }
 
 int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
   if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
     LOG(LS_ERROR) << "Invalid send buffer value.";
     return -1;
   }
   if (target_delay_ms == 0) {
     // Real-time mode.
-    nack_history_size_sender_ = kSendSidePacketHistorySize;
+    nack_history_size_sender_ = kMinSendSidePacketHistorySize;
   } else {
     nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
     // Don't allow a number lower than the default value.
-    if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
-      nack_history_size_sender_ = kSendSidePacketHistorySize;
+    if (nack_history_size_sender_ < kMinSendSidePacketHistorySize) {
+      nack_history_size_sender_ = kMinSendSidePacketHistorySize;
     }
   }
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
   return 0;
 }
 
 int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
   if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
     LOG(LS_ERROR) << "Invalid receive buffer delay value.";
@@ skipping 663 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc