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2ff1f2a342
This is a reland of 0b9e01d605
Original change's description:
> Refactor RtpVideoStreamReceiver without RtpReceiver.
>
> Bug: webrtc:7135
> Change-Id: Iabf3330e579b892efc160683f9f90efbf6ff9a40
> Reviewed-on: https://webrtc-review.googlesource.com/92398
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24232}
Bug: webrtc:7135
Change-Id: I707d4c5262e7b428bc7ceac2d886ff34c4a8d76a
Reviewed-on: https://webrtc-review.googlesource.com/93261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24254}
66 lines
2.4 KiB
C++
66 lines
2.4 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_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(VideoCodecType 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 RTPVideoHeader* rtp_video_header,
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FrameType frame_type) {
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RTC_CHECK(type == kVideoCodecGeneric || rtp_video_header);
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switch (type) {
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case kVideoCodecH264: {
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const auto& h264 =
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absl::get<RTPVideoHeaderH264>(rtp_video_header->video_type_header);
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return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
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h264.packetization_mode);
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}
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case kVideoCodecVP8:
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return new RtpPacketizerVp8(rtp_video_header->vp8(), max_payload_len,
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last_packet_reduction_len);
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case kVideoCodecVP9: {
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const auto& vp9 =
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absl::get<RTPVideoHeaderVP9>(rtp_video_header->video_type_header);
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return new RtpPacketizerVp9(vp9, max_payload_len,
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last_packet_reduction_len);
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}
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case kVideoCodecGeneric:
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RTC_CHECK(rtp_video_header);
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return new RtpPacketizerGeneric(*rtp_video_header, frame_type,
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max_payload_len,
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last_packet_reduction_len);
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default:
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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(VideoCodecType type) {
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switch (type) {
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case kVideoCodecH264:
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return new RtpDepacketizerH264();
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case kVideoCodecVP8:
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return new RtpDepacketizerVp8();
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case kVideoCodecVP9:
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return new RtpDepacketizerVp9();
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default:
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return new RtpDepacketizerGeneric();
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}
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}
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} // namespace webrtc
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