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Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"

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This is a reland of 17608dc4592fe25c1effdd75bf856f4af251942e

Downstream fixed, relanding.

Original change's description:
> RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
>
> Downstream usage of SetSsrc() / SetRtxSsrc() should now be gone. Let's
> remove them, make the members const, and remove now unnecessary locking.
>
> Bug: webrtc:10774
> Change-Id: Ie4c1b3935508cf329c5553030f740c565d32e04b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155660
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29475}

TBR=nisse@webrtc.org

Bug: webrtc:10774
Change-Id: I759bed3ff1909857696c6d1b13df595a5e552f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157049
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29486}
This commit is contained in:
Erik Språng
2019-10-14 13:00:35 +02:00
committed by Commit Bot
parent ac0a4cbbd8
commit c9348218cf
14 changed files with 44 additions and 215 deletions
Download Patch File Download Diff File Expand all files Collapse all files

28
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -2562,34 +2562,6 @@ TEST_P(RtpSenderTest, SetsCaptureTimeAndPopulatesTransmissionOffset) {
EXPECT_EQ(*transmission_time_extension, 2 * kOffsetMs * kTimestampTicksPerMs);
}
TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSsrcChange) {
const int64_t kRtt = 10;
rtp_sender_->SetSendingMediaStatus(true);
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
rtp_sender_->SetStorePacketsStatus(true, 10);
rtp_sender_->SetRtt(kRtt);
// Send a packet and record its sequence numbers.
SendGenericPacket();
ASSERT_EQ(1u, transport_.sent_packets_.size());
const uint16_t packet_seqence_number =
transport_.sent_packets_.back().SequenceNumber();
// Advance time and make sure it can be retransmitted, even if we try to set
// the ssrc the what it already is.
rtp_sender_->SetSSRC(kSsrc);
fake_clock_.AdvanceTimeMilliseconds(kRtt);
EXPECT_GT(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
// Change the SSRC, then move the time and try to retransmit again. The old
// packet should now be gone.
rtp_sender_->SetSSRC(kSsrc + 1);
fake_clock_.AdvanceTimeMilliseconds(kRtt);
EXPECT_EQ(rtp_sender_->ReSendPacket(packet_seqence_number), 0);
}
TEST_P(RtpSenderTestWithoutPacer, ClearHistoryOnSequenceNumberCange) {
const int64_t kRtt = 10;