Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.h b/webrtc/modules/audio_coding/neteq/tools/packet.h
new file mode 100644
index 0000000..eb8ce28
--- /dev/null
+++ b/webrtc/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_