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Remove version function from ACM codec database.

Browse Source 
BUG=461
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/667007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2619 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2012-08-16 08:17:32 +00:00
parent 5dffebc4d1
commit 6b01bfdc92
15 changed files with 4 additions and 215 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
src/modules/audio_coding/main/source/acm_amr.cc
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@ -43,7 +43,6 @@
// int16_t WebRtcAmr_DecoderInit(AMR_decinst_t_* decInst);
// int16_t WebRtcAmr_DecodeBitmode(AMR_decinst_t_* decInst,
// int format);
// void WebRtcAmr_Version(char *versionStr, short len);
#include "amr_interface.h"
#endif

1
src/modules/audio_coding/main/source/acm_amrwb.cc
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@ -40,7 +40,6 @@
// int16_t WebRtcAmrWb_DecoderInit(AMRWB_decinst_t_* decInst);
// int16_t WebRtcAmrWb_DecodeBitmode(AMRWB_decinst_t_* decInst,
// int format);
// void WebRtcAmrWb_Version(char *versionStr, short len);
#include "amrwb_interface.h"
#endif

126
src/modules/audio_coding/main/source/acm_codec_database.cc
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@ -22,7 +22,7 @@
#include "acm_common_defs.h"
#include "trace.h"
// Includes needed to get version info and to create the codecs.
// Includes needed to create the codecs.
// G.711, PCM mu-law and A-law.
#include "acm_pcma.h"
#include "acm_pcmu.h"
@ -597,130 +597,6 @@ const WebRtcNetEQDecoder* ACMCodecDB::NetEQDecoders() {
return neteq_decoders_;
}
// All version numbers for the codecs in the database are listed in text.
// TODO(tlegrand): change to use std::string.
int ACMCodecDB::CodecsVersion(char* version, size_t* remaining_buffer_bytes,
size_t* position) {
const size_t kTemporaryBufferSize = 500;
const size_t kVersionBufferSize = 1000;
char versions_buffer[kVersionBufferSize];
char version_num_buf[kTemporaryBufferSize];
size_t len = *position;
size_t remaining_size = kVersionBufferSize;
versions_buffer[0] = '\0';
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
remaining_size = kVersionBufferSize - strlen(versions_buffer);
ACM_ISAC_VERSION(version_num_buf);
strncat(versions_buffer, "ISAC\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, "\n", remaining_size);
#endif
#ifdef WEBRTC_CODEC_PCM16
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, "L16\t\t1.0.0\n", remaining_size);
#endif
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG711_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.711\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, "\n", remaining_size);
#ifdef WEBRTC_CODEC_ILBC
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcIlbcfix_version(version_num_buf);
strncat(versions_buffer, "ILBC\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, "\n", remaining_size);
#endif
#ifdef WEBRTC_CODEC_AMR
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcAmr_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "AMR\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_AMRWB
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcAmrWb_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "AMR-WB\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_G722
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG722_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.722\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_G722_1
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG7221_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.722.1\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_G722_1C
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG7221c_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.722.1C\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_G729
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG729_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.729\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_G729_1
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcG7291_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "G.729.1\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_GSMFR
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcGSMFR_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "GSM-FR\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
#ifdef WEBRTC_CODEC_SPEEX
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcSpeex_Version(version_num_buf, kTemporaryBufferSize);
strncat(versions_buffer, "Speex\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#endif
remaining_size = kVersionBufferSize - strlen(versions_buffer);
WebRtcCng_Version(version_num_buf);
strncat(versions_buffer, "CNG\t\t", remaining_size);
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, version_num_buf, remaining_size);
#ifdef WEBRTC_CODEC_AVT
remaining_size = kVersionBufferSize - strlen(versions_buffer);
strncat(versions_buffer, "Tone Generation\t1.0.0\n", remaining_size);
#endif
strncpy(&version[len], versions_buffer, *remaining_buffer_bytes);
*position = strlen(version);
*remaining_buffer_bytes -= (*position - len);
if (*remaining_buffer_bytes < strlen(versions_buffer)) {
return -1;
}
return 0;
}
// Gets mirror id. The Id is used for codecs sharing struct for settings that
// need different payload types.
int ACMCodecDB::MirrorID(int codec_id) {

12
src/modules/audio_coding/main/source/acm_codec_database.h
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@ -266,18 +266,6 @@ class ACMCodecDB {
// Returns the NetEQ decoder database.
static const WebRtcNetEQDecoder* NetEQDecoders();
// All version numbers for the codecs in the database are listed in text.
// Input/Output:
// [version] - pointer to a char vector with minimum size 1000 bytes.
// Audio coding module's and all component's versions is
// written here.
// [remaining_buffer_bytes] - remaining space in buffer.
// [position] - current position to write at in buffer.
// Return:
// -1 if version information doesn't fit, 0 on success.
static int CodecsVersion(char* version, size_t* remaining_buffer_bytes,
size_t* position);
// Returns mirror id, which is a number that tells where to find the codec's
// memory (instance). It is either the same as codec id (most common), or a
// number pointing at a different entry in the database, if the codec have

9
src/modules/audio_coding/main/source/acm_dtmf_detection.cc
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -34,11 +34,4 @@ WebRtc_Word16 ACMDTMFDetection::Detect(
return -1;
}
WebRtc_Word16 ACMDTMFDetection::GetVersion(
WebRtc_Word8* /* version */,
WebRtc_UWord32& /* remainingBufferInBytes */,
WebRtc_UWord32& /* position */) {
return -1;
}
} // namespace webrtc

6
src/modules/audio_coding/main/source/acm_dtmf_detection.h
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -29,10 +29,6 @@ class ACMDTMFDetection {
bool& toneDetected,
WebRtc_Word16& tone);
static WebRtc_Word16 GetVersion(WebRtc_Word8* version,
WebRtc_UWord32& remainingBufferInBytes,
WebRtc_UWord32& position);
private:
ACMResampler _resampler;
};

1
src/modules/audio_coding/main/source/acm_g7221.cc
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@ -80,7 +80,6 @@
// int16_t WebRtcG7221_DecodePlc32(G722_1_32_decinst_t_* decInst,
// int16_t* output,
// int16_t nrLostFrames);
// void WebRtcG7221_Version(char *versionStr, short len);
#include "g7221_interface.h"
#endif

1
src/modules/audio_coding/main/source/acm_g7221c.cc
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@ -80,7 +80,6 @@
// int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* decInst,
// int16_t* output,
// int16_t nrLostFrames);
// void WebRtcG7221C_Version(char *versionStr, short len);
#include "g7221c_interface.h"
#endif

1
src/modules/audio_coding/main/source/acm_g729.cc
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@ -34,7 +34,6 @@
// int16_t WebRtcG729_DecodeBwe(G729_decinst_t_* decInst, int16_t* input);
// int16_t WebRtcG729_DecodePlc(G729_decinst_t_* decInst);
// int16_t WebRtcG729_DecoderInit(G729_decinst_t_* decInst);
// void WebRtcG729_Version(char *versionStr, short len);
#include "g729_interface.h"
#endif

1
src/modules/audio_coding/main/source/acm_g7291.cc
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@ -34,7 +34,6 @@
// int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
// int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
// int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
// void WebRtcG7291_Version(char *versionStr, short len);
#include "g7291_interface.h"
#endif

1
src/modules/audio_coding/main/source/acm_gsmfr.cc
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@ -34,7 +34,6 @@
// int16_t WebRtcGSMFR_DecodeBwe(GSMFR_decinst_t_* decInst, int16_t* input);
// int16_t WebRtcGSMFR_DecodePlc(GSMFR_decinst_t_* decInst);
// int16_t WebRtcGSMFR_DecoderInit(GSMFR_decinst_t_* decInst);
// void WebRtcGSMFR_Version(char *versionStr, short len);
#include "gsmfr_interface.h"
#endif

4
src/modules/audio_coding/main/source/acm_isac_macros.h
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -28,7 +28,6 @@ namespace webrtc
# define ACM_ISAC_CONTROL WebRtcIsac_Control
# define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe
# define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen
# define ACM_ISAC_VERSION WebRtcIsac_version
# define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode
# define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw
# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize
@ -55,7 +54,6 @@ namespace webrtc
# define ACM_ISAC_CONTROL ACMISACFixControl // local Impl
# define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl
# define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen
# define ACM_ISAC_VERSION WebRtcIsacfix_version
# define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode
# define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl
# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize

32
src/modules/audio_coding/main/source/acm_neteq.cc
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@ -958,38 +958,6 @@ ACMNetEQ::FlushBuffers()
return 0;
}
WebRtc_Word32
ACMNetEQ::GetVersion(
char* version,
WebRtc_UWord32& remainingBufferInBytes,
WebRtc_UWord32& position)
{
WebRtc_UWord32 len = position;
strncpy(&version[position], "NetEq\t\t", remainingBufferInBytes);
position = (WebRtc_UWord32)strlen(version);
remainingBufferInBytes -= (position - len);
len = position;
char myVersion[100];
if(WebRtcNetEQ_GetVersion(myVersion) < 0)
{
return -1;
}
strncpy(&version[position], myVersion, remainingBufferInBytes);
position = (WebRtc_UWord32)strlen(version);
remainingBufferInBytes -= (position - len);
len = position;
strncpy(&version[position], "\n", remainingBufferInBytes);
position = (WebRtc_UWord32)strlen(version);
remainingBufferInBytes -= (position - len);
len = position;
return 0;
}
WebRtc_Word16
ACMNetEQ::RemoveCodec(
WebRtcNetEQDecoder codecIdx,

22
src/modules/audio_coding/main/source/acm_neteq.h
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@ -38,28 +38,6 @@ public:
// Destructor of the class.
~ACMNetEQ();
//
// GetVersion()
// Fills the version array with the NetEQ version and updates the
// remainingBufferInBytes and position variables accordingly.
//
// Output:
// - version : An array to be filled with the version
// data.
//
// Input/Output:
// - remainingBuffInBytes : The number of free bytes at the end of
// the version array.
// - position : Position where the free space starts.
//
// Return value : 0 if ok.
// -1 if NetEQ returned an error.
//
static WebRtc_Word32 GetVersion(
char* version,
WebRtc_UWord32& remainingBuffInBytes,
WebRtc_UWord32& position);
//
// Init()
// Allocates memory for NetEQ and VAD and initializes them.

1
src/modules/audio_coding/main/source/acm_speex.cc
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@ -45,7 +45,6 @@
// int16_t *encoded, int16_t len,
// int16_t *decoded, int16_t *speechType);
// int16_t WebRtcSpeex_DecoderInit(SPEEX_decinst_t *SPEEXdec_inst);
// void WebRtcSpeex_Version(char *versionStr, short len);
#include "speex_interface.h"
#endif