From 8a61d0f2338505f4e2ceb6094125deb95faab299 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= <sprang@webrtc.org>
Date: Mon, 26 Aug 2019 17:12:21 +0200
Subject: [PATCH] Remove deprecated RTPSender ctor variant
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Bug: webrtc:10774
Change-Id: Ie0f7c04a7687aa442fd69f0cfe7c041acb0317ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150529
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28961}
---
 modules/rtp_rtcp/source/rtp_sender.cc | 72 ---------------------------
 modules/rtp_rtcp/source/rtp_sender.h  | 20 --------
 2 files changed, 92 deletions(-)

diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index b53d4655b8..ac0c8e8bac 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -161,78 +161,6 @@ RTPSender::RTPSender(const RtpRtcp::Configuration& config)
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
 }
 
-RTPSender::RTPSender(
-    bool audio,
-    Clock* clock,
-    Transport* transport,
-    RtpPacketSender* paced_sender,
-    absl::optional<uint32_t> flexfec_ssrc,
-    TransportSequenceNumberAllocator* sequence_number_allocator,
-    TransportFeedbackObserver* transport_feedback_observer,
-    BitrateStatisticsObserver* bitrate_callback,
-    SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log,
-    SendPacketObserver* send_packet_observer,
-    RateLimiter* retransmission_rate_limiter,
-    OverheadObserver* overhead_observer,
-    bool populate_network2_timestamp,
-    FrameEncryptorInterface* frame_encryptor,
-    bool require_frame_encryption,
-    bool extmap_allow_mixed,
-    const WebRtcKeyValueConfig& field_trials)
-    : clock_(clock),
-      random_(clock_->TimeInMicroseconds()),
-      audio_configured_(audio),
-      flexfec_ssrc_(flexfec_ssrc),
-      paced_sender_(paced_sender),
-      transport_sequence_number_allocator_(sequence_number_allocator),
-      transport_feedback_observer_(transport_feedback_observer),
-      transport_(transport),
-      sending_media_(true),  // Default to sending media.
-      force_part_of_allocation_(false),
-      max_packet_size_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
-      last_payload_type_(-1),
-      rtp_header_extension_map_(extmap_allow_mixed),
-      packet_history_(clock),
-      // Statistics
-      send_delays_(),
-      max_delay_it_(send_delays_.end()),
-      sum_delays_ms_(0),
-      total_packet_send_delay_ms_(0),
-      rtp_stats_callback_(nullptr),
-      total_bitrate_sent_(kBitrateStatisticsWindowMs,
-                          RateStatistics::kBpsScale),
-      nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
-      send_side_delay_observer_(send_side_delay_observer),
-      event_log_(event_log),
-      send_packet_observer_(send_packet_observer),
-      bitrate_callback_(bitrate_callback),
-      // RTP variables
-      sequence_number_forced_(false),
-      ssrc_has_acked_(false),
-      rtx_ssrc_has_acked_(false),
-      last_rtp_timestamp_(0),
-      capture_time_ms_(0),
-      last_timestamp_time_ms_(0),
-      media_has_been_sent_(false),
-      last_packet_marker_bit_(false),
-      csrcs_(),
-      rtx_(kRtxOff),
-      rtp_overhead_bytes_per_packet_(0),
-      supports_bwe_extension_(false),
-      retransmission_rate_limiter_(retransmission_rate_limiter),
-      overhead_observer_(overhead_observer),
-      populate_network2_timestamp_(populate_network2_timestamp),
-      send_side_bwe_with_overhead_(
-          field_trials.Lookup("WebRTC-SendSideBwe-WithOverhead")
-              .find("Enabled") == 0) {
-  // This random initialization is not intended to be cryptographic strong.
-  timestamp_offset_ = random_.Rand<uint32_t>();
-  // Random start, 16 bits. Can't be 0.
-  sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
-  sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
-}
-
 RTPSender::~RTPSender() {
   // TODO(tommi): Use a thread checker to ensure the object is created and
   // deleted on the same thread.  At the moment this isn't possible due to
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 3a2d86e92a..f384f75856 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -48,26 +48,6 @@ class RTPSender {
  public:
   explicit RTPSender(const RtpRtcp::Configuration& config);
 
-  // TODO(bugs.webrtc.org/10774): Remove once downstream projects are fixed.
-  RTPSender(bool audio,
-            Clock* clock,
-            Transport* transport,
-            RtpPacketSender* paced_sender,
-            absl::optional<uint32_t> flexfec_ssrc,
-            TransportSequenceNumberAllocator* sequence_number_allocator,
-            TransportFeedbackObserver* transport_feedback_callback,
-            BitrateStatisticsObserver* bitrate_callback,
-            SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log,
-            SendPacketObserver* send_packet_observer,
-            RateLimiter* nack_rate_limiter,
-            OverheadObserver* overhead_observer,
-            bool populate_network2_timestamp,
-            FrameEncryptorInterface* frame_encryptor,
-            bool require_frame_encryption,
-            bool extmap_allow_mixed,
-            const WebRtcKeyValueConfig& field_trials);
-
   ~RTPSender();
 
   void ProcessBitrate();