/* * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef WEBRTC_ACM_COMMON_DEFS_H #define WEBRTC_ACM_COMMON_DEFS_H #include #include "audio_coding_module_typedefs.h" #include "common_types.h" #include "engine_configurations.h" #include "typedefs.h" // Checks for enabled codecs, we prevent enabling codecs which are not // compatible. #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX)) # error iSAC and iSACFX codecs cannot be enabled at the same time #endif #ifdef WIN32 // OS-dependent case-insensitive string comparison #define STR_CASE_CMP(x,y) ::_stricmp(x,y) #else // OS-dependent case-insensitive string comparison #define STR_CASE_CMP(x,y) ::strcasecmp(x,y) #endif namespace webrtc { // 60 ms is the maximum block size we support. An extra 20 ms is considered // for safety if process() method is not called when it should be, i.e. we // accept 20 ms of jitter. 80 ms @ 32 kHz (super wide-band) is 2560 samples. #define AUDIO_BUFFER_SIZE_W16 2560 // There is one timestamp per each 10 ms of audio // the audio buffer, at max, may contain 32 blocks of 10ms // audio if the sampling frequency is 8000 Hz (80 samples per block). // Therefore, The size of the buffer where we keep timestamps // is defined as follows #define TIMESTAMP_BUFFER_SIZE_W32 AUDIO_BUFFER_SIZE_W16/80 // The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo #define MAX_PAYLOAD_SIZE_BYTE 7680 // General codec specific defines #define ISACWB_DEFAULT_RATE 32000 #define ISACSWB_DEFAULT_RATE 56000 #define ISACWB_PAC_SIZE 480 #define ISACSWB_PAC_SIZE 960 // An encoded bit-stream is labeled by one of the following enumerators. // // kNoEncoding : There has been no encoding. // kActiveNormalEncoded : Active audio frame coded by the codec. // kPassiveNormalEncoded : Passive audio frame coded by the codec. // kPassiveDTXNB : Passive audio frame coded by narrow-band CN. // kPassiveDTXWB : Passive audio frame coded by wide-band CN. // kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN. // enum WebRtcACMEncodingType { kNoEncoding, kActiveNormalEncoded, kPassiveNormalEncoded, kPassiveDTXNB, kPassiveDTXWB, kPassiveDTXSWB }; // A structure which contains codec parameters. For instance, used when // initializing encoder and decoder. // // codecInstant : c.f. common_types.h // enableDTX : set true to enable DTX. If codec does not have // internal DTX, this will enable VAD. // enableVAD : set true to enable VAD. // vadMode : VAD mode, c.f. audio_coding_module_typedefs.h // for possible values. struct WebRtcACMCodecParams { CodecInst codecInstant; bool enableDTX; bool enableVAD; ACMVADMode vadMode; }; // A structure that encapsulates audio buffer and related parameters // used for synchronization of audio of two ACMs. // // inAudio : same as ACMGenericCodec::_inAudio // inAudioIxRead : same as ACMGenericCodec::_inAudioIxRead // inAudioIxWrite : same as ACMGenericCodec::_inAudioIxWrite // inTimestamp : same as ACMGenericCodec::_inTimestamp // inTimestampIxWrite : same as ACMGenericCodec::_inTImestampIxWrite // lastTimestamp : same as ACMGenericCodec::_lastTimestamp // lastInTimestamp : same as AudioCodingModuleImpl::_lastInTimestamp // struct WebRtcACMAudioBuff { WebRtc_Word16 inAudio[AUDIO_BUFFER_SIZE_W16]; WebRtc_Word16 inAudioIxRead; WebRtc_Word16 inAudioIxWrite; WebRtc_UWord32 inTimestamp[TIMESTAMP_BUFFER_SIZE_W32]; WebRtc_Word16 inTimestampIxWrite; WebRtc_UWord32 lastTimestamp; WebRtc_UWord32 lastInTimestamp; }; } // namespace webrtc #endif