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Reland "Add stereo codec header and pass it through RTP"

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This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}

TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org

Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This commit is contained in:
Emircan Uysaler
2017-11-28 09:45:25 -08:00
committed by Commit Bot
parent f72ab8395a
commit 90612a681b
27 changed files with 658 additions and 75 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
modules/rtp_rtcp/source/rtp_format.cc
View File

@ -14,6 +14,7 @@
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
@ -36,6 +37,10 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
RTC_CHECK(rtp_type_header);
return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
last_packet_reduction_len);
case kRtpVideoStereo:
return new RtpPacketizerStereo(rtp_type_header->stereo, frame_type,
max_payload_len,
last_packet_reduction_len);
case kRtpVideoGeneric:
return new RtpPacketizerGeneric(frame_type, max_payload_len,
last_packet_reduction_len);
@ -53,6 +58,8 @@ RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
return new RtpDepacketizerVp8();
case kRtpVideoVp9:
return new RtpDepacketizerVp9();
case kRtpVideoStereo:
return new RtpDepacketizerStereo();
case kRtpVideoGeneric:
return new RtpDepacketizerGeneric();
case kRtpVideoNone: