mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
89 lines
3.1 KiB
C++
89 lines
3.1 KiB
C++
/*
|
|
* Copyright (c) 2013 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_RTP_FORMAT_VIDEO_GENERIC_H_
|
|
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
|
|
|
|
#include <stdint.h>
|
|
|
|
#include <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/rtp_rtcp/source/rtp_format.h"
|
|
#include "rtc_base/constructor_magic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpPacketToSend;
|
|
struct RTPVideoHeader;
|
|
|
|
namespace RtpFormatVideoGeneric {
|
|
static const uint8_t kKeyFrameBit = 0x01;
|
|
static const uint8_t kFirstPacketBit = 0x02;
|
|
// If this bit is set, there will be an extended header contained in this
|
|
// packet. This was added later so old clients will not send this.
|
|
static const uint8_t kExtendedHeaderBit = 0x04;
|
|
} // namespace RtpFormatVideoGeneric
|
|
|
|
class RtpPacketizerGeneric : public RtpPacketizer {
|
|
public:
|
|
// Initialize with payload from encoder.
|
|
// The payload_data must be exactly one encoded generic frame.
|
|
// Packets returned by |NextPacket| will contain the generic payload header.
|
|
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
|
|
PayloadSizeLimits limits,
|
|
const RTPVideoHeader& rtp_video_header,
|
|
VideoFrameType frametype);
|
|
// Initialize with payload from encoder.
|
|
// The payload_data must be exactly one encoded generic frame.
|
|
// Packets returned by |NextPacket| will contain raw payload without the
|
|
// generic payload header.
|
|
RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
|
|
PayloadSizeLimits limits);
|
|
|
|
~RtpPacketizerGeneric() override;
|
|
|
|
size_t NumPackets() const override;
|
|
|
|
// Get the next payload.
|
|
// Write payload and set marker bit of the |packet|.
|
|
// Returns true on success, false otherwise.
|
|
bool NextPacket(RtpPacketToSend* packet) override;
|
|
|
|
private:
|
|
// Fills header_ and header_size_ members.
|
|
void BuildHeader(const RTPVideoHeader& rtp_video_header,
|
|
VideoFrameType frame_type);
|
|
|
|
uint8_t header_[3];
|
|
size_t header_size_;
|
|
rtc::ArrayView<const uint8_t> remaining_payload_;
|
|
std::vector<int> payload_sizes_;
|
|
std::vector<int>::const_iterator current_packet_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
|
|
};
|
|
|
|
// Depacketizer for generic codec.
|
|
class RtpDepacketizerGeneric : public RtpDepacketizer {
|
|
public:
|
|
// Parses the generic payload header if |generic_header_enabled| is true,
|
|
// returns raw payload otherwise.
|
|
explicit RtpDepacketizerGeneric(bool generic_header_enabled);
|
|
~RtpDepacketizerGeneric() override;
|
|
|
|
bool Parse(ParsedPayload* parsed_payload,
|
|
const uint8_t* payload_data,
|
|
size_t payload_data_length) override;
|
|
|
|
private:
|
|
bool generic_header_enabled_;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
|