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New method RtpReceiver::GetLatestTimestamps.

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The two timestamps, rtp time and corresponding system time, are always
used together, for audio/video sync. The new method reads both
timestamps, without releasing a lock in between. Ensures that the
caller gets values corresponding to the same packet.

Bug: webrtc:7135
Change-Id: I25bdcbe9ad620016bfad39841b339c266efade14
Reviewed-on: https://webrtc-review.googlesource.com/4062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20120}
This commit is contained in:
Niels Möller
2017-10-03 15:28:26 +02:00
committed by Commit Bot
parent 1cfa95b8b8
commit c3fa8e1ce7
6 changed files with 31 additions and 35 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -185,8 +185,11 @@ TEST_F(RtpRtcpAudioTest, Basic) {
EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
uint32_t timestamp;
EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
int64_t receive_time_ms;
EXPECT_TRUE(
rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
EXPECT_EQ(test_timestamp, timestamp);
EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
}
TEST_F(RtpRtcpAudioTest, DTMF) {
@ -264,23 +267,30 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) {
uint32_t in_timestamp = 0;
for (const auto& c : kCngCodecs) {
uint32_t timestamp;
int64_t receive_time_ms;
EXPECT_TRUE(module1->SendOutgoingData(
webrtc::kAudioFrameSpeech, kPcmuPayloadType, in_timestamp, -1,
kTestPayload, 4, nullptr, nullptr, nullptr));
EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
EXPECT_TRUE(
rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
in_timestamp += 10;
fake_clock.AdvanceTimeMilliseconds(20);
EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameCN, c.payload_type,
in_timestamp, -1, kTestPayload, 1,
nullptr, nullptr, nullptr));
EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
EXPECT_TRUE(
rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
in_timestamp += 10;
fake_clock.AdvanceTimeMilliseconds(20);
}
}