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Introduce VideoRtpDepacketizer interface to replace RtpDepacketizer

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Bug: webrtc:11152
Change-Id: I20fd81233080d45d8978e5e57d0be6b592f44f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161322
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30018}
This commit is contained in:
Danil Chapovalov 2019-12-05 14:41:45 +01:00 committed by Commit Bot
parent fc9079700c
commit fc50e44a03
3 changed files with 36 additions and 1 deletions
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1
modules/rtp_rtcp/BUILD.gn
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@ -210,6 +210,7 @@ rtc_library("rtp_rtcp") {
"source/ulpfec_header_reader_writer.h",
"source/ulpfec_receiver_impl.cc",
"source/ulpfec_receiver_impl.h",
"source/video_rtp_depacketizer.h",
]
if (rtc_enable_bwe_test_logging) {

2
modules/rtp_rtcp/source/rtp_format.h
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@ -60,7 +60,7 @@ class RtpPacketizer {
const PayloadSizeLimits& limits);
};
// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
// TODO(bugs.webrtc.org/11152): Update the depacketizer to return a copy
// of the parsed payload, rather than just a pointer into the incoming buffer.
// This way we can move some parsing out from the jitter buffer into here, and
// the jitter buffer can just store that pointer rather than doing a copy there.

34
modules/rtp_rtcp/source/video_rtp_depacketizer.h Normal file
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@ -0,0 +1,34 @@
/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
class VideoRtpDepacketizer {
public:
struct ParsedRtpPayload {
RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer video_payload;
};
virtual ~VideoRtpDepacketizer() = default;
virtual absl::optional<ParsedRtpPayload> Parse(
rtc::CopyOnWriteBuffer rtp_payload) = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_