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https://github.com/mollyim/webrtc.git
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90612a681b
This is a reland of 20f2133d5d
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}
70 lines
2.6 KiB
C++
70 lines
2.6 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include <utility>
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#include "modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
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namespace webrtc {
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RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
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size_t max_payload_len,
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size_t last_packet_reduction_len,
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const RTPVideoTypeHeader* rtp_type_header,
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FrameType frame_type) {
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switch (type) {
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case kRtpVideoH264:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
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rtp_type_header->H264.packetization_mode);
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case kRtpVideoVp8:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoVp9:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoStereo:
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return new RtpPacketizerStereo(rtp_type_header->stereo, frame_type,
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max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoGeneric:
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return new RtpPacketizerGeneric(frame_type, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoNone:
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RTC_NOTREACHED();
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}
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return nullptr;
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}
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RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
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switch (type) {
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case kRtpVideoH264:
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return new RtpDepacketizerH264();
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case kRtpVideoVp8:
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return new RtpDepacketizerVp8();
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case kRtpVideoVp9:
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return new RtpDepacketizerVp9();
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case kRtpVideoStereo:
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return new RtpDepacketizerStereo();
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case kRtpVideoGeneric:
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return new RtpDepacketizerGeneric();
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case kRtpVideoNone:
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assert(false);
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}
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return nullptr;
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}
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} // namespace webrtc
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