rtcp::App moved into own file and got Parse function

webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h

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..4a3edcbe458d7bae12ed3446b0a7ebaed7c53b48
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
@@ -0,0 +1,68 @@
+/*
+ *  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 <vector>

[åsapersson, 2015/11/17 15:31:36]

needed?

[danilchap, 2015/11/17 17:12:36]

Done.

+
+#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; }
+  bool WithSubType(uint8_t subtype);
+  void WithName(uint32_t name) { name_ = name; }
+  bool 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_