git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_coding/main/source/acm_common_defs.h

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+/*
+ *  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 <string.h>
+
+#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