| Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..16bd3fc2a2d6cc209f5536de603f042c7ddabfd6
|
| --- /dev/null
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
|
| @@ -0,0 +1,66 @@
|
| +/*
|
| + * Copyright (c) 2015 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
|
| +#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
|
| +
|
| +#include "webrtc/base/buffer.h"
|
| +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
|
| +
|
| +namespace webrtc {
|
| +namespace rtcp {
|
| +
|
| +class App : public RtcpPacket {
|
| + public:
|
| + static const uint8_t kPacketType = 204;
|
| + // 28 bytes for UDP header
|
| + // 12 bytes for RTCP app header
|
| + static const size_t kMaxDataSize = IP_PACKET_SIZE - 12 - 28;
|
| + App() : sub_type_(0), ssrc_(0), name_(0) {}
|
| +
|
| + virtual ~App() {}
|
| +
|
| + // Parse assumes header is already parsed and validated.
|
| + bool Parse(const RTCPUtility::RtcpCommonHeader& header,
|
| + const uint8_t* payload); // Size of the payload is in the header.
|
| +
|
| + void From(uint32_t ssrc) { ssrc_ = ssrc; }
|
| + void WithSubType(uint8_t subtype);
|
| + void WithName(uint32_t name) { name_ = name; }
|
| + void WithData(const uint8_t* data, size_t data_length);
|
| +
|
| + uint8_t sub_type() const { return sub_type_; }
|
| + uint32_t ssrc() const { return ssrc_; }
|
| + uint32_t name() const { return name_; }
|
| + size_t data_size() const { return data_.size(); }
|
| + const uint8_t* data() const { return data_.data(); }
|
| +
|
| + protected:
|
| + bool Create(uint8_t* packet,
|
| + size_t* index,
|
| + size_t max_length,
|
| + RtcpPacket::PacketReadyCallback* callback) const override;
|
| +
|
| + private:
|
| + size_t BlockLength() const override { return 12 + data_.size(); }
|
| +
|
| + uint8_t sub_type_;
|
| + uint32_t ssrc_;
|
| + uint32_t name_;
|
| + rtc::Buffer data_;
|
| +
|
| + RTC_DISALLOW_COPY_AND_ASSIGN(App);
|
| +};
|
| +
|
| +} // namespace rtcp
|
| +} // namespace webrtc
|
| +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
|
|
|