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/rtcp.h b/webrtc/modules/audio_coding/neteq/rtcp.h
new file mode 100644
index 0000000..2a765ef
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/rtcp.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (c) 2011 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_RTCP_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declaration.
+struct RTPHeader;
+
+class Rtcp {
+ public:
+ Rtcp() {
+ Init(0);
+ }
+
+ ~Rtcp() {}
+
+ // Resets the RTCP statistics, and sets the first received sequence number.
+ void Init(uint16_t start_sequence_number);
+
+ // Updates the RTCP statistics with a new received packet.
+ void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
+
+ // Returns the current RTCP statistics. If |no_reset| is true, the statistics
+ // are not reset, otherwise they are.
+ void GetStatistics(bool no_reset, RtcpStatistics* stats);
+
+ private:
+ uint16_t cycles_; // The number of wrap-arounds for the sequence number.
+ uint16_t max_seq_no_; // The maximum sequence number received. Starts over
+ // from 0 after wrap-around.
+ uint16_t base_seq_no_; // The sequence number of the first received packet.
+ uint32_t received_packets_; // The number of packets that have been received.
+ uint32_t received_packets_prior_; // Number of packets received when last
+ // report was generated.
+ uint32_t expected_prior_; // Expected number of packets, at the time of the
+ // last report.
+ uint32_t jitter_; // Current jitter value.
+ int32_t transit_; // Clock difference for previous packet.
+
+ DISALLOW_COPY_AND_ASSIGN(Rtcp);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_