diff --git a/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h b/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h index aba525b162..266595c536 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 f390f5330b..8e8de11ab0 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 @@ -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: diff --git a/webrtc/modules/media_file/media_file_utility.h b/webrtc/modules/media_file/media_file_utility.h index bc2fa5a2f0..52cd8078df 100644 --- a/webrtc/modules/media_file/media_file_utility.h +++ b/webrtc/modules/media_file/media_file_utility.h @@ -246,9 +246,7 @@ private: kCodecG726_40, kCodecG726_32, kCodecG726_24, - kCodecG726_16, - kCodecSpeex8Khz, - kCodecSpeex16Khz + kCodecG726_16 }; // TODO (hellner): why store multiple formats. Just store either codec_info_