diff --git a/webrtc/modules/audio_coding/codecs/g711/g711_interface.c b/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
index 9ef7884c5..087e3e11c 100644
--- a/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
+++ b/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
@@ -31,7 +31,7 @@ int16_t WebRtcG711_EncodeA(void* state,
   for (n = 0; n < len; n++) {
     tempVal = (uint16_t) linear_to_alaw(speechIn[n]);
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     if ((n & 0x1) == 1) {
       encoded[n >> 1] |= ((uint16_t) tempVal);
     } else {
@@ -69,7 +69,7 @@ int16_t WebRtcG711_EncodeU(void* state,
   for (n = 0; n < len; n++) {
     tempVal = (uint16_t) linear_to_ulaw(speechIn[n]);
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     if ((n & 0x1) == 1) {
       encoded[n >> 1] |= ((uint16_t) tempVal);
     } else {
@@ -103,7 +103,7 @@ int16_t WebRtcG711_DecodeA(void* state,
   }
 
   for (n = 0; n < len; n++) {
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     if ((n & 0x1) == 1) {
       tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
     } else {
@@ -140,7 +140,7 @@ int16_t WebRtcG711_DecodeU(void* state,
   }
 
   for (n = 0; n < len; n++) {
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     if ((n & 0x1) == 1) {
       tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
     } else {
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
index 5da968543..febd4ceb0 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -28,7 +28,7 @@
 #include "decode_residual.h"
 #include "unpack_bits.h"
 #include "hp_output.h"
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
 #include "swap_bytes.h"
 #endif
 
@@ -54,7 +54,7 @@ void WebRtcIlbcfix_DecodeImpl(
   int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
   int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
   int16_t PLClpc[LPC_FILTERORDER + 1];
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   uint16_t swapped[NO_OF_WORDS_30MS];
 #endif
   iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
@@ -68,7 +68,7 @@ void WebRtcIlbcfix_DecodeImpl(
 
     /* Unpacketize bits into parameters */
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
     WebRtcIlbcfix_SwapBytes(bytes, iLBCdec_inst->no_of_words, swapped);
     last_bit = WebRtcIlbcfix_UnpackBits(swapped, iLBCbits_inst, iLBCdec_inst->mode);
 #else
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
index 75d1672b8..2f899a53b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -32,7 +32,7 @@
 #include "unpack_bits.h"
 #include "index_conv_dec.h"
 #endif
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
 #include "swap_bytes.h"
 #endif
 
@@ -489,7 +489,7 @@ void WebRtcIlbcfix_EncodeImpl(
   WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
 #endif
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   /* Swap bytes for LITTLE ENDIAN since the packbits()
      function assumes BIG_ENDIAN machine */
 #ifdef SPLIT_10MS
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 945475f80..8baa30738 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -327,7 +327,7 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
 {
   ISACFIX_SubStruct *ISAC_inst;
   int16_t stream_len;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
 
@@ -352,7 +352,7 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
 
 
   /* convert from bytes to int16_t */
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
     encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
                                   | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
@@ -442,7 +442,7 @@ int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
 
 
   /* convert from bytes to int16_t */
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
     encoded[k] = (int16_t)(((uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8)
                                  | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
@@ -485,7 +485,7 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
 {
   ISACFIX_SubStruct *ISAC_inst;
   int16_t stream_len;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
 
@@ -507,7 +507,7 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
     return -1;
   }
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0;k<(stream_len+1)>>1;k++) {
     encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
                                   | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
@@ -588,7 +588,7 @@ int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
   uint16_t partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -621,7 +621,7 @@ int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
   streamdata.stream_index = 0;
   streamdata.full = 1;
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<5; k++) {
     streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
@@ -676,7 +676,7 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
   ISACFIX_SubStruct *ISAC_inst;
   Bitstr_dec streamdata;
   uint16_t partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -709,7 +709,7 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
   streamdata.stream_index = 0;
   streamdata.full = 1;
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<5; k++) {
     streamdata.stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
@@ -765,7 +765,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
   /* number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
   int16_t     number_of_samples;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t declen = 0;
@@ -793,7 +793,7 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
   (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded;
 
   /* convert bitstream from int16_t to bytes */
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<(len>>1); k++) {
     (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
@@ -868,7 +868,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
   /* twice the number of samples (480 or 960), output from decoder */
   /* that were actually used in the encoder/decoder (determined on the fly) */
   int16_t     number_of_samples;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t declen = 0;
@@ -894,7 +894,7 @@ int16_t WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
   (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (uint16_t *)encoded;
 
   /* convert bitstream from int16_t to bytes */
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<(len>>1); k++) {
     (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
@@ -1267,7 +1267,7 @@ int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
 {
   Bitstr_dec streamdata;
   uint16_t partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -1280,7 +1280,7 @@ int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
   streamdata.stream_index = 0;
   streamdata.full = 1;
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<5; k++) {
     streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
@@ -1316,7 +1316,7 @@ int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
 {
   Bitstr_dec streamdata;
   uint16_t partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -1329,7 +1329,7 @@ int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
   streamdata.stream_index = 0;
   streamdata.full = 1;
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k=0; k<5; k++) {
     streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
   }
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index 1e9027200..f3f1650b4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -742,7 +742,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
 
     WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
                       streamLenUB + garbageLen, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
     for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
       ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
         (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
@@ -805,7 +805,7 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
   int32_t currentBN;
   uint8_t* encodedPtrUW8 = (uint8_t*)encoded;
   uint32_t crc;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int16_t  k;
 #endif
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
@@ -896,7 +896,7 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct*  ISAC_main_inst,
 
   WebRtcIsac_GetCrc((int16_t*)(&(encodedPtrUW8[streamLenLB + 1])),
                     streamLenUB, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
     encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
       (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
@@ -1008,7 +1008,7 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
                                     uint32_t arr_ts) {
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
   Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -1029,7 +1029,7 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
 
   WebRtcIsac_ResetBitstream(&(streamdata));
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
     streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
                                             ((k & 1) << 3)) & 0xFF);
@@ -1741,14 +1741,14 @@ int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst,
 int16_t WebRtcIsac_ReadBwIndex(const int16_t* encoded,
                                int16_t* bweIndex) {
   Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
 
   WebRtcIsac_ResetBitstream(&(streamdata));
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
     streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
         ((k & 1) << 3)) & 0xFF);
@@ -1790,7 +1790,7 @@ int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
                                 const int16_t* encoded,
                                 int16_t* frameLength) {
   Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
   int16_t err;
@@ -1798,7 +1798,7 @@ int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
 
   WebRtcIsac_ResetBitstream(&(streamdata));
 
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   for (k = 0; k < 10; k++) {
     streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
                                             ((k & 1) << 3)) & 0xFF);
@@ -2108,7 +2108,7 @@ int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
   int16_t totalLenUB;
   uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
   ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
   int k;
 #endif
 
@@ -2164,7 +2164,7 @@ int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
 
     WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
                       streamLenUB, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
     for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
       ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
         (uint8_t)((crc >> (24 - k * 8)) & 0xFF);
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
index 04814b767..34aadc3f2 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -15,7 +15,7 @@
 
 #include "typedefs.h"
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
 #include "signal_processing_library.h"
 #endif
 
@@ -29,7 +29,7 @@ int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
                                int16_t len,
                                int16_t *speechOut16b)
 {
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);
 #else
     int i;
@@ -68,7 +68,7 @@ int16_t WebRtcPcm16b_DecodeW16(void *inst,
                                int16_t *speechOut16b,
                                int16_t* speechType)
 {
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(int16_t)+1)>>1));
 #else
     int i;
diff --git a/webrtc/modules/audio_coding/neteq/dtmf_buffer.c b/webrtc/modules/audio_coding/neteq/dtmf_buffer.c
index 9e3212646..1788635c7 100644
--- a/webrtc/modules/audio_coding/neteq/dtmf_buffer.c
+++ b/webrtc/modules/audio_coding/neteq/dtmf_buffer.c
@@ -93,7 +93,7 @@ int16_t WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
     if (len == 4)
     {
         EventStart = encoded;
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
         value=((*EventStart)>>8);
         endEvent=((*EventStart)&0x80)>>7;
         Volume=((*EventStart)&0x3F);
diff --git a/webrtc/modules/audio_coding/neteq/rtp.c b/webrtc/modules/audio_coding/neteq/rtp.c
index f23f3512b..6ab5944b5 100644
--- a/webrtc/modules/audio_coding/neteq/rtp.c
+++ b/webrtc/modules/audio_coding/neteq/rtp.c
@@ -31,7 +31,7 @@ int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen,
         return RTP_TOO_SHORT_PACKET;
     }
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */
     i_P = (((uint16_t) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */
     i_X = (((uint16_t) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */
@@ -62,7 +62,7 @@ int WebRtcNetEQ_RTPPayloadInfo(int16_t* pw16_Datagram, int i_DatagramLen,
             i_padlength = ((pw16_Datagram[(i_DatagramLen >> 1) - 1]) & 0xFF);
         }
     }
-#else /* WEBRTC_LITTLE_ENDIAN */
+#else /* WEBRTC_ARCH_LITTLE_ENDIAN */
     i_IPver = (((uint16_t) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */
     i_P = (((uint16_t) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */
     i_X = (((uint16_t) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */
@@ -126,7 +126,7 @@ int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
     int i_discardedBlockLength = 0;
     int singlePayload = 0;
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     if ((pw16_data[0] & 0x8000) == 0)
     {
         /* Only one payload in this packet*/
@@ -155,7 +155,7 @@ int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
         ((((uint16_t)pw16_data[1]) & 0xFC00) >> 10);
         i_blockLength = (((uint16_t)pw16_data[1]) & 0x3FF);
     }
-#else /* WEBRTC_LITTLE_ENDIAN */
+#else /* WEBRTC_ARCH_LITTLE_ENDIAN */
     if ((pw16_data[0] & 0x80) == 0)
     {
         /* Only one payload in this packet */
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index b3d87c67e..caa1efed8 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -1177,7 +1177,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayout()
 
     _playoutFramesIn10MS = _playoutFreq/100;
     if ((errVal = LATE(snd_pcm_set_params)( _handlePlayout,
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
         SND_PCM_FORMAT_S16_BE,
 #else
         SND_PCM_FORMAT_S16_LE, //format
@@ -1333,7 +1333,7 @@ int32_t AudioDeviceLinuxALSA::InitRecording()
 
     _recordingFramesIn10MS = _recordingFreq/100;
     if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
         SND_PCM_FORMAT_S16_BE, //format
 #else
         SND_PCM_FORMAT_S16_LE, //format
@@ -1352,7 +1352,7 @@ int32_t AudioDeviceLinuxALSA::InitRecording()
            _recChannels = 1;
 
          if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
              SND_PCM_FORMAT_S16_BE, //format
 #else
              SND_PCM_FORMAT_S16_LE, //format
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.cc b/webrtc/modules/audio_device/mac/audio_device_mac.cc
index 9da188013..b07c94dd1 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.cc
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.cc
@@ -97,7 +97,7 @@ void AudioDeviceMac::logCAMsg(const TraceLevel level,
     assert(msg != NULL);
     assert(err != NULL);
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
     // We need to flip the characters in this case.
@@ -1457,7 +1457,7 @@ int32_t AudioDeviceMac::InitPlayout()
 
     _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
         | kLinearPCMFormatFlagIsPacked;
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
     _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
@@ -1681,7 +1681,7 @@ int32_t AudioDeviceMac::InitRecording()
 
     _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
         | kLinearPCMFormatFlagIsPacked;
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
     _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
diff --git a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
index 08e419750..952dc11d8 100644
--- a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
+++ b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
@@ -1154,7 +1154,7 @@ void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
     assert(msg != NULL);
     assert(err != NULL);
 
-#ifdef WEBRTC_BIG_ENDIAN
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
     WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
     // We need to flip the characters in this case.
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index edf20bc2f..b33049e48 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -638,7 +638,7 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
   if (size <= 0) {
     return kUnspecifiedError;
   }
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
   // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
   //            pretty safe in assuming little-endian.
 #endif
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index bc0be8b34..f50b20a44 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -125,16 +125,12 @@ bool StringCompare(const char* str1, const char* str2,
 }
 #endif
 
-#if !defined(WEBRTC_LITTLE_ENDIAN) && !defined(WEBRTC_BIG_ENDIAN)
-#error Either WEBRTC_LITTLE_ENDIAN or WEBRTC_BIG_ENDIAN must be defined
-#endif
-
 /* for RTP/RTCP
     All integer fields are carried in network byte order, that is, most
     significant byte (octet) first.  AKA big-endian.
 */
 void AssignUWord32ToBuffer(uint8_t* dataBuffer, uint32_t value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
   dataBuffer[0] = static_cast<uint8_t>(value >> 24);
   dataBuffer[1] = static_cast<uint8_t>(value >> 16);
   dataBuffer[2] = static_cast<uint8_t>(value >> 8);
@@ -146,7 +142,7 @@ void AssignUWord32ToBuffer(uint8_t* dataBuffer, uint32_t value) {
 }
 
 void AssignUWord24ToBuffer(uint8_t* dataBuffer, uint32_t value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
   dataBuffer[0] = static_cast<uint8_t>(value >> 16);
   dataBuffer[1] = static_cast<uint8_t>(value >> 8);
   dataBuffer[2] = static_cast<uint8_t>(value);
@@ -158,7 +154,7 @@ void AssignUWord24ToBuffer(uint8_t* dataBuffer, uint32_t value) {
 }
 
 void AssignUWord16ToBuffer(uint8_t* dataBuffer, uint16_t value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
   dataBuffer[0] = static_cast<uint8_t>(value >> 8);
   dataBuffer[1] = static_cast<uint8_t>(value);
 #else
@@ -168,7 +164,7 @@ void AssignUWord16ToBuffer(uint8_t* dataBuffer, uint16_t value) {
 }
 
 uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
   return (dataBuffer[0] << 8) + dataBuffer[1];
 #else
   return *reinterpret_cast<const uint16_t*>(dataBuffer);
@@ -180,7 +176,7 @@ uint32_t BufferToUWord24(const uint8_t* dataBuffer) {
 }
 
 uint32_t BufferToUWord32(const uint8_t* dataBuffer) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
   return (dataBuffer[0] << 24) + (dataBuffer[1] << 16) + (dataBuffer[2] << 8) +
       dataBuffer[3];
 #else
diff --git a/webrtc/modules/utility/source/rtp_dump_impl.cc b/webrtc/modules/utility/source/rtp_dump_impl.cc
index 1f8715dfa..39316f478 100644
--- a/webrtc/modules/utility/source/rtp_dump_impl.cc
+++ b/webrtc/modules/utility/source/rtp_dump_impl.cc
@@ -245,37 +245,25 @@ inline uint32_t RtpDumpImpl::GetTimeInMS() const
     gettimeofday(&tv, &tz);
     val = tv.tv_sec * 1000 + tv.tv_usec / 1000;
     return val;
-#else
-    #error Either _WIN32 or LINUX or WEBRTC_MAC has to be defined!
-    assert(false);
-    return 0;
 #endif
 }
 
 inline uint32_t RtpDumpImpl::RtpDumpHtonl(uint32_t x) const
 {
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
     return x;
-#elif defined(WEBRTC_LITTLE_ENDIAN)
+#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
     return (x >> 24) + ((((x >> 16) & 0xFF) << 8) + ((((x >> 8) & 0xFF) << 16) +
                                                      ((x & 0xFF) << 24)));
-#else
-#error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
-    assert(false);
-    return 0;
 #endif
 }
 
 inline uint16_t RtpDumpImpl::RtpDumpHtons(uint16_t x) const
 {
-#if defined(WEBRTC_BIG_ENDIAN)
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
     return x;
-#elif defined(WEBRTC_LITTLE_ENDIAN)
+#elif defined(WEBRTC_ARCH_LITTLE_ENDIAN)
     return (x >> 8) + ((x & 0xFF) << 8);
-#else
-    #error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
-    assert(false);
-    return 0;
 #endif
 }
 }  // namespace webrtc
diff --git a/webrtc/typedefs.h b/webrtc/typedefs.h
index 37c8fc9ce..e15bb1b3d 100644
--- a/webrtc/typedefs.h
+++ b/webrtc/typedefs.h
@@ -25,38 +25,37 @@
 //   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
 //   http://www.agner.org/optimize/calling_conventions.pdf
 //   or with gcc, run: "echo | gcc -E -dM -"
-// TODO(andrew): replace WEBRTC_LITTLE_ENDIAN with WEBRTC_ARCH_LITTLE_ENDIAN.
 #if defined(_M_X64) || defined(__x86_64__)
 #define WEBRTC_ARCH_X86_FAMILY
 #define WEBRTC_ARCH_X86_64
 #define WEBRTC_ARCH_64_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
-#define WEBRTC_LITTLE_ENDIAN
 #elif defined(_M_IX86) || defined(__i386__)
 #define WEBRTC_ARCH_X86_FAMILY
 #define WEBRTC_ARCH_X86
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
-#define WEBRTC_LITTLE_ENDIAN
 #elif defined(__ARMEL__)
-// TODO(andrew): We'd prefer to control platform defines here, but this is
+// TODO(ajm): We'd prefer to control platform defines here, but this is
 // currently provided by the Android makefiles. Commented to avoid duplicate
 // definition warnings.
 //#define WEBRTC_ARCH_ARM
-// TODO(andrew): Chromium uses the following two defines. Should we switch?
+// TODO(ajm): Chromium uses the following two defines. Should we switch?
 //#define WEBRTC_ARCH_ARM_FAMILY
 //#define WEBRTC_ARCH_ARMEL
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
-#define WEBRTC_LITTLE_ENDIAN
 #elif defined(__MIPSEL__)
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
-#define WEBRTC_LITTLE_ENDIAN
 #else
 #error Please add support for your architecture in typedefs.h
 #endif
 
+#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
+#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
+#endif
+
 #if defined(__SSE2__) || defined(_MSC_VER)
 #define WEBRTC_USE_SSE2
 #endif