Remove now unused code in RtpHeaderExtensionMap

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 74 matching lines @@
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       payload_type_(-1),
       payload_type_map_(),
       rtp_header_extension_map_(),
-      video_rotation_active_(false),
-      playout_delay_active_(false),
       packet_history_(clock),
       // Statistics
       rtp_stats_callback_(nullptr),
       total_bitrate_sent_(kBitrateStatisticsWindowMs,
                           RateStatistics::kBpsScale),
       nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
       send_packet_observer_(send_packet_observer),
@@ skipping 71 matching lines @@
 uint32_t RTPSender::NackOverheadRate() const {
   rtc::CritScope cs(&statistics_crit_);
   return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
                                               uint8_t id) {
   rtc::CritScope lock(&send_critsect_);
   switch (type) {
     case kRtpExtensionVideoRotation:
-      video_rotation_active_ = false;
-      return rtp_header_extension_map_.RegisterInactive(type, id);
     case kRtpExtensionPlayoutDelay:
-      playout_delay_active_ = false;
-      return rtp_header_extension_map_.RegisterInactive(type, id);
     case kRtpExtensionTransmissionTimeOffset:
     case kRtpExtensionAbsoluteSendTime:
     case kRtpExtensionAudioLevel:
     case kRtpExtensionTransportSequenceNumber:
       return rtp_header_extension_map_.Register(type, id);
     case kRtpExtensionNone:
     case kRtpExtensionNumberOfExtensions:
       LOG(LS_ERROR) << "Invalid RTP extension type for registration";
       return -1;
   }
@@ skipping 168 matching lines @@
   SetSendPayloadType(payload_type);
   RtpUtility::Payload* payload = it->second;
   assert(payload);
   if (!payload->audio && !audio_configured_) {
     video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
     *video_type = payload->typeSpecific.Video.videoCodecType;
   }
   return 0;
 }
 
-bool RTPSender::ActivateCVORtpHeaderExtension() {
-  if (!video_rotation_active_) {
-    rtc::CritScope lock(&send_critsect_);
-    if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) {
-      video_rotation_active_ = true;
-    }
-  }
-  return video_rotation_active_;
-}
-
 bool 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_header,
                                  uint32_t* transport_frame_id_out) {
   uint32_t ssrc;
@@ skipping 32 matching lines @@
     assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
 
     if (frame_type == kEmptyFrame)
       return true;
 
     if (rtp_header) {
       playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
                                           sequence_number);
     }
 
-    // Update the active/inactive status of playout delay extension based
-    // on what the oracle indicates.
-    {
-      rtc::CritScope lock(&send_critsect_);
-      bool send_playout_delay = playout_delay_oracle_.send_playout_delay();
-      if (playout_delay_active_ != send_playout_delay) {
-        playout_delay_active_ = send_playout_delay;
-        rtp_header_extension_map_.SetActive(kRtpExtensionPlayoutDelay,
-                                            playout_delay_active_);
-      }
-    }
-
     result = video_->SendVideo(video_type, frame_type, payload_type,
                                rtp_timestamp, capture_time_ms, payload_data,
                                payload_size, fragmentation, rtp_header);
   }
 
   rtc::CritScope cs(&statistics_crit_);
   // Note: This is currently only counting for video.
   if (frame_type == kVideoFrameKey) {
     ++frame_counts_.key_frames;
   } else if (frame_type == kVideoFrameDelta) {
@@ skipping 555 matching lines @@
   // Remember marker bit to determine if padding can be inserted with
   // sequence number following |packet|.
   last_packet_marker_bit_ = packet->Marker();
   // Save timestamps to generate timestamp field and extensions for the padding.
   last_rtp_timestamp_ = packet->Timestamp();
   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
   capture_time_ms_ = packet->capture_time_ms();
   return true;
 }
 
-bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
-                                            const uint8_t* rtp_packet,
-                                            size_t rtp_packet_length,
-                                            const RTPHeader& rtp_header,
-                                            size_t* position) const {
-  // Get length until start of header extension block.
-  int extension_block_pos =
-      rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(type);
-  if (extension_block_pos < 0) {
-    LOG(LS_WARNING) << "Failed to find extension position for " << type
-                    << " as it is not registered.";
-    return false;
-  }
-
-  HeaderExtension header_extension(type);
-
-  size_t extension_pos =
-      kRtpHeaderLength + rtp_header.numCSRCs * sizeof(uint32_t);
-  size_t block_pos = extension_pos + extension_block_pos;
-  if (rtp_packet_length < block_pos + header_extension.length ||
-      rtp_header.headerLength < block_pos + header_extension.length) {
-    LOG(LS_WARNING) << "Failed to find extension position for " << type
-                    << " as the length is invalid.";
-    return false;
-  }
-
-  // Verify that header contains extension.
-  if (!(rtp_packet[extension_pos] == 0xBE &&
-        rtp_packet[extension_pos + 1] == 0xDE)) {
-    LOG(LS_WARNING) << "Failed to find extension position for " << type
-                    << "as hdr extension not found.";
-    return false;
-  }
-
-  *position = block_pos;
-  return true;
-}
-
 bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
                                               int* packet_id) const {
   RTC_DCHECK(packet);
   RTC_DCHECK(packet_id);
   rtc::CritScope lock(&send_critsect_);
   if (!rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId))
     return false;
 
   if (!transport_sequence_number_allocator_)
     return false;
@@ skipping 235 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc