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Add sender controlled playout delay limits

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This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.

The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.

There are no tests at this time and most of testing is done with chromium
webrtc prototype.

On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.

BUG=webrtc:5895

Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This commit is contained in:
isheriff
2016-06-08 00:24:21 -07:00
committed by Commit bot
parent 5d910286e1
commit 6b4b5f3770
41 changed files with 859 additions and 407 deletions
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25
webrtc/modules/video_coding/packet.cc
View File

@ -32,7 +32,9 @@ VCMPacket::VCMPacket()
insertStartCode(false),
width(0),
height(0),
codecSpecificHeader() {}
video_header() {
video_header.playout_delay = {-1, -1};
}
VCMPacket::VCMPacket(const uint8_t* ptr,
const size_t size,
@ -45,7 +47,6 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
sizeBytes(size),
markerBit(rtpHeader.header.markerBit),
timesNacked(-1),
frameType(rtpHeader.frameType),
codec(kVideoCodecUnknown),
isFirstPacket(rtpHeader.type.Video.isFirstPacket),
@ -53,8 +54,18 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
insertStartCode(false),
width(rtpHeader.type.Video.width),
height(rtpHeader.type.Video.height),
codecSpecificHeader(rtpHeader.type.Video) {
video_header(rtpHeader.type.Video) {
CopyCodecSpecifics(rtpHeader.type.Video);
if (markerBit) {
video_header.rotation = rtpHeader.type.Video.rotation;
}
// Playout decisions are made entirely based on first packet in a frame.
if (isFirstPacket) {
video_header.playout_delay = rtpHeader.type.Video.playout_delay;
} else {
video_header.playout_delay = {-1, -1};
}
}
VCMPacket::VCMPacket(const uint8_t* ptr,
@ -70,7 +81,6 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
sizeBytes(size),
markerBit(mBit),
timesNacked(-1),
frameType(kVideoFrameDelta),
codec(kVideoCodecUnknown),
isFirstPacket(false),
@ -78,7 +88,7 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
insertStartCode(false),
width(0),
height(0),
codecSpecificHeader() {}
video_header() {}
void VCMPacket::Reset() {
payloadType = 0;
@ -96,13 +106,10 @@ void VCMPacket::Reset() {
insertStartCode = false;
width = 0;
height = 0;
memset(&codecSpecificHeader, 0, sizeof(RTPVideoHeader));
memset(&video_header, 0, sizeof(RTPVideoHeader));
}
void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
if (markerBit) {
codecSpecificHeader.rotation = videoHeader.rotation;
}
switch (videoHeader.codec) {
case kRtpVideoVp8:
// Handle all packets within a frame as depending on the previous packet