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Diffstat (limited to 'modules/audio_coding/neteq/tools/packet.h')
-rw-r--r-- | modules/audio_coding/neteq/tools/packet.h | 117 |
1 files changed, 117 insertions, 0 deletions
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h new file mode 100644 index 00000000..eb8ce28a --- /dev/null +++ b/modules/audio_coding/neteq/tools/packet.h @@ -0,0 +1,117 @@ +/* + * Copyright (c) 2014 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_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ +#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ + +#include <list> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/common_types.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class RtpHeaderParser; + +namespace test { + +// Class for handling RTP packets in test applications. +class Packet { + public: + // Creates a packet, with the packet payload (including header bytes) in + // |packet_memory|. The length of |packet_memory| is |allocated_bytes|. + // The new object assumes ownership of |packet_memory| and will delete it + // when the Packet object is deleted. The |time_ms| is an extra time + // associated with this packet, typically used to denote arrival time. + // The first bytes in |packet_memory| will be parsed using |parser|. + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + double time_ms, + const RtpHeaderParser& parser); + + // Same as above, but with the extra argument |virtual_packet_length_bytes|. + // This is typically used when reading RTP dump files that only contain the + // RTP headers, and no payload (a.k.a RTP dummy files or RTP light). The + // |virtual_packet_length_bytes| tells what size the packet had on wire, + // including the now discarded payload, whereas |allocated_bytes| is the + // length of the remaining payload (typically only the RTP header). + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms, + const RtpHeaderParser& parser); + + // The following two constructors are the same as above, but without a + // parser. Note that when the object is constructed using any of these + // methods, the header will be parsed using a default RtpHeaderParser object. + // In particular, RTP header extensions won't be parsed. + Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms); + + Packet(uint8_t* packet_memory, + size_t allocated_bytes, + size_t virtual_packet_length_bytes, + double time_ms); + + virtual ~Packet() {} + + // Parses the first bytes of the RTP payload, interpreting them as RED headers + // according to RFC 2198. The headers will be inserted into |headers|. The + // caller of the method assumes ownership of the objects in the list, and + // must delete them properly. + bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const; + + // Deletes all RTPHeader objects in |headers|, but does not delete |headers| + // itself. + static void DeleteRedHeaders(std::list<RTPHeader*>* headers); + + const uint8_t* payload() const { return payload_; } + + size_t packet_length_bytes() const { return packet_length_bytes_; } + + size_t payload_length_bytes() const { return payload_length_bytes_; } + + size_t virtual_packet_length_bytes() const { + return virtual_packet_length_bytes_; + } + + size_t virtual_payload_length_bytes() const { + return virtual_payload_length_bytes_; + } + + const RTPHeader& header() const { return header_; } + + void set_time_ms(double time) { time_ms_ = time; } + double time_ms() const { return time_ms_; } + bool valid_header() const { return valid_header_; } + + private: + bool ParseHeader(const RtpHeaderParser& parser); + void CopyToHeader(RTPHeader* destination) const; + + RTPHeader header_; + scoped_ptr<uint8_t[]> payload_memory_; + const uint8_t* payload_; // First byte after header. + const size_t packet_length_bytes_; // Total length of packet. + size_t payload_length_bytes_; // Length of the payload, after RTP header. + // Zero for dummy RTP packets. + // Virtual lengths are used when parsing RTP header files (dummy RTP files). + const size_t virtual_packet_length_bytes_; + size_t virtual_payload_length_bytes_; + double time_ms_; // Used to denote a packet's arrival time. + bool valid_header_; // Set by the RtpHeaderParser. + + DISALLOW_COPY_AND_ASSIGN(Packet); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ |