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Account for extra capacity rtx packet might need

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When calculating maximum allowed size for a media packet.
In particular take in account that rtx packet might need to
include mid and repaired-rsid extensions when media packet can omit them.

Bug: webrtc:11031
Change-Id: I3e7bc36437c23e0330316588d2a46978407c8c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206060
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33184}
This commit is contained in:
Danil Chapovalov 2021-02-05 19:30:16 +01:00 committed by Commit Bot
parent d42413a4b4
commit 9554a7b641
3 changed files with 31 additions and 16 deletions
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2
call/rtp_video_sender_unittest.cc
View File

@ -911,7 +911,7 @@ TEST(RtpVideoSenderTest, OverheadIsSubtractedFromTargetBitrate) {
constexpr uint32_t kTransportPacketOverheadBytes = 40;
constexpr uint32_t kOverheadPerPacketBytes =
kRtpHeaderSizeBytes + kTransportPacketOverheadBytes;
RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {});
RtpVideoSenderTestFixture test({kSsrc1}, {}, kPayloadType, {});
test.router()->OnTransportOverheadChanged(kTransportPacketOverheadBytes);
test.router()->SetActive(true);

21
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -172,8 +172,6 @@ RTPSender::RTPSender(const RtpRtcpInterface::Configuration& config,
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(config.extmap_allow_mixed),
max_media_packet_header_(kRtpHeaderSize),
max_padding_fec_packet_header_(kRtpHeaderSize),
// RTP variables
sequence_number_forced_(false),
always_send_mid_and_rid_(config.always_send_mid_and_rid),
@ -187,6 +185,7 @@ RTPSender::RTPSender(const RtpRtcpInterface::Configuration& config,
rtx_(kRtxOff),
supports_bwe_extension_(false),
retransmission_rate_limiter_(config.retransmission_rate_limiter) {
UpdateHeaderSizes();
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
@ -360,7 +359,11 @@ void RTPSender::OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number) {
void RTPSender::OnReceivedAckOnRtxSsrc(
int64_t extended_highest_sequence_number) {
MutexLock lock(&send_mutex_);
bool update_required = !rtx_ssrc_has_acked_;
rtx_ssrc_has_acked_ = true;
if (update_required) {
UpdateHeaderSizes();
}
}
void RTPSender::OnReceivedNack(
@ -874,10 +877,12 @@ void RTPSender::UpdateHeaderSizes() {
rtp_header_extension_map_);
// RtpStreamId and Mid are treated specially in that we check if they
// currently are being sent. RepairedRtpStreamId is still ignored since we
// assume RTX will not make up large enough bitrate to treat overhead
// differently.
const bool send_mid_rid = always_send_mid_and_rid_ || !ssrc_has_acked_;
// currently are being sent. RepairedRtpStreamId is ignored because it is sent
// instead of RtpStreamId on rtx packets and require the same size.
const bool send_mid_rid_on_rtx =
rtx_ssrc_.has_value() && !rtx_ssrc_has_acked_;
const bool send_mid_rid =
always_send_mid_and_rid_ || !ssrc_has_acked_ || send_mid_rid_on_rtx;
std::vector<RtpExtensionSize> non_volatile_extensions;
for (auto& extension :
audio_configured_ ? AudioExtensionSizes() : VideoExtensionSizes()) {
@ -901,5 +906,9 @@ void RTPSender::UpdateHeaderSizes() {
max_media_packet_header_ =
rtp_header_length + RtpHeaderExtensionSize(non_volatile_extensions,
rtp_header_extension_map_);
// Reserve extra bytes if packet might be resent in an rtx packet.
if (rtx_ssrc_.has_value()) {
max_media_packet_header_ += kRtxHeaderSize;
}
}
} // namespace webrtc

24
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -34,6 +34,7 @@
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/rtp_rtcp/source/video_fec_generator.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/logging.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/task_utils/to_queued_task.h"
@ -415,7 +416,12 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
std::unique_ptr<RtpPacketToSend> SendGenericPacket() {
const int64_t kCaptureTimeMs = clock_->TimeInMilliseconds();
return SendPacket(kCaptureTimeMs, sizeof(kPayloadData));
// Use maximum allowed size to catch corner cases when packet is dropped
// because of lack of capacity for the media packet, or for an rtx packet
// containing the media packet.
return SendPacket(kCaptureTimeMs,
/*payload_length=*/rtp_sender()->MaxRtpPacketSize() -
rtp_sender()->ExpectedPerPacketOverhead());
}
size_t GenerateAndSendPadding(size_t target_size_bytes) {
@ -596,6 +602,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
TEST_P(RtpSenderTestWithoutPacer,
TransportFeedbackObserverGetsCorrectByteCount) {
constexpr size_t kRtpOverheadBytesPerPacket = 12 + 8;
constexpr size_t kPayloadSize = 1400;
RtpRtcpInterface::Configuration config;
config.clock = clock_;
@ -612,10 +619,9 @@ TEST_P(RtpSenderTestWithoutPacer,
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
const size_t expected_bytes =
GetParam().with_overhead
? sizeof(kPayloadData) + kRtpOverheadBytesPerPacket
: sizeof(kPayloadData);
const size_t expected_bytes = GetParam().with_overhead
? kPayloadSize + kRtpOverheadBytesPerPacket
: kPayloadSize;
EXPECT_CALL(feedback_observer_,
OnAddPacket(AllOf(
@ -629,7 +635,7 @@ TEST_P(RtpSenderTestWithoutPacer,
.Times(1);
EXPECT_EQ(rtp_sender()->ExpectedPerPacketOverhead(),
kRtpOverheadBytesPerPacket);
SendGenericPacket();
SendPacket(clock_->TimeInMilliseconds(), kPayloadSize);
}
TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
@ -1968,12 +1974,12 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacksUlpfec) {
}
TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
// XXX const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;
const size_t kPayloadSize = 1400;
rtp_sender()->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
SendGenericPacket();
SendPacket(clock_->TimeInMilliseconds(), kPayloadSize);
// Will send 2 full-size padding packets.
GenerateAndSendPadding(1);
GenerateAndSendPadding(1);
@ -1984,7 +1990,7 @@ TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
// Payload
EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(kPayloadData));
EXPECT_EQ(rtp_stats.transmitted.payload_bytes, kPayloadSize);
EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
EXPECT_EQ(rtx_stats.transmitted.payload_bytes, 0u);