diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
index ab4003abb..64e323c0e 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amr.cc
@@ -43,6 +43,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_AMR
 ACMAMR::ACMAMR(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -307,4 +309,6 @@ ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
 }
 
 #endif
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amr.h b/webrtc/modules/audio_coding/main/acm2/acm_amr.h
index 4471e6bca..efa568740 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amr.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amr.h
@@ -21,6 +21,8 @@ namespace webrtc {
 
 enum ACMAMRPackingFormat;
 
+namespace acm2 {
+
 class ACMAMR : public ACMGenericCodec {
  public:
   explicit ACMAMR(int16_t codec_id);
@@ -60,6 +62,8 @@ class ACMAMR : public ACMGenericCodec {
   ACMAMRPackingFormat encoder_packing_format_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMR_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
index 849353a93..4dd4e97b2 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc
@@ -40,6 +40,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_AMRWB
 ACMAMRwb::ACMAMRwb(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -313,4 +315,6 @@ ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
index e5bd99d9b..c8db65933 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_amrwb.h
@@ -19,6 +19,8 @@ struct AMRWB_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMAMRwb : public ACMGenericCodec {
  public:
   explicit ACMAMRwb(int16_t codec_id);
@@ -61,6 +63,8 @@ class ACMAMRwb : public ACMGenericCodec {
   ACMAMRPackingFormat encoder_packing_format_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_AMRWB_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_celt.cc b/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
index 21fa3a9d0..3cd6e84d4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_celt.cc
@@ -20,6 +20,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_CELT
 
 ACMCELT::ACMCELT(int16_t /* codec_id */)
@@ -188,4 +190,6 @@ int16_t ACMCELT::SetBitRateSafe(const int32_t rate) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_celt.h b/webrtc/modules/audio_coding/main/acm2/acm_celt.h
index 4b40f799e..dc8d3ee0f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_celt.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_celt.h
@@ -19,6 +19,8 @@ struct CELT_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMCELT : public ACMGenericCodec {
  public:
   explicit ACMCELT(int16_t codec_id);
@@ -45,6 +47,8 @@ class ACMCELT : public ACMGenericCodec {
   uint16_t channels_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CELT_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_cng.cc b/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
index 9e658bdad..7f0adf819 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_cng.cc
@@ -17,6 +17,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 ACMCNG::ACMCNG(int16_t codec_id) {
   encoder_inst_ptr_ = NULL;
   codec_id_ = codec_id;
@@ -76,4 +78,6 @@ void ACMCNG::InternalDestructEncoderInst(void* ptr_inst) {
   return;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_cng.h b/webrtc/modules/audio_coding/main/acm2/acm_cng.h
index 3816fa2a8..a0c1c5544 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_cng.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_cng.h
@@ -19,6 +19,8 @@ struct WebRtcCngDecInst;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMCNG: public ACMGenericCodec {
  public:
   explicit ACMCNG(int16_t codec_id);
@@ -51,6 +53,8 @@ class ACMCNG: public ACMGenericCodec {
   uint16_t samp_freq_hz_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CNG_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
index 225098aad..fd30a137a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
@@ -102,6 +102,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 // Not yet used payload-types.
 // 83,  82,  81, 80, 79,  78,  77,  76,  75,  74,  73,  72,  71,  70,  69, 68,
 // 67, 66, 65
@@ -954,4 +956,6 @@ bool ACMCodecDB::OwnsDecoder(int codec_id) {
   return ACMCodecDB::codec_settings_[codec_id].owns_decoder;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
index b992b7d27..98869efee 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
@@ -22,6 +22,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 // TODO(tlegrand): replace class ACMCodecDB with a namespace.
 class ACMCodecDB {
  public:
@@ -350,6 +352,8 @@ class ACMCodecDB {
   static const NetEqDecoder neteq_decoders_[kMaxNumCodecs];
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
index ca7e86fd8..07bab6520 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc
@@ -18,6 +18,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_AVT
 
 ACMDTMFPlayout::ACMDTMFPlayout(int16_t /* codec_id */) { return; }
@@ -83,4 +85,6 @@ void ACMDTMFPlayout::DestructEncoderSafe() {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
index 4c3154ca9..5adb0ddc1 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMDTMFPlayout : public ACMGenericCodec {
  public:
   explicit ACMDTMFPlayout(int16_t codec_id);
@@ -35,6 +37,8 @@ class ACMDTMFPlayout : public ACMGenericCodec {
   void InternalDestructEncoderInst(void* ptr_inst);
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DTMF_PLAYOUT_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g722.cc b/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
index fe2bd6cb9..8d457723b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g722.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_G722
 
 ACMG722::ACMG722(int16_t /* codec_id */)
@@ -194,4 +196,6 @@ void ACMG722::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g722.h b/webrtc/modules/audio_coding/main/acm2/acm_g722.h
index 34b6c8516..7216a574a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g722.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g722.h
@@ -18,6 +18,8 @@ typedef struct WebRtcG722DecInst G722DecInst;
 
 namespace webrtc {
 
+namespace acm2 {
+
 // Forward declaration.
 struct ACMG722EncStr;
 struct ACMG722DecStr;
@@ -52,6 +54,8 @@ class ACMG722 : public ACMGenericCodec {
   G722EncInst* encoder_inst_ptr_right_;  // Prepared for stereo
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
index 0cba71084..3609f072b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc
@@ -80,6 +80,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_G722_1
 
 ACMG722_1::ACMG722_1(int16_t /* codec_id */)
@@ -323,4 +325,6 @@ void ACMG722_1::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221.h b/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
index 4a0bd480d..74d9b2709 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221.h
@@ -24,6 +24,8 @@ struct G722_1_Inst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMG722_1 : public ACMGenericCodec {
  public:
   explicit ACMG722_1(int16_t codec_id);
@@ -57,6 +59,8 @@ class ACMG722_1 : public ACMGenericCodec {
   G722_1_32_encinst_t_* encoder_inst32_ptr_right_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
index 531008af2..f6555f5db 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc
@@ -80,6 +80,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_G722_1C
 
 ACMG722_1C::ACMG722_1C(int16_t /* codec_id */)
@@ -329,4 +331,6 @@ void ACMG722_1C::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
index 961ed4e17..66b0f54a3 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7221c.h
@@ -24,6 +24,8 @@ struct G722_1_Inst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMG722_1C : public ACMGenericCodec {
  public:
   explicit ACMG722_1C(int16_t codec_id);
@@ -57,6 +59,8 @@ class ACMG722_1C : public ACMGenericCodec {
   G722_1C_48_encinst_t_* encoder_inst48_ptr_right_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7221C_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g729.cc b/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
index 91dbb43ee..6f95f3685 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g729.cc
@@ -22,6 +22,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_G729
 
 ACMG729::ACMG729(int16_t /* codec_id */) : encoder_inst_ptr_(NULL) {}
@@ -252,4 +254,6 @@ void ACMG729::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g729.h b/webrtc/modules/audio_coding/main/acm2/acm_g729.h
index f7e762cba..88b1bf7c4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g729.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g729.h
@@ -19,6 +19,8 @@ struct G729_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMG729 : public ACMGenericCodec {
  public:
   explicit ACMG729(int16_t codec_id);
@@ -49,6 +51,8 @@ class ACMG729 : public ACMGenericCodec {
   G729_encinst_t_* encoder_inst_ptr_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G729_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc b/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
index f16eec89b..3cf9f22fc 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_G729_1
 
 ACMG729_1::ACMG729_1(int16_t /* codec_id */)
@@ -237,4 +239,6 @@ int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_g7291.h b/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
index 5a38e59a3..38c8fe176 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_g7291.h
@@ -19,6 +19,8 @@ struct G729_1_inst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMG729_1 : public ACMGenericCodec {
  public:
   explicit ACMG729_1(int16_t codec_id);
@@ -47,6 +49,8 @@ class ACMG729_1 : public ACMGenericCodec {
   int16_t flag_g729_mode_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G7291_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
index 9754251a5..aa8e8be06 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 // Enum for CNG
 enum {
   kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
@@ -1002,4 +1004,6 @@ int16_t ACMGenericCodec::REDPayloadISAC(const int32_t /* isac_rate */,
   return -1;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
index 78feb99f0..d41580fff 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h
@@ -26,8 +26,12 @@ struct WebRtcCngEncInst;
 
 namespace webrtc {
 
-// forward declaration
+struct WebRtcACMCodecParams;
 struct CodecInst;
+
+namespace acm2 {
+
+// forward declaration
 class AcmReceiver;
 
 class ACMGenericCodec {
@@ -909,6 +913,8 @@ class ACMGenericCodec {
   uint32_t unique_id_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
index 44e6e3d91..90f9fce7d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_GSMFR
 
 ACMGSMFR::ACMGSMFR(int16_t /* codec_id */) : encoder_inst_ptr_(NULL) {}
@@ -154,4 +156,6 @@ void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
index 51c29eea4..79c3a180b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h
@@ -19,6 +19,8 @@ struct GSMFR_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMGSMFR : public ACMGenericCodec {
  public:
   explicit ACMGSMFR(int16_t codec_id);
@@ -45,6 +47,8 @@ class ACMGSMFR : public ACMGenericCodec {
   GSMFR_encinst_t_* encoder_inst_ptr_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
index 14fbbd450..eaa079c46 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc
@@ -17,6 +17,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_ILBC
 
 ACMILBC::ACMILBC(int16_t /* codec_id */) : encoder_inst_ptr_(NULL) {}
@@ -138,4 +140,6 @@ int16_t ACMILBC::SetBitRateSafe(const int32_t rate) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
index e02c789d3..fd6e85379 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_ilbc.h
@@ -19,6 +19,8 @@ struct iLBC_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMILBC : public ACMGenericCodec {
  public:
   explicit ACMILBC(int16_t codec_id);
@@ -43,6 +45,8 @@ class ACMILBC : public ACMGenericCodec {
   iLBC_encinst_t_* encoder_inst_ptr_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ILBC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index 7957fd3dc..e27284212 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -31,6 +31,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 // we need this otherwise we cannot use forward declaration
 // in the header file
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
@@ -826,4 +828,6 @@ AudioDecoder* ACMISAC::Decoder(int codec_id) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.h b/webrtc/modules/audio_coding/main/acm2/acm_isac.h
index 2e6657fb4..a3227d5d0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 struct ACMISACInst;
 class AcmAudioDecoderIsac;
 
@@ -93,6 +95,8 @@ class ACMISAC : public ACMGenericCodec {
   bool decoder_initialized_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
index c2a782095..df0309769 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifdef WEBRTC_CODEC_ISAC
 #define ACM_ISAC_CREATE            WebRtcIsac_Create
 #define ACM_ISAC_FREE              WebRtcIsac_Free
@@ -70,6 +72,8 @@ namespace webrtc {
                                                          // decoder
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_MACROS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc
index 3f5a200bf..607b933de 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc
@@ -10,4 +10,6 @@
 
 // This file contains unit tests for ACM's NetEQ wrapper (class ACMNetEQ).
 
-namespace webrtc {}  // namespace
+namespace webrtc {
+
+namespace acm2 {}  // namespace
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.cc b/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
index d627fad8d..c00a9203a 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_OPUS
 
 ACMOpus::ACMOpus(int16_t /* codec_id */)
@@ -184,4 +186,6 @@ int16_t ACMOpus::SetBitRateSafe(const int32_t rate) {
 
 #endif  // WEBRTC_CODEC_OPUS
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_opus.h b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
index caac01093..a346e3c8f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_opus.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_opus.h
@@ -19,6 +19,8 @@ struct WebRtcOpusDecInst;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMOpus : public ACMGenericCodec {
  public:
   explicit ACMOpus(int16_t codec_id);
@@ -45,6 +47,8 @@ class ACMOpus : public ACMGenericCodec {
   int channels_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
index 7c5b0bd32..89688a874 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_PCM16
 
 ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) { return; }
@@ -89,4 +91,6 @@ void ACMPCM16B::DestructEncoderSafe() {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
index 32490209a..23b8c121e 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMPCM16B : public ACMGenericCodec {
  public:
   explicit ACMPCM16B(int16_t codec_id);
@@ -37,6 +39,8 @@ class ACMPCM16B : public ACMGenericCodec {
   int32_t sampling_freq_hz_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCM16B_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
index cb5ebccfd..1dd6e48fb 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc
@@ -18,6 +18,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 ACMPCMA::ACMPCMA(int16_t codec_id) { codec_id_ = codec_id; }
 
 ACMPCMA::~ACMPCMA() { return; }
@@ -55,4 +57,6 @@ void ACMPCMA::DestructEncoderSafe() {
   return;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcma.h b/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
index 4102e17d9..2da873cb7 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcma.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMPCMA : public ACMGenericCodec {
  public:
   explicit ACMPCMA(int16_t codec_id);
@@ -35,6 +37,8 @@ class ACMPCMA : public ACMGenericCodec {
   void InternalDestructEncoderInst(void* ptr_inst);
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMA_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
index 6f479ed21..7d2536fc0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc
@@ -18,6 +18,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 ACMPCMU::ACMPCMU(int16_t codec_id) { codec_id_ = codec_id; }
 
 ACMPCMU::~ACMPCMU() {}
@@ -56,4 +58,6 @@ void ACMPCMU::DestructEncoderSafe() {
   encoder_initialized_ = false;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
index 2898df637..18d8279dc 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_pcmu.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMPCMU : public ACMGenericCodec {
  public:
   explicit ACMPCMU(int16_t codec_id);
@@ -35,6 +37,8 @@ class ACMPCMU : public ACMGenericCodec {
   void InternalDestructEncoderInst(void* ptr_inst);
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_PCMU_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 9c9f05d9b..6d4a8dfe0 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -30,6 +30,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 namespace {
 
 const int kNeteqInitSampleRateHz = 16000;
@@ -829,4 +831,6 @@ void AcmReceiver::InsertStreamOfSyncPackets(
   }
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 5f6d684b0..9267c1e68 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -26,11 +26,14 @@
 
 namespace webrtc {
 
+struct CodecInst;
 class CriticalSectionWrapper;
 class RWLockWrapper;
 class NetEq;
+
+namespace acm2 {
+
 class Nack;
-struct CodecInst;
 
 class AcmReceiver {
  public:
@@ -360,6 +363,8 @@ class AcmReceiver {
   scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index a00db9278..7d85d15d8 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -23,6 +23,8 @@
 #include "webrtc/test/testsupport/gtest_disable.h"
 
 namespace webrtc {
+
+namespace acm2 {
 namespace {
 
 bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) {
@@ -418,4 +420,6 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) {
   }
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_red.cc b/webrtc/modules/audio_coding/main/acm2/acm_red.cc
index f4a1f6f2a..4f2c70cfd 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_red.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_red.cc
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 ACMRED::ACMRED(int16_t codec_id) { codec_id_ = codec_id; }
 
 ACMRED::~ACMRED() {}
@@ -47,4 +49,6 @@ void ACMRED::DestructEncoderSafe() {
   // RED has no instance
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_red.h b/webrtc/modules/audio_coding/main/acm2/acm_red.h
index ab8d913fa..ac381709d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_red.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_red.h
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMRED : public ACMGenericCodec {
  public:
   explicit ACMRED(int16_t codec_id);
@@ -35,6 +37,8 @@ class ACMRED : public ACMGenericCodec {
   void InternalDestructEncoderInst(void* ptr_inst);
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RED_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
index 13eed0ba6..3abe4f1ec 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 ACMResampler::ACMResampler()
     : resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
 }
@@ -63,4 +65,6 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
   return out_len / num_audio_channels;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
index 8abb2f4f7..e992955f5 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.h
@@ -18,6 +18,8 @@ namespace webrtc {
 
 class CriticalSectionWrapper;
 
+namespace acm2 {
+
 class ACMResampler {
  public:
   ACMResampler();
@@ -35,6 +37,8 @@ class ACMResampler {
   CriticalSectionWrapper* resampler_crit_sect_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_speex.cc b/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
index 829026549..84a0592a4 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_speex.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 #ifndef WEBRTC_CODEC_SPEEX
 ACMSPEEX::ACMSPEEX(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
@@ -326,4 +328,6 @@ int16_t ACMSPEEX::SetComplMode(int16_t mode) {
 
 #endif
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_speex.h b/webrtc/modules/audio_coding/main/acm2/acm_speex.h
index 2fac8fd2e..f9cf78706 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_speex.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_speex.h
@@ -19,6 +19,8 @@ struct SPEEX_decinst_t_;
 
 namespace webrtc {
 
+namespace acm2 {
+
 class ACMSPEEX : public ACMGenericCodec {
  public:
   explicit ACMSPEEX(int16_t codec_id);
@@ -60,6 +62,8 @@ class ACMSPEEX : public ACMGenericCodec {
   uint16_t samples_in_20ms_audio_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SPEEX_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index 7acd49bdf..c7fe9fa9a 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -30,13 +30,13 @@ AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
 
 // Get number of supported codecs
 int AudioCodingModule::NumberOfCodecs() {
-  return ACMCodecDB::kNumCodecs;
+  return acm2::ACMCodecDB::kNumCodecs;
 }
 
 // Get supported codec parameters with id
 int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
   // Get the codec settings for the codec with the given list ID
-  return ACMCodecDB::Codec(list_id, codec);
+  return acm2::ACMCodecDB::Codec(list_id, codec);
 }
 
 // Get supported codec parameters with name, frequency and number of channels.
@@ -47,7 +47,8 @@ int AudioCodingModule::Codec(const char* payload_name,
   int codec_id;
 
   // Get the id of the codec from the database.
-  codec_id = ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+  codec_id = acm2::ACMCodecDB::CodecId(
+      payload_name, sampling_freq_hz, channels);
   if (codec_id < 0) {
     // We couldn't find a matching codec, set the parameters to unacceptable
     // values and return.
@@ -60,7 +61,7 @@ int AudioCodingModule::Codec(const char* payload_name,
   }
 
   // Get default codec settings.
-  ACMCodecDB::Codec(codec_id, codec);
+  acm2::ACMCodecDB::Codec(codec_id, codec);
 
   // Keep the number of channels from the function call. For most codecs it
   // will be the same value as in default codec settings, but not for all.
@@ -73,14 +74,14 @@ int AudioCodingModule::Codec(const char* payload_name,
 int AudioCodingModule::Codec(const char* payload_name,
                              int sampling_freq_hz,
                              int channels) {
-  return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
+  return acm2::ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
 }
 
 // Checks the validity of the parameters of the given codec
 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   int mirror_id;
 
-  int codec_number = ACMCodecDB::CodecNumber(codec, &mirror_id);
+  int codec_number = acm2::ACMCodecDB::CodecNumber(codec, &mirror_id);
 
   if (codec_number < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
@@ -97,7 +98,7 @@ AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
 }
 
 AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
-  return new AudioCodingModuleImpl(id);
+  return new acm2::AudioCodingModuleImpl(id);
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index fb6fe39a8..911605e8e 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -27,6 +27,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 enum {
   kACMToneEnd = 999
 };
@@ -1974,4 +1976,6 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
   return receiver_.LeastRequiredDelayMs();
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 435c7aeab..6bf90cf26 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -22,11 +22,14 @@
 
 namespace webrtc {
 
-class ACMDTMFDetection;
-class ACMGenericCodec;
 class CriticalSectionWrapper;
 class RWLockWrapper;
 
+namespace acm2 {
+
+class ACMDTMFDetection;
+class ACMGenericCodec;
+
 class AudioCodingModuleImpl : public AudioCodingModule {
  public:
   // Constructor
@@ -349,6 +352,8 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   bool first_10ms_data_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
index 0dccb1129..c2b218cb6 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
@@ -12,6 +12,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 InitialDelayManager::InitialDelayManager(int initial_delay_ms,
                                          int late_packet_threshold)
     : last_packet_type_(kUndefinedPacket),
@@ -227,4 +229,6 @@ void InitialDelayManager::UpdatePlayoutTimestamp(
       initial_delay_ms_ * sample_rate_hz / 1000);
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
index da08f8bd8..3c5ba3c01 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
@@ -16,6 +16,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 class InitialDelayManager {
  public:
   enum PacketType {
@@ -110,6 +112,8 @@ class InitialDelayManager {
   const int late_packet_threshold_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
index 7e3bda5b5..1e129f37e 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
@@ -15,6 +15,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 namespace {
 
 const uint8_t kAudioPayloadType = 0;
@@ -368,4 +370,6 @@ TEST_F(InitialDelayManagerTest, BufferingAudio) {
   EXPECT_FALSE(manager_->buffering());
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.cc b/webrtc/modules/audio_coding/main/acm2/nack.cc
index e26ad611f..7265fe63c 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack.cc
@@ -19,6 +19,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 namespace {
 
 const int kDefaultSampleRateKhz = 48;
@@ -222,4 +224,6 @@ std::vector<uint16_t> Nack::GetNackList(int round_trip_time_ms) const {
   return sequence_numbers;
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.h b/webrtc/modules/audio_coding/main/acm2/nack.h
index 490c03818..380932719 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.h
+++ b/webrtc/modules/audio_coding/main/acm2/nack.h
@@ -49,6 +49,8 @@
 //
 namespace webrtc {
 
+namespace acm2 {
+
 class Nack {
  public:
   // A limit for the size of the NACK list.
@@ -204,6 +206,8 @@ class Nack {
   size_t max_nack_list_size_;
 };
 
+}  // namespace acm2
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
index b047fd6d0..8011d8856 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+namespace acm2 {
+
 namespace {
 
 const int kNackThreshold = 3;
@@ -479,4 +481,6 @@ TEST(NackTest, RoudTripTimeIsApplied) {
   EXPECT_EQ(5, nack_list[1]);
 }
 
+}  // namespace acm2
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
index d6403f5c6..c1f9cdc55 100644
--- a/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/webrtc/modules/audio_coding/main/source/acm_generic_codec.h
@@ -27,6 +27,7 @@ namespace webrtc {
 
 // forward declaration
 struct CodecInst;
+struct WebRtcACMCodecParams;
 
 namespace acm1 {
 
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index c6ef8843f..62a13c572 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -3002,12 +3002,13 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
 
 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
   // Don't do anything if |max_nack_list_size| is out of range.
-  if (max_nack_list_size == 0 || max_nack_list_size > Nack::kNackListSizeLimit)
+  if (max_nack_list_size == 0 ||
+      max_nack_list_size > acm2::Nack::kNackListSizeLimit)
     return -1;
 
   CriticalSectionScoped lock(acm_crit_sect_);
   if (!nack_enabled_) {
-    nack_.reset(Nack::Create(kNackThresholdPackets));
+    nack_.reset(acm2::Nack::Create(kNackThresholdPackets));
     nack_enabled_ = true;
 
     // Sampling rate might need to be updated if we change from disable to
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
index 7b179901e..7d58f375a 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -28,7 +28,10 @@ struct WebRtcACMCodecParams;
 class CriticalSectionWrapper;
 class RWLockWrapper;
 class Clock;
+
+namespace acm2 {
 class Nack;
+}
 
 namespace acm1 {
 
@@ -437,7 +440,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
   int64_t last_receive_timestamp_;
 
   Clock* clock_;
-  scoped_ptr<Nack> nack_;
+  scoped_ptr<acm2::Nack> nack_;
   bool nack_enabled_;
 };