blob: fd44bd1980967923cd2354dd08f594a73ff584f3 [file] [log] [blame]
/*
* 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"
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);
// 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;
RtpPacketizerGeneric(const RtpPacketizerGeneric&) = delete;
RtpPacketizerGeneric& operator=(const RtpPacketizerGeneric&) = delete;
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);
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_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_