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Add extended header containing frame ID to the generic packetizer.

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Also changes default value of frame ID in RTPVideoHeader to
kNoPictureId. Special care should be take so that picture ID will not
be set in RTPVideoHeader unless the client on the end supports
deserializing extended generic header.

Bug: webrtc:9582
Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167
Reviewed-on: https://webrtc-review.googlesource.com/92084
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24250}
This commit is contained in:
Sami Kalliomäki
2018-08-08 11:37:59 +02:00
committed by Commit Bot
parent 9489c3a2ea
commit 426a80ce08
8 changed files with 185 additions and 46 deletions
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37
modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -244,9 +244,10 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
0, 1500));
RTPVideoHeader video_header;
EXPECT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
sizeof(kPayloadData), nullptr, nullptr, nullptr,
sizeof(kPayloadData), nullptr, &video_header, nullptr,
kDefaultExpectedRetransmissionTimeMs));
}
};
@ -968,9 +969,10 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
uint8_t generic_header = sent_payload[0];
@ -985,7 +987,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
sent_payload = transport_.last_sent_packet().payload();
generic_header = sent_payload[0];
@ -1299,9 +1301,10 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
.Times(::testing::AtLeast(2));
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
EXPECT_EQ(1U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@ -1310,7 +1313,7 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
EXPECT_EQ(2U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@ -1373,10 +1376,11 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
rtp_sender_->ProcessBitrate();
// Send a few frames.
RTPVideoHeader video_header;
for (uint32_t i = 0; i < kNumPackets; ++i) {
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
}
@ -1457,9 +1461,10 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@ -1501,7 +1506,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
rtp_sender_->SetFecParameters(fec_params, fec_params);
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
@ -1518,9 +1523,10 @@ TEST_P(RtpSenderAudioTest, SendAudio) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
@ -1537,9 +1543,10 @@ TEST_P(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
@ -1576,22 +1583,23 @@ TEST_P(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms, 0, nullptr, 0, nullptr,
nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0, nullptr, 0, nullptr,
nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(transport_.last_sent_packet().Marker());
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0, nullptr, 0, nullptr,
nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// Marker Bit should be set to 0 for rest of the packets.
EXPECT_FALSE(transport_.last_sent_packet().Marker());
}
@ -1608,9 +1616,10 @@ TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kVideoFrameKey, kPayloadType, 1234, 4321, payload, sizeof(payload),
nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());