Moves RtpSequenceNumberMap from RtpSenderVideo to RtpSenderEgress.
Bug: webrtc:11340
Change-Id: Icd9032e3589324cb9ee7b699b38a35e733081e55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168192
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30481}
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 99fb822..532e069 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -41,7 +41,6 @@
namespace {
constexpr size_t kRedForFecHeaderLength = 1;
-constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
constexpr int64_t kMaxUnretransmittableFrameIntervalMs = 33 * 4;
// This is experimental field trial to exclude transport sequence number from
@@ -253,7 +252,6 @@
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
- bool need_rtp_packet_infos,
bool enable_retransmit_all_layers,
const WebRtcKeyValueConfig& field_trials)
: RTPSenderVideo([&] {
@@ -263,7 +261,6 @@
config.flexfec_sender = flexfec_sender;
config.frame_encryptor = frame_encryptor;
config.require_frame_encryption = require_frame_encryption;
- config.need_rtp_packet_infos = need_rtp_packet_infos;
config.enable_retransmit_all_layers = enable_retransmit_all_layers;
config.field_trials = &field_trials;
return config;
@@ -280,10 +277,6 @@
transmit_color_space_next_frame_(false),
current_playout_delay_{-1, -1},
playout_delay_pending_(false),
- rtp_sequence_number_map_(config.need_rtp_packet_infos
- ? std::make_unique<RtpSequenceNumberMap>(
- kRtpSequenceNumberMapMaxEntries)
- : nullptr),
red_payload_type_(config.red_payload_type),
ulpfec_payload_type_(config.ulpfec_payload_type),
flexfec_sender_(config.flexfec_sender),
@@ -729,7 +722,6 @@
if (num_packets == 0)
return false;
- uint16_t first_sequence_number;
bool first_frame = first_frame_sent_();
std::vector<std::unique_ptr<RtpPacketToSend>> rtp_packets;
for (size_t i = 0; i < num_packets; ++i) {
@@ -753,16 +745,14 @@
expected_payload_capacity = limits.max_payload_len;
}
+ packet->set_first_packet_of_frame(i == 0);
+
if (!packetizer->NextPacket(packet.get()))
return false;
RTC_DCHECK_LE(packet->payload_size(), expected_payload_capacity);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;
- if (rtp_sequence_number_map_ && i == 0) {
- first_sequence_number = packet->SequenceNumber();
- }
-
// No FEC protection for upper temporal layers, if used.
bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx;
@@ -802,13 +792,6 @@
}
}
- if (rtp_sequence_number_map_) {
- const uint32_t timestamp = rtp_timestamp - rtp_sender_->TimestampOffset();
- rtc::CritScope cs(&crit_);
- rtp_sequence_number_map_->InsertFrame(first_sequence_number, num_packets,
- timestamp);
- }
-
LogAndSendToNetwork(std::move(rtp_packets), unpacketized_payload_size);
TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
@@ -832,37 +815,6 @@
.value_or(0);
}
-std::vector<RtpSequenceNumberMap::Info> RTPSenderVideo::GetSentRtpPacketInfos(
- rtc::ArrayView<const uint16_t> sequence_numbers) const {
- RTC_DCHECK(!sequence_numbers.empty());
-
- std::vector<RtpSequenceNumberMap::Info> results;
- if (!rtp_sequence_number_map_) {
- return results;
- }
- results.reserve(sequence_numbers.size());
-
- {
- rtc::CritScope cs(&crit_);
- for (uint16_t sequence_number : sequence_numbers) {
- const absl::optional<RtpSequenceNumberMap::Info> info =
- rtp_sequence_number_map_->Get(sequence_number);
- if (!info) {
- // The empty vector will be returned. We can delay the clearing
- // of the vector until after we exit the critical section.
- break;
- }
- results.push_back(*info);
- }
- }
-
- if (results.size() != sequence_numbers.size()) {
- results.clear(); // Some sequence number was not found.
- }
-
- return results;
-}
-
bool RTPSenderVideo::AllowRetransmission(
uint8_t temporal_id,
int32_t retransmission_settings,
