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Remove RTPSender::SetSendPayloadType

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Bug: None
Change-Id: Id99c9eda5e377de68c8bff053511534c66bd60a0
Reviewed-on: https://webrtc-review.googlesource.com/63801
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22559}
This commit is contained in:
Danil Chapovalov
2018-03-22 10:13:07 +01:00
committed by Commit Bot
parent 76ff6221ea
commit b3179c75ed
3 changed files with 40 additions and 50 deletions
Download Patch File Download Diff File Expand all files Collapse all files

17
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -123,7 +123,7 @@ RTPSender::RTPSender(
transport_(transport),
sending_media_(true), // Default to sending media.
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
payload_type_(-1),
last_payload_type_(-1),
payload_type_map_(),
rtp_header_extension_map_(),
packet_history_(clock),
@ -304,12 +304,6 @@ int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
return 0;
}
// TODO(nisse): Delete this method, only used internally and by test code.
void RTPSender::SetSendPayloadType(int8_t payload_type) {
rtc::CritScope lock(&send_critsect_);
payload_type_ = payload_type;
}
void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
RTC_DCHECK_GE(max_packet_size, 100);
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
@ -363,7 +357,7 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
RTC_LOG(LS_ERROR) << "Invalid payload_type " << payload_type << ".";
return -1;
}
if (payload_type_ == payload_type) {
if (last_payload_type_ == payload_type) {
if (!audio_configured_) {
*video_type = video_->VideoCodecType();
}
@ -376,7 +370,6 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
<< " not registered.";
return -1;
}
SetSendPayloadType(payload_type);
RtpUtility::Payload* payload = it->second;
RTC_DCHECK(payload);
if (payload->typeSpecific.is_video() && !audio_configured_) {
@ -531,7 +524,7 @@ size_t RTPSender::SendPadData(size_t bytes,
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
if (rtx_ == kRtxOff) {
if (payload_type_ == -1)
if (last_payload_type_ == -1)
break;
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
@ -550,7 +543,7 @@ size_t RTPSender::SendPadData(size_t bytes,
sequence_number = sequence_number_;
++sequence_number_;
payload_type = payload_type_;
payload_type = last_payload_type_;
over_rtx = false;
} else {
// Without abs-send-time or transport sequence number a media packet
@ -1093,6 +1086,8 @@ bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
// Remember marker bit to determine if padding can be inserted with
// sequence number following |packet|.
last_packet_marker_bit_ = packet->Marker();
// Remember payload type to use in the padding packet if rtx is disabled.
last_payload_type_ = packet->PayloadType();
// Save timestamps to generate timestamp field and extensions for the padding.
last_rtp_timestamp_ = packet->Timestamp();
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();

4
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -81,8 +81,6 @@ class RTPSender {
int32_t DeRegisterSendPayload(const int8_t payload_type);
void SetSendPayloadType(int8_t payload_type);
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
@ -277,7 +275,7 @@ class RTPSender {
size_t max_packet_size_;
int8_t payload_type_ RTC_GUARDED_BY(send_critsect_);
int8_t last_payload_type_ RTC_GUARDED_BY(send_critsect_);
std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
RtpHeaderExtensionMap rtp_header_extension_map_

69
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -172,7 +172,6 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, populate_network2));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
@ -346,7 +345,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberFailsOnNotSending) {
EXPECT_FALSE(rtp_sender_->AssignSequenceNumber(packet.get()));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet);
@ -354,7 +353,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
ASSERT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
packet->SetMarker(false);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
// Packet without marker bit doesn't allow padding.
// Packet without marker bit doesn't allow padding on video stream.
EXPECT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
packet->SetMarker(true);
@ -363,6 +362,37 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
EXPECT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
}
TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
MockTransport transport;
const bool kEnableAudio = true;
rtp_sender_.reset(new RTPSender(
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
std::unique_ptr<RtpPacketToSend> audio_packet = rtp_sender_->AllocatePacket();
// Padding on audio stream allowed regardless of marker in the last packet.
audio_packet->SetMarker(false);
audio_packet->SetPayloadType(kPayload);
rtp_sender_->AssignSequenceNumber(audio_packet.get());
const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPaddingSize,
rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(
kMinPaddingSize,
rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5, PacedPacketInfo()));
}
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
constexpr size_t kPaddingSize = 100;
auto packet = rtp_sender_->AllocatePacket();
@ -962,7 +992,6 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
rtp_sender_->SetStorePacketsStatus(true, 10);
// Parameters selected to generate a single FEC packet per media packet.
@ -1022,7 +1051,6 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
rtp_sender_->SetStorePacketsStatus(true, 10);
// Need extension to be registered for timing frames to be sent.
@ -1122,7 +1150,6 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
@ -1145,7 +1172,6 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
}
TEST_P(RtpSenderTest, FecOverheadRate) {
constexpr int kMediaPayloadType = 127;
constexpr int kFlexfecPayloadType = 118;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
@ -1163,7 +1189,6 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
@ -1963,40 +1988,12 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
SendGenericPayload();
}
TEST_P(RtpSenderTest, SendAudioPadding) {
MockTransport transport;
const bool kEnableAudio = true;
rtp_sender_.reset(new RTPSender(
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(kPaddingSize,
rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
// Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(testing::Return(true));
EXPECT_EQ(
kMinPaddingSize,
rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5, PacedPacketInfo()));
}
TEST_P(RtpSenderTest, SendsKeepAlive) {
MockTransport transport;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);