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87e2d785a0
This cl deprecates the FrameType enum, and adds aliases AudioFrameType and VideoFrameType. After downstream usage is updated, the enums will be separated and be moved out of common_types.h. Bug: webrtc:6883 Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27011}
87 lines
2.9 KiB
C++
87 lines
2.9 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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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#include <stdint.h>
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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namespace webrtc {
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class RtpPacketToSend;
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class RtpPacketizer {
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public:
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struct PayloadSizeLimits {
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int max_payload_len = 1200;
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int first_packet_reduction_len = 0;
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int last_packet_reduction_len = 0;
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// Reduction len for packet that is first & last at the same time.
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int single_packet_reduction_len = 0;
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};
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static std::unique_ptr<RtpPacketizer> Create(
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VideoCodecType type,
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rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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// Codec-specific details.
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const RTPVideoHeader& rtp_video_header,
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VideoFrameType frame_type,
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const RTPFragmentationHeader* fragmentation);
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virtual ~RtpPacketizer() = default;
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// Returns number of remaining packets to produce by the packetizer.
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virtual size_t NumPackets() const = 0;
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// Get the next payload with payload header.
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// Write payload and set marker bit of the |packet|.
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// Returns true on success, false otherwise.
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virtual bool NextPacket(RtpPacketToSend* packet) = 0;
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// Split payload_len into sum of integers with respect to |limits|.
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// Returns empty vector on failure.
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static std::vector<int> SplitAboutEqually(int payload_len,
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const PayloadSizeLimits& limits);
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};
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// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
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// of the parsed payload, rather than just a pointer into the incoming buffer.
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// This way we can move some parsing out from the jitter buffer into here, and
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// the jitter buffer can just store that pointer rather than doing a copy there.
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class RtpDepacketizer {
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public:
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struct ParsedPayload {
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RTPVideoHeader& video_header() { return video; }
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const RTPVideoHeader& video_header() const { return video; }
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RTPVideoHeader video;
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const uint8_t* payload;
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size_t payload_length;
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VideoFrameType frame_type;
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};
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static RtpDepacketizer* Create(VideoCodecType type);
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virtual ~RtpDepacketizer() {}
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// Parses the RTP payload, parsed result will be saved in |parsed_payload|.
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virtual bool Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload_data,
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size_t payload_data_length) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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