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

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

webrtc/modules/audio_coding/neteq/neteq_error_codes.h

@@ -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

webrtc/modules/audio_coding/neteq/recin.c

@@ -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 */
+    if (!is_sync_rtp) { /* Update only if it not sync packet. */
-    i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber,
+      /* Call RTCP statistics if it is not sync packet. */
-        RTPpacket[0].timeStamp, uw32_timeRec);
+      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
+     * If not sync RTP, update Bandwidth Estimate.
-     * Only send the main payload to BWE
+     * 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];

webrtc/modules/audio_coding/neteq/split_and_insert.c

@@ -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,