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Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS.

Browse Source 
BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2013-05-16 23:54:54 +00:00
parent fe307e1332
commit 8630cfe016
3 changed files with 44 additions and 13 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/neteq/neteq_error_codes.h
View File

@ -40,6 +40,8 @@
#define RECIN_CNG_ERROR -3001
#define RECIN_UNKNOWNPAYLOAD -3002
#define RECIN_BUFFERINSERT_ERROR -3003
#define RECIN_SYNC_RTP_CHANGED_CODEC -3004
#define RECIN_SYNC_RTP_NOT_ACCEPTABLE -3005
/* PBUFFER/BUFSTAT ERRORS */
#define PBUFFER_INIT_ERROR -4001

44
webrtc/modules/audio_coding/neteq/recin.c
View File

@ -17,13 +17,12 @@
#include <string.h>
#include "signal_processing_library.h"
#include "automode.h"
#include "dtmf_buffer.h"
#include "mcu_dsp_common.h"
#include "neteq_defines.h"
#include "neteq_error_codes.h"
#include "signal_processing_library.h"
int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
uint32_t uw32_timeRec)
@ -36,6 +35,8 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
int curr_Codec;
int16_t isREDPayload = 0;
int32_t temp_bufsize;
int is_sync_rtp = MCU_inst->av_sync && WebRtcNetEQ_IsSyncPayload(
RTPpacketInput->payload, RTPpacketInput->payloadLen);
#ifdef NETEQ_RED_CODEC
RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */
RTPpacketPtr[0] = &RTPpacket[0];
@ -77,15 +78,23 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
}
/* Call RTCP statistics */
i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber,
if (!is_sync_rtp) { /* Update only if it not sync packet. */
/* Call RTCP statistics if it is not sync packet. */
i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst),
RTPpacket[0].seqNumber,
RTPpacket[0].timeStamp, uw32_timeRec);
}
/* If Redundancy is supported and this is the redundancy payload, separate the payloads */
#ifdef NETEQ_RED_CODEC
if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
kDecoderRED))
{
if (is_sync_rtp)
{
/* Sync packet should not have RED payload type. */
return RECIN_SYNC_RTP_NOT_ACCEPTABLE;
}
/* Split the payload into a main and a redundancy payloads */
i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads);
@ -127,7 +136,7 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
/* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */
if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
kDecoderILBC))
kDecoderILBC) && !is_sync_rtp) /* Don't update if sync RTP. */
{
i_ok = WebRtcNetEQ_DbGetSplitInfo(
&MCU_inst->PayloadSplit_inst,
@ -175,6 +184,12 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
kDecoderAVT))
{
if (is_sync_rtp)
{
/* Sync RTP should not have AVT payload type. */
return RECIN_SYNC_RTP_NOT_ACCEPTABLE;
}
#ifdef NETEQ_ATEVENT_DECODE
if (MCU_inst->AVT_PlayoutOn)
{
@ -205,6 +220,11 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
/* Get CNG sample rate */
uint16_t fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
RTPpacket[i_k].payloadType);
if (is_sync_rtp)
{
/* Sync RTP should not have CNG payload type. */
return RECIN_SYNC_RTP_NOT_ACCEPTABLE;
}
/* Force sampling frequency to 32000 Hz CNG 48000 Hz. */
/* TODO(tlegrand): remove limitation once ACM has full 48 kHz
@ -245,6 +265,11 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
{
return RECIN_UNKNOWNPAYLOAD;
}
if (is_sync_rtp)
{
/* Sync RTP should not cause codec change. */
return RECIN_SYNC_RTP_CHANGED_CODEC;
}
MCU_inst->current_Codec = curr_Codec;
MCU_inst->current_Payload = RTPpacket[i_k].payloadType;
i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst,
@ -280,10 +305,11 @@ int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
}
/*
* Update Bandwidth Estimate
* Only send the main payload to BWE
* If not sync RTP, update Bandwidth Estimate.
* Only send the main payload to BWE.
*/
if ((curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
if (!is_sync_rtp &&
(curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
RTPpacket[0].payloadType)) >= 0)
{
codecPos = MCU_inst->codec_DB_inst.position[curr_Codec];

9
webrtc/modules/audio_coding/neteq/split_and_insert.c
View File

@ -16,9 +16,9 @@
#include <string.h>
#include "signal_processing_library.h"
#include "mcu_dsp_common.h"
#include "neteq_error_codes.h"
#include "signal_processing_library.h"
int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t* packet,
PacketBuf_t* Buffer_inst,
@ -33,6 +33,8 @@ int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t* packet,
RTPPacket_t temp_packet;
int16_t localFlushed = 0;
const int16_t *pw16_startPayload;
const int is_sync_rtp = av_sync &&
WebRtcNetEQ_IsSyncPayload(packet->payload, packet->payloadLen);
*flushed = 0;
len = packet->payloadLen;
@ -41,7 +43,8 @@ int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t* packet,
WEBRTC_SPL_MEMCPY_W8(&temp_packet,packet,sizeof(RTPPacket_t));
if (split_inst->deltaBytes == NO_SPLIT)
if (split_inst->deltaBytes == NO_SPLIT ||
is_sync_rtp) /* Don't split sync RTPs just insert. */
{
/* Not splittable codec */
i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, packet,