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Remove unused deprecated code in RTPSender.

Browse Source 
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}
This commit is contained in:
Erik Språng 2021-08-04 18:38:43 +02:00 committed by WebRTC LUCI CQ
parent 363b174567
commit 31c0cfdf7f
3 changed files with 5 additions and 30 deletions
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7
modules/rtp_rtcp/source/packet_sequencer.cc
View File

@ -39,9 +39,10 @@ PacketSequencer::PacketSequencer(uint32_t media_ssrc,
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);

15
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -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 @@ RTPSender::RTPSender(const RtpRtcpInterface::Configuration& config,
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

13
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -49,12 +49,6 @@ class RTPSender {
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 @@ class RTPSender {
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 @@ class RTPSender {
// 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_);