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