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Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

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This is a reland of 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2

Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}

TBR=stefan@webrtc.org

Bug: webrtc:11340
Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30479}
This commit is contained in:
Erik Språng
2020-02-06 17:10:08 +01:00
committed by Commit Bot
parent ecd6fc84cf
commit 56e611bbda
18 changed files with 163 additions and 277 deletions
Download Patch File Download Diff File Expand all files Collapse all files

62
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View File

@ -54,6 +54,7 @@ enum : int { // The first valid value is 1.
kVideoRotationExtensionId,
kVideoTimingExtensionId,
kAbsoluteCaptureTimeExtensionId,
kPlayoutDelayExtensionId
};
constexpr int kPayload = 100;
@ -87,6 +88,8 @@ class LoopbackTransportTest : public webrtc::Transport {
kFrameMarkingExtensionId);
receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
kAbsoluteCaptureTimeExtensionId);
receivers_extensions_.Register<PlayoutDelayLimits>(
kPlayoutDelayExtensionId);
}
bool SendRtp(const uint8_t* data,
@ -121,7 +124,6 @@ class TestRtpSenderVideo : public RTPSenderVideo {
config.clock = clock;
config.rtp_sender = rtp_sender;
config.flexfec_sender = flexfec_sender;
config.playout_delay_oracle = &playout_delay_oracle_;
config.field_trials = &field_trials;
return config;
}()) {}
@ -134,7 +136,6 @@ class TestRtpSenderVideo : public RTPSenderVideo {
retransmission_settings,
expected_retransmission_time_ms);
}
PlayoutDelayOracle playout_delay_oracle_;
};
class FieldTrials : public WebRtcKeyValueConfig {
@ -792,6 +793,63 @@ TEST_P(RtpSenderVideoTest, AbsoluteCaptureTime) {
EXPECT_EQ(packets_with_abs_capture_time, 1);
}
TEST_P(RtpSenderVideoTest, PopulatesPlayoutDelay) {
// Single packet frames.
constexpr size_t kPacketSize = 123;
uint8_t kFrame[kPacketSize];
rtp_module_->RegisterRtpHeaderExtension(PlayoutDelayLimits::kUri,
kPlayoutDelayExtensionId);
const PlayoutDelay kExpectedDelay = {10, 20};
// Send initial key-frame without playout delay.
RTPVideoHeader hdr;
hdr.frame_type = VideoFrameType::kVideoFrameKey;
hdr.codec = VideoCodecType::kVideoCodecVP8;
auto& vp8_header = hdr.video_type_header.emplace<RTPVideoHeaderVP8>();
vp8_header.temporalIdx = 0;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
EXPECT_FALSE(
transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
// Set playout delay on a discardable frame.
hdr.playout_delay = kExpectedDelay;
hdr.frame_type = VideoFrameType::kVideoFrameDelta;
vp8_header.temporalIdx = 1;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
PlayoutDelay received_delay = PlayoutDelay::Noop();
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
// Set playout delay on a non-discardable frame, the extension should still
// be populated since dilvery wasn't guaranteed on the last one.
hdr.playout_delay = PlayoutDelay::Noop(); // Inidcates "no change".
vp8_header.temporalIdx = 0;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
// The next frame does not need the extensions since it's delivery has
// already been guaranteed.
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
EXPECT_FALSE(
transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
// Insert key-frame, we need to refresh the state here.
hdr.frame_type = VideoFrameType::kVideoFrameKey;
rtp_sender_video_.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, nullptr,
hdr, kDefaultExpectedRetransmissionTimeMs);
ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
&received_delay));
EXPECT_EQ(received_delay, kExpectedDelay);
}
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderVideoTest,
::testing::Bool());