Remove last mentions of speex from webrtc/modules

BUG=webrtc:4844

Review-Url: https://codereview.webrtc.org/2763543002
Cr-Commit-Position: refs/heads/master@{#17309}
diff --git a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
index aba525b..266595c 100644
--- a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
+++ b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
@@ -9,68 +9,25 @@
  */
 
 /* PayloadTypes.h */
-/* Used by NetEqRTPplay application */
+/* Used by RTPencode application */
+// TODO(henrik.lundin) Remove this once RTPencode is refactored.
 
 /* RTP defined codepoints */
 #define NETEQ_CODEC_PCMU_PT				0
-#define NETEQ_CODEC_GSMFR_PT			3
-#define NETEQ_CODEC_G723_PT				4
-#define NETEQ_CODEC_DVI4_PT				125 // 8 kHz version
-//#define NETEQ_CODEC_DVI4_16_PT			6  // 16 kHz version
 #define NETEQ_CODEC_PCMA_PT				8
 #define NETEQ_CODEC_G722_PT				9
 #define NETEQ_CODEC_CN_PT				13
-//#define NETEQ_CODEC_G728_PT				15
-//#define NETEQ_CODEC_DVI4_11_PT			16  // 11.025 kHz version
-//#define NETEQ_CODEC_DVI4_22_PT			17  // 22.050 kHz version
-#define NETEQ_CODEC_G729_PT				18
 
-/* Dynamic RTP codepoints as defined in VoiceEngine (file VEAPI.cpp) */
-#define NETEQ_CODEC_IPCMWB_PT			97
-#define NETEQ_CODEC_SPEEX8_PT			98
-#define NETEQ_CODEC_SPEEX16_PT			99
-#define NETEQ_CODEC_EG711U_PT			100
-#define NETEQ_CODEC_EG711A_PT			101
+/* Dynamic RTP codepoints */
 #define NETEQ_CODEC_ILBC_PT				102
 #define NETEQ_CODEC_ISAC_PT				103
-#define NETEQ_CODEC_ISACLC_PT			119
 #define NETEQ_CODEC_ISACSWB_PT			104
 #define NETEQ_CODEC_AVT_PT				106
-#define NETEQ_CODEC_G722_1_16_PT		108
-#define NETEQ_CODEC_G722_1_24_PT		109
-#define NETEQ_CODEC_G722_1_32_PT		110
 #define NETEQ_CODEC_OPUS_PT             111
-#define NETEQ_CODEC_AMR_PT				112
-#define NETEQ_CODEC_GSMEFR_PT			113
-//#define NETEQ_CODEC_ILBCRCU_PT			114
-#define NETEQ_CODEC_G726_16_PT			115
-#define NETEQ_CODEC_G726_24_PT			116
-#define NETEQ_CODEC_G726_32_PT			121
 #define NETEQ_CODEC_RED_PT				117
-#define NETEQ_CODEC_G726_40_PT			118
-//#define NETEQ_CODEC_ENERGY_PT			120
 #define NETEQ_CODEC_CN_WB_PT			105
 #define NETEQ_CODEC_CN_SWB_PT           126
-#define NETEQ_CODEC_G729_1_PT			107
-#define NETEQ_CODEC_G729D_PT			123
-#define NETEQ_CODEC_MELPE_PT			124
-
-/* Extra dynamic codepoints */
-#define NETEQ_CODEC_AMRWB_PT			120
 #define NETEQ_CODEC_PCM16B_PT			93
 #define NETEQ_CODEC_PCM16B_WB_PT		94
 #define NETEQ_CODEC_PCM16B_SWB32KHZ_PT	95
 #define NETEQ_CODEC_PCM16B_SWB48KHZ_PT	96
-#define NETEQ_CODEC_MPEG4AAC_PT			122
-
-
-/* Not default in VoiceEngine */
-#define NETEQ_CODEC_G722_1C_24_PT		84
-#define NETEQ_CODEC_G722_1C_32_PT		85
-#define NETEQ_CODEC_G722_1C_48_PT		86
-
-#define NETEQ_CODEC_SILK_8_PT			80
-#define NETEQ_CODEC_SILK_12_PT			81
-#define NETEQ_CODEC_SILK_16_PT			82
-#define NETEQ_CODEC_SILK_24_PT			83
-
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index f390f53..8e8de11 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// TODO(hlundin): Reformat file to meet style guide.
+// TODO(henrik.lundin): Refactor or replace all of this application.
 
 /* header includes */
 #include <stdio.h>
@@ -196,9 +196,6 @@
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
 #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #endif
-#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
-#include "SpeexInterface.h"
-#endif
 #ifdef CODEC_OPUS
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #endif
@@ -267,12 +264,6 @@
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
 webrtc::ComfortNoiseEncoder *CNG_encoder[2];
 #endif
-#ifdef CODEC_SPEEX_8
-SPEEX_encinst_t* SPEEX8enc_inst[2];
-#endif
-#ifdef CODEC_SPEEX_16
-SPEEX_encinst_t* SPEEX16enc_inst[2];
-#endif
 #ifdef CODEC_OPUS
 OpusEncInst* opus_inst[2];
 #endif
@@ -427,12 +418,6 @@
     printf("             : g722         g722 coder (16kHz) (the 64kbps "
            "version)\n");
 #endif
-#ifdef CODEC_SPEEX_8
-    printf("             : speex8       speex coder (8 kHz)\n");
-#endif
-#ifdef CODEC_SPEEX_16
-    printf("             : speex16      speex coder (16 kHz)\n");
-#endif
 #ifdef CODEC_RED
 #ifdef CODEC_G711
     printf("             : red_pcm      Redundancy RTP packet with 2*G711A "
@@ -1012,68 +997,6 @@
         }
         break;
 #endif
-#ifdef CODEC_SPEEX_8
-      case webrtc::kDecoderSPEEX_8:
-        if (sampfreq == 8000) {
-          if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
-              (enc_frameSize == 480)) {
-            ok = WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
-            if (ok != 0) {
-              printf("Error: Couldn't allocate memory for Speex encoding "
-                     "instance\n");
-              exit(0);
-            }
-          } else {
-            printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
-            exit(0);
-          }
-          if ((vad == 1) && (enc_frameSize != 160)) {
-            printf("\nError - This simulation only supports VAD for Speex at "
-                   "20ms packets (not %" PRIuS "ms)\n",
-                (enc_frameSize >> 3));
-            vad = 0;
-          }
-          ok = WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0 /*vbr*/,
-                                       3 /*complexity*/, vad);
-          if (ok != 0)
-            exit(0);
-        } else {
-          printf("\nError - Speex8 called with sample frequency other than 8 "
-                 "kHz.\n\n");
-        }
-        break;
-#endif
-#ifdef CODEC_SPEEX_16
-      case webrtc::kDecoderSPEEX_16:
-        if (sampfreq == 16000) {
-          if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
-              (enc_frameSize == 960)) {
-            ok = WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
-            if (ok != 0) {
-              printf("Error: Couldn't allocate memory for Speex encoding "
-                     "instance\n");
-              exit(0);
-            }
-          } else {
-            printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
-            exit(0);
-          }
-          if ((vad == 1) && (enc_frameSize != 320)) {
-            printf("\nError - This simulation only supports VAD for Speex at "
-                   "20ms packets (not %" PRIuS "ms)\n",
-                (enc_frameSize >> 4));
-            vad = 0;
-          }
-          ok = WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0 /*vbr*/,
-                                       3 /*complexity*/, vad);
-          if (ok != 0)
-            exit(0);
-        } else {
-          printf("\nError - Speex16 called with sample frequency other than 16 "
-                 "kHz.\n\n");
-        }
-        break;
-#endif
 
 #ifdef CODEC_G722_1_16
       case webrtc::kDecoderG722_1_16:
@@ -1485,16 +1408,6 @@
         WebRtcG7291_Free(G729_1_inst[k]);
         break;
 #endif
-#ifdef CODEC_SPEEX_8
-      case webrtc::NetEqDecoder::kDecoderSPEEX_8:
-        WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
-        break;
-#endif
-#ifdef CODEC_SPEEX_16
-      case webrtc::NetEqDecoder::kDecoderSPEEX_16:
-        WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
-        break;
-#endif
 
 #ifdef CODEC_G722_1_16
       case webrtc::NetEqDecoder::kDecoderG722_1_16: