| Index: webrtc/modules/audio_coding/neteq/test/RTPencode.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
|
| index f390f5330bb26f576e0f70caddf4369e60f5d96e..8e8de11ab00c4612a25eade61eddaa3c82f30a61 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 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
|
| 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 @@ GSMFR_encinst_t* GSMFRenc_inst[2];
|
| 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 @@ int main(int argc, char* argv[]) {
|
| 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 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
|
| }
|
| 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 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
|
| 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:
|
|
|