diff options
Diffstat (limited to 'webrtc/modules/audio_coding/neteq/test/RTPencode.cc')
| -rw-r--r-- | webrtc/modules/audio_coding/neteq/test/RTPencode.cc | 63 |
1 files changed, 52 insertions, 11 deletions
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc index cbb7436152..45586ee111 100644 --- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc +++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc @@ -25,7 +25,9 @@ #include <algorithm> +#include "webrtc/base/checks.h" #include "webrtc/typedefs.h" + // needed for NetEqDecoder #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" @@ -36,6 +38,10 @@ #include "PayloadTypes.h" +namespace { +const size_t kRtpDataSize = 8000; +} + /*********************/ /* Misc. definitions */ /*********************/ @@ -126,10 +132,10 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride); #include "webrtc_vad.h" #if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC)) -#include "pcm16b.h" +#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" #endif #ifdef CODEC_G711 -#include "g711_interface.h" +#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h" #endif #ifdef CODEC_G729 #include "G729Interface.h" @@ -146,19 +152,19 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride); #include "AMRWBCreation.h" #endif #ifdef CODEC_ILBC -#include "ilbc.h" +#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h" #endif #if (defined CODEC_ISAC || defined CODEC_ISAC_SWB) -#include "isac.h" +#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h" #endif #ifdef NETEQ_ISACFIX_CODEC -#include "isacfix.h" +#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h" #ifdef CODEC_ISAC #error Cannot have both ISAC and ISACfix defined. Please de-select one. #endif #endif #ifdef CODEC_G722 -#include "g722_interface.h" +#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" #endif #ifdef CODEC_G722_1_24 #include "G722_1Interface.h" @@ -188,11 +194,14 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride); #endif #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) -#include "webrtc_cng.h" +#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 /***********************************/ /* Global codec instance variables */ @@ -264,6 +273,9 @@ SPEEX_encinst_t* SPEEX8enc_inst[2]; #ifdef CODEC_SPEEX_16 SPEEX_encinst_t* SPEEX16enc_inst[2]; #endif +#ifdef CODEC_OPUS +OpusEncInst* opus_inst[2]; +#endif int main(int argc, char* argv[]) { size_t packet_size; @@ -275,7 +287,7 @@ int main(int argc, char* argv[]) { int useRed = 0; size_t len, enc_len; int16_t org_data[4000]; - unsigned char rtp_data[8000]; + unsigned char rtp_data[kRtpDataSize]; int16_t seqNo = 0xFFF; uint32_t ssrc = 1235412312; uint32_t timestamp = 0xAC1245; @@ -286,12 +298,12 @@ int main(int argc, char* argv[]) { uint32_t red_TS[2] = {0}; uint16_t red_len[2] = {0}; size_t RTPheaderLen = 12; - uint8_t red_data[8000]; + uint8_t red_data[kRtpDataSize]; #ifdef INSERT_OLD_PACKETS uint16_t old_length, old_plen; size_t old_enc_len; int first_old_packet = 1; - unsigned char old_rtp_data[8000]; + unsigned char old_rtp_data[kRtpDataSize]; size_t packet_age = 0; #endif #ifdef INSERT_DTMF_PACKETS @@ -429,6 +441,10 @@ int main(int argc, char* argv[]) { printf(" : red_isac Redundancy RTP packet with 2*iSAC " "frames\n"); #endif +#endif // CODEC_RED +#ifdef CODEC_OPUS + printf(" : opus Opus codec with FEC (48kHz, 32kbps, FEC" + " on and tuned for 5%% packet losses)\n"); #endif printf("\n"); @@ -880,6 +896,10 @@ void NetEQTest_GetCodec_and_PT(char* name, *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */ *fs = 16000; *useRed = 1; + } else if (!strcmp(name, "opus")) { + *codec = webrtc::NetEqDecoder::kDecoderOpus; + *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */ + *fs = 48000; } else { printf("Error: Not a supported codec (%s)\n", name); exit(0); @@ -1411,12 +1431,23 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder, } break; #endif +#ifdef CODEC_OPUS + case webrtc::NetEqDecoder::kDecoderOpus: + ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0); + if (ok != 0) { + printf("Error: Couldn't allocate memory for Opus encoding " + "instance\n"); + exit(0); + } + WebRtcOpus_EnableFec(opus_inst[k]); + WebRtcOpus_SetPacketLossRate(opus_inst[k], 5); + break; +#endif default: printf("Error: unknown codec in call to NetEQTest_init_coders.\n"); exit(0); break; } - if (ok != 0) { return (ok); } @@ -1543,6 +1574,11 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) { WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]); break; #endif +#ifdef CODEC_OPUS + case webrtc::NetEqDecoder::kDecoderOpus: + WebRtcOpus_EncoderFree(opus_inst[k]); + break; +#endif default: printf("Error: unknown codec in call to NetEQTest_init_coders.\n"); exit(0); @@ -1687,6 +1723,11 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder, cdlen = static_cast<size_t>(res); } #endif +#ifdef CODEC_OPUS + cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12, + encoded); + RTC_CHECK_GT(cdlen, 0u); +#endif indata += frameLen; encoded += cdlen; totalLen += cdlen; |