1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ 13 14 #include <string.h> 15 16 #include "webrtc/common_types.h" 17 #include "webrtc/engine_configurations.h" 18 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" 19 #include "webrtc/typedefs.h" 20 21 // Checks for enabled codecs, we prevent enabling codecs which are not 22 // compatible. 23 #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX)) 24 #error iSAC and iSACFX codecs cannot be enabled at the same time 25 #endif 26 27 28 namespace webrtc { 29 30 // 60 ms is the maximum block size we support. An extra 20 ms is considered 31 // for safety if process() method is not called when it should be, i.e. we 32 // accept 20 ms of jitter. 80 ms @ 48 kHz (full-band) stereo is 7680 samples. 33 #define AUDIO_BUFFER_SIZE_W16 7680 34 35 // There is one timestamp per each 10 ms of audio 36 // the audio buffer, at max, may contain 32 blocks of 10ms 37 // audio if the sampling frequency is 8000 Hz (80 samples per block). 38 // Therefore, The size of the buffer where we keep timestamps 39 // is defined as follows 40 #define TIMESTAMP_BUFFER_SIZE_W32 (AUDIO_BUFFER_SIZE_W16/80) 41 42 // The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo 43 #define MAX_PAYLOAD_SIZE_BYTE 7680 44 45 // General codec specific defines 46 const int kIsacWbDefaultRate = 32000; 47 const int kIsacSwbDefaultRate = 56000; 48 const int kIsacPacSize480 = 480; 49 const int kIsacPacSize960 = 960; 50 const int kIsacPacSize1440 = 1440; 51 52 // An encoded bit-stream is labeled by one of the following enumerators. 53 // 54 // kNoEncoding : There has been no encoding. 55 // kActiveNormalEncoded : Active audio frame coded by the codec. 56 // kPassiveNormalEncoded : Passive audio frame coded by the codec. 57 // kPassiveDTXNB : Passive audio frame coded by narrow-band CN. 58 // kPassiveDTXWB : Passive audio frame coded by wide-band CN. 59 // kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN. 60 // kPassiveDTXFB : Passive audio frame coded by full-band CN. 61 enum WebRtcACMEncodingType { 62 kNoEncoding, 63 kActiveNormalEncoded, 64 kPassiveNormalEncoded, 65 kPassiveDTXNB, 66 kPassiveDTXWB, 67 kPassiveDTXSWB, 68 kPassiveDTXFB 69 }; 70 71 // A structure which contains codec parameters. For instance, used when 72 // initializing encoder and decoder. 73 // 74 // codec_inst: c.f. common_types.h 75 // enable_dtx: set true to enable DTX. If codec does not have 76 // internal DTX, this will enable VAD. 77 // enable_vad: set true to enable VAD. 78 // vad_mode: VAD mode, c.f. audio_coding_module_typedefs.h 79 // for possible values. 80 struct WebRtcACMCodecParams { 81 CodecInst codec_inst; 82 bool enable_dtx; 83 bool enable_vad; 84 ACMVADMode vad_mode; 85 }; 86 87 // TODO(turajs): Remove when ACM1 is removed. 88 struct WebRtcACMAudioBuff { 89 int16_t in_audio[AUDIO_BUFFER_SIZE_W16]; 90 int16_t in_audio_ix_read; 91 int16_t in_audio_ix_write; 92 uint32_t in_timestamp[TIMESTAMP_BUFFER_SIZE_W32]; 93 int16_t in_timestamp_ix_write; 94 uint32_t last_timestamp; 95 uint32_t last_in_timestamp; 96 }; 97 98 } // namespace webrtc 99 100 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ 101