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9bb8f0553d
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481. With the new approach we can just use the generic RTP packetizer to pass frames over the wire as the specific info is contained within the bitstream. This makes the new codec more modular and reduces its footprint. Bug: webrtc:7671 Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552 Reviewed-on: https://webrtc-review.googlesource.com/43220 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21753}
63 lines
2.3 KiB
C++
63 lines
2.3 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(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 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 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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