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Deleting all NetEq3 files

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NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-05-13 10:40:52 +00:00
parent 4d363ae305
commit c3e8abda7c
78 changed files with 0 additions and 26952 deletions
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102
webrtc/modules/audio_coding/neteq/rtcp.h
View File

@ -1,102 +0,0 @@
/*
* 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.
*/
/*
* RTCP statistics reporting.
*/
#ifndef RTCP_H
#define RTCP_H
#include "typedefs.h"
typedef struct
{
uint16_t cycles; /* The number of wrap-arounds for the sequence number */
uint16_t max_seq; /* The maximum sequence number received
(starts from 0 again after wrap around) */
uint16_t base_seq; /* The sequence number of the first packet that arrived */
uint32_t received; /* The number of packets that has been received */
uint32_t rec_prior; /* Number of packets received when last report was generated */
uint32_t exp_prior; /* Number of packets that should have been received if no
packets were lost. Stored value from last report. */
uint32_t jitter; /* Jitter statistics at this instance (calculated according to RFC) */
int32_t transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */
} WebRtcNetEQ_RTCP_t;
/****************************************************************************
* WebRtcNetEQ_RTCPInit(...)
*
* This function calculates the parameters that are needed for the RTCP
* report.
*
* Input:
* - RTCP_inst : RTCP instance, that contains information about the
* packets that have been received etc.
* - seqNo : Packet number of the first received frame.
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo);
/****************************************************************************
* WebRtcNetEQ_RTCPUpdate(...)
*
* This function calculates the parameters that are needed for the RTCP
* report.
*
* Input:
* - RTCP_inst : RTCP instance, that contains information about the
* packets that have been received etc.
* - seqNo : Packet number of the first received frame.
* - timeStamp : Time stamp from the RTP header.
* - recTime : Time (in RTP timestamps) when this packet was received.
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, uint16_t uw16_seqNo,
uint32_t uw32_timeStamp, uint32_t uw32_recTime);
/****************************************************************************
* WebRtcNetEQ_RTCPGetStats(...)
*
* This function calculates the parameters that are needed for the RTCP
* report.
*
* Input:
* - RTCP_inst : RTCP instance, that contains information about the
* packets that have been received etc.
* - doNotReset : If non-zero, the fraction lost statistics will not
* be reset.
*
* Output:
* - RTCP_inst : Updated RTCP information (some statistics are
* reset when generating this report)
* - fraction_lost : Number of lost RTP packets divided by the number of
* expected packets, since the last RTCP Report.
* - cum_lost : Cumulative number of lost packets during this
* session.
* - ext_max : Extended highest sequence number received.
* - jitter : Inter-arrival jitter.
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
uint16_t *puw16_fraction_lost,
uint32_t *puw32_cum_lost, uint32_t *puw32_ext_max,
uint32_t *puw32_jitter, int16_t doNotReset);
#endif