diff --git a/src/modules/audio_coding/main/source/acm_amr.cc b/src/modules/audio_coding/main/source/acm_amr.cc
index bc572ac68b..4e074cefea 100644
--- a/src/modules/audio_coding/main/source/acm_amr.cc
+++ b/src/modules/audio_coding/main/source/acm_amr.cc
@@ -18,9 +18,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_AMR
-    // NOTE! GSM AMR is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! GSM AMR is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/amr/main/interface/amr_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcAmr_CreateEnc(AMR_encinst_t_** encInst);
     // int16_t WebRtcAmr_CreateDec(AMR_decinst_t_** decInst);
@@ -34,12 +37,13 @@
     //  int16_t WebRtcAmr_EncoderInit(AMR_encinst_t_* encInst,
     //                               int16_t dtxMode);
     // int16_t WebRtcAmr_EncodeBitmode(AMR_encinst_t_* encInst,
-    //                                 ACMAMRPackingFormat format);
+    //                                 int format);
     // int16_t WebRtcAmr_Decode(AMR_decinst_t_* decInst);
     // int16_t WebRtcAmr_DecodePlc(AMR_decinst_t_* decInst);
     // int16_t WebRtcAmr_DecoderInit(AMR_decinst_t_* decInst);
     // int16_t WebRtcAmr_DecodeBitmode(AMR_decinst_t_* decInst,
-    //                                 ACMAMRPackingFormat format);
+    //                                 int format);
+    // void WebRtcAmr_Version(char *versionStr, short len);
     #include "amr_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_amrwb.cc b/src/modules/audio_coding/main/source/acm_amrwb.cc
index 7b41eb6ca9..5016896547 100644
--- a/src/modules/audio_coding/main/source/acm_amrwb.cc
+++ b/src/modules/audio_coding/main/source/acm_amrwb.cc
@@ -18,9 +18,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_AMRWB
-    // NOTE! GSM AMR-wb is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! GSM AMR-wb is not included in the open-source package. The
+    // following interface file is needed:
+    //
+    // /modules/audio_coding/codecs/amrwb/main/interface/amrwb_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcAmrWb_CreateEnc(AMRWB_encinst_t_** encInst);
     // int16_t WebRtcAmrWb_CreateDec(AMRWB_decinst_t_** decInst);
@@ -31,12 +34,13 @@
     // int16_t WebRtcAmrWb_EncoderInit(AMRWB_encinst_t_* encInst,
     //                                 int16_t dtxMode);
     // int16_t WebRtcAmrWb_EncodeBitmode(AMRWB_encinst_t_* encInst,
-    //                                    ACMAMRPackingFormat format);
+    //                                    int format);
     // int16_t WebRtcAmrWb_Decode(AMRWB_decinst_t_* decInst);
     // int16_t WebRtcAmrWb_DecodePlc(AMRWB_decinst_t_* decInst);
     // int16_t WebRtcAmrWb_DecoderInit(AMRWB_decinst_t_* decInst);
     // int16_t WebRtcAmrWb_DecodeBitmode(AMRWB_decinst_t_* decInst,
-    //                                   ACMAMRPackingFormat format);
+    //                                   int format);
+    // void WebRtcAmrWb_Version(char *versionStr, short len);
     #include "amrwb_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_codec_database.cc b/src/modules/audio_coding/main/source/acm_codec_database.cc
index b4d59c98ed..9f995584f2 100644
--- a/src/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/src/modules/audio_coding/main/source/acm_codec_database.cc
@@ -52,9 +52,11 @@
 #endif
 #ifdef WEBRTC_CODEC_AMR
     #include "acm_amr.h"
+    #include "amr_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_AMRWB
     #include "acm_amrwb.h"
+    #include "amrwb_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_G722
     #include "acm_g722.h"
@@ -62,21 +64,27 @@
 #endif
 #ifdef WEBRTC_CODEC_G722_1
     #include "acm_g7221.h"
+    #include "g7221_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_G722_1C
     #include "acm_g7221c.h"
+    #include "g7221c_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_G729
     #include "acm_g729.h"
+    #include "g729_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_G729_1
     #include "acm_g7291.h"
+    #include "g7291_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_GSMFR
     #include "acm_gsmfr.h"
+    #include "gsmfr_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
     #include "acm_speex.h"
+    #include "speex_interface.h"
 #endif
 #ifdef WEBRTC_CODEC_AVT
     #include "acm_dtmf_playout.h"
@@ -572,11 +580,17 @@ int ACMCodecDB::CodecsVersion(char* version, size_t* remaining_buffer_bytes,
 #endif
 #ifdef WEBRTC_CODEC_AMR
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "AMR\t\tX.X.X\n", remaining_size);
+  WebRtcAmr_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "AMR\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_AMRWB
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "AMR-WB\t\tX.X.X\n", remaining_size);
+  WebRtcAmrWb_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "AMR-WB\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_G722
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
@@ -587,27 +601,45 @@ int ACMCodecDB::CodecsVersion(char* version, size_t* remaining_buffer_bytes,
 #endif
 #ifdef WEBRTC_CODEC_G722_1
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "G.722.1\t\tX.X.X\n", remaining_size);
+  WebRtcG7221_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "G.722.1\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_G722_1C
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "G.722.1C\tX.X.X\n", remaining_size);
+  WebRtcG7221c_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "G.722.1C\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_G729
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "G.729\t\tX.X.X\n", remaining_size);
+  WebRtcG729_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "G.729\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_G729_1
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "G.729.1\t\tX.X.X\n", remaining_size);
+  WebRtcG7291_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "G.729.1\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_GSMFR
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "GSM-FR\t\tX.X.X\n", remaining_size);
+  WebRtcGSMFR_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "GSM-FR\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
-  strncat(versions_buffer, "Speex\t\tX.X.X\n", remaining_size);
+  WebRtcSpeex_Version(version_num_buf, kTemporaryBufferSize);
+  strncat(versions_buffer, "Speex\t\t", remaining_size);
+  remaining_size = kVersionBufferSize - strlen(versions_buffer);
+  strncat(versions_buffer, version_num_buf, remaining_size);
 #endif
   remaining_size = kVersionBufferSize - strlen(versions_buffer);
   WebRtcCng_Version(version_num_buf);
diff --git a/src/modules/audio_coding/main/source/acm_g7221.cc b/src/modules/audio_coding/main/source/acm_g7221.cc
index 711eb860cb..91a2e8a7cf 100644
--- a/src/modules/audio_coding/main/source/acm_g7221.cc
+++ b/src/modules/audio_coding/main/source/acm_g7221.cc
@@ -17,9 +17,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_G722_1
-    // NOTE! G.722.1 is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! G.722.1 is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/g7221/main/interface/g7221_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcG7221_CreateEnc16(G722_1_16_encinst_t_** encInst);
     // int16_t WebRtcG7221_CreateEnc24(G722_1_24_encinst_t_** encInst);
@@ -77,6 +80,7 @@
     // int16_t WebRtcG7221_DecodePlc32(G722_1_32_decinst_t_* decInst,
     //                                 int16_t* output,
     //                                 int16_t nrLostFrames);
+    // void WebRtcG7221_Version(char *versionStr, short len);
     #include "g7221_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_g7221c.cc b/src/modules/audio_coding/main/source/acm_g7221c.cc
index bbbbbb1cc2..4b1ead4c00 100644
--- a/src/modules/audio_coding/main/source/acm_g7221c.cc
+++ b/src/modules/audio_coding/main/source/acm_g7221c.cc
@@ -17,9 +17,12 @@
 #include "trace.h"
 
 #ifdef WEBRTC_CODEC_G722_1C
-    // NOTE! G.722.1C is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! G.722.1C is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/g7221c/main/interface/g7221c_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** encInst);
     // int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** encInst);
@@ -77,6 +80,7 @@
     // int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* decInst,
     //                                  int16_t* output,
     //                                  int16_t nrLostFrames);
+    // void WebRtcG7221C_Version(char *versionStr, short len);
     #include "g7221c_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_g729.cc b/src/modules/audio_coding/main/source/acm_g729.cc
index 380f9beeb6..718bcbd387 100644
--- a/src/modules/audio_coding/main/source/acm_g729.cc
+++ b/src/modules/audio_coding/main/source/acm_g729.cc
@@ -16,9 +16,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_G729
-    // NOTE! G.729 is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! G.729 is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/g729/main/interface/g729_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcG729_CreateEnc(G729_encinst_t_** inst);
     // int16_t WebRtcG729_CreateDec(G729_decinst_t_** inst);
@@ -31,6 +34,7 @@
     // int16_t WebRtcG729_DecodeBwe(G729_decinst_t_* decInst, int16_t* input);
     // int16_t WebRtcG729_DecodePlc(G729_decinst_t_* decInst);
     // int16_t WebRtcG729_DecoderInit(G729_decinst_t_* decInst);
+    // void WebRtcG729_Version(char *versionStr, short len);
     #include "g729_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_g7291.cc b/src/modules/audio_coding/main/source/acm_g7291.cc
index a6217c39c7..682ba08d56 100644
--- a/src/modules/audio_coding/main/source/acm_g7291.cc
+++ b/src/modules/audio_coding/main/source/acm_g7291.cc
@@ -16,9 +16,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_G729_1
-    // NOTE! G.729.1 is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! G.729.1 is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/g7291/main/interface/g7291_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcG7291_Create(G729_1_inst_t_** inst);
     // int16_t WebRtcG7291_Free(G729_1_inst_t_* inst);
@@ -31,6 +34,7 @@
     // int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
     // int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
     // int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
+    // void WebRtcG7291_Version(char *versionStr, short len);
     #include "g7291_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_gsmfr.cc b/src/modules/audio_coding/main/source/acm_gsmfr.cc
index 8e84f3c493..3a4e4a6169 100644
--- a/src/modules/audio_coding/main/source/acm_gsmfr.cc
+++ b/src/modules/audio_coding/main/source/acm_gsmfr.cc
@@ -16,9 +16,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_GSMFR
-    // NOTE! GSM-FR is not included in the open-source package. A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! GSM-FR is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/gsmfr/main/interface/gsmfr_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcGSMFR_CreateEnc(GSMFR_encinst_t_** inst);
     // int16_t WebRtcGSMFR_CreateDec(GSMFR_decinst_t_** inst);
@@ -31,6 +34,7 @@
     // int16_t WebRtcGSMFR_DecodeBwe(GSMFR_decinst_t_* decInst, int16_t* input);
     // int16_t WebRtcGSMFR_DecodePlc(GSMFR_decinst_t_* decInst);
     // int16_t WebRtcGSMFR_DecoderInit(GSMFR_decinst_t_* decInst);
+    // void WebRtcGSMFR_Version(char *versionStr, short len);
     #include "gsmfr_interface.h"
 #endif
 
diff --git a/src/modules/audio_coding/main/source/acm_speex.cc b/src/modules/audio_coding/main/source/acm_speex.cc
index 3943f1583b..0d58ab0742 100644
--- a/src/modules/audio_coding/main/source/acm_speex.cc
+++ b/src/modules/audio_coding/main/source/acm_speex.cc
@@ -17,9 +17,12 @@
 #include "webrtc_neteq_help_macros.h"
 
 #ifdef WEBRTC_CODEC_SPEEX
-    // NOTE! Speex is not included in the open-source package.  A wrapper is
-    // needed with interface file named as below. The API should match the one
-    // below.
+    // NOTE! Speex is not included in the open-source package. The following
+    // interface file is needed:
+    //
+    // /modules/audio_coding/codecs/speex/main/interface/speex_interface.h
+    //
+    // The API in the header file should match the one below.
     //
     // int16_t WebRtcSpeex_CreateEnc(SPEEX_encinst_t **SPEEXenc_inst,
     //                               int32_t fs);
@@ -42,6 +45,7 @@
     //                            int16_t *encoded, int16_t len,
     //                            int16_t *decoded, int16_t *speechType);
     // int16_t WebRtcSpeex_DecoderInit(SPEEX_decinst_t *SPEEXdec_inst);
+    // void WebRtcSpeex_Version(char *versionStr, short len);
     #include "speex_interface.h"
 #endif