Remove unused deprecated code in RTPSender.
Bug: webrtc:11340, webrtc:12470
Change-Id: I01a6262cfeb33d1900f8f3cd93cceee2ff73a8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227643
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34646}
diff --git a/modules/rtp_rtcp/source/packet_sequencer.cc b/modules/rtp_rtcp/source/packet_sequencer.cc
index 39d40eb..a4e7b46 100644
--- a/modules/rtp_rtcp/source/packet_sequencer.cc
+++ b/modules/rtp_rtcp/source/packet_sequencer.cc
@@ -39,9 +39,10 @@
last_timestamp_time_ms_(0),
last_packet_marker_bit_(false) {
Random random(clock_->TimeInMicroseconds());
- // TODO(bugs.webrtc.org/11340): Check if we can allow the full range of
- // [0, 2^16[ to be used instead.
- // Random start, 16 bits. Can't be 0.
+ // Random start, 16 bits. Upper half of range is avoided in order to prevent
+ // wraparound issues during startup. Sequence number 0 is avoided for
+ // historical reasons, presumably to avoid debugability or test usage
+ // conflicts.
constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff; // 2^15 - 1.
media_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
rtx_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index de79957..a0ea1e8 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -16,7 +16,6 @@
#include <string>
#include <utility>
-#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
@@ -190,20 +189,6 @@
RTC_DCHECK(packet_history_);
}
-RTPSender::RTPSender(const RtpRtcpInterface::Configuration& config,
- RtpPacketHistory* packet_history,
- RtpPacketSender* packet_sender)
- : RTPSender(config,
- packet_history,
- packet_sender,
- new PacketSequencer(
- config.local_media_ssrc,
- config.rtx_send_ssrc,
- /*require_marker_before_media_padding_=*/!config.audio,
- config.clock)) {
- owned_sequencer_ = absl::WrapUnique(sequencer_);
-}
-
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 6662c33..6d5826e 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -49,12 +49,6 @@
RtpPacketSender* packet_sender,
PacketSequencer* packet_sequencer);
- // TODO(bugs.webrtc.org/11340): Remove when downstream usage is gone.
- RTPSender(const RtpRtcpInterface::Configuration& config,
- RtpPacketHistory* packet_history,
- RtpPacketSender* packet_sender)
- ABSL_DEPRECATED("bugs.webrtc.org/11340");
-
RTPSender() = delete;
RTPSender(const RTPSender&) = delete;
RTPSender& operator=(const RTPSender&) = delete;
@@ -100,10 +94,7 @@
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes,
bool media_has_been_sent,
- // TODO(bugs.webrtc.org/11340): Remove default value when downstream usage
- // is fixed.
- bool can_send_padding_on_media_ssrc = false)
- RTC_LOCKS_EXCLUDED(send_mutex_);
+ bool can_send_padding_on_media_ssrc) RTC_LOCKS_EXCLUDED(send_mutex_);
// NACK.
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
@@ -220,8 +211,6 @@
// RTP variables
uint32_t timestamp_offset_ RTC_GUARDED_BY(send_mutex_);
- // TODO(bugs.webrtc.org/11340): Remove when downstream usage is gone.
- std::unique_ptr<PacketSequencer> owned_sequencer_ RTC_GUARDED_BY(send_mutex_);
PacketSequencer* const sequencer_ RTC_GUARDED_BY(send_mutex_);
// RID value to send in the RID or RepairedRID header extension.
std::string rid_ RTC_GUARDED_BY(send_mutex_);
