52033d6ea1
The VUI part an SPS may specify max_num_reorder_frames and max_dec_frame_buffering. These may cause a decoder to buffer a number of frame prior allowing decode, leading to delays, even if no frames using such references (ie B-frames) are sent. Because of this we update any SPS block emitted by the encoder. Also, a bunch of refactoring of H264-related code to reduce code duplication. BUG= Review-Url: https://codereview.webrtc.org/1979443004 Cr-Commit-Position: refs/heads/master@{#13010}
77 lines
2.8 KiB
C++
77 lines
2.8 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#include <string>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtpPacketizer {
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public:
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static RtpPacketizer* Create(RtpVideoCodecTypes type,
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size_t max_payload_len,
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const RTPVideoTypeHeader* rtp_type_header,
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FrameType frame_type);
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virtual ~RtpPacketizer() {}
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virtual void SetPayloadData(const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation) = 0;
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// Get the next payload with payload header.
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// buffer is a pointer to where the output will be written.
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// bytes_to_send is an output variable that will contain number of bytes
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// written to buffer. The parameter last_packet is true for the last packet of
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// the frame, false otherwise (i.e., call the function again to get the
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// next packet).
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// Returns true on success or false if there was no payload to packetize.
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virtual bool NextPacket(uint8_t* buffer,
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size_t* bytes_to_send,
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bool* last_packet) = 0;
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virtual ProtectionType GetProtectionType() = 0;
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virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
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virtual std::string ToString() = 0;
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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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const uint8_t* payload;
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size_t payload_length;
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FrameType frame_type;
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RTPTypeHeader type;
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};
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static RtpDepacketizer* Create(RtpVideoCodecTypes 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 // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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