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Fixed all VS6 and VS8 projects with new third party projects layout 

git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/split-3rd-party@1177 74dad513-b988-da41-8d7b-12977e46ad98
This commit is contained in:
Benny Prijono 2007-04-09 07:06:08 +00:00
parent f03861b719
commit c95a0f0c33
44 changed files with 1464 additions and 4521 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
pjlib-util/build/pjlib_util.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjlib_util"
ProjectGUID="{D1823239-F34E-4134-8385-3C85FE23C7DD}"
ProjectGUID="{FE07F272-AE7F-4549-9E9F-EF9B80CB1693}"
RootNamespace="pjlib_util"
>
<Platforms>

2
pjlib/build/pjlib.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjlib"
ProjectGUID="{5D0985B4-DAA1-42DB-BDB6-30C2409EAE14}"
ProjectGUID="{DA0E03ED-53A7-4050-8A85-90541C5509F8}"
RootNamespace="pjlib"
>
<Platforms>

22
pjmedia/build/pjmedia.dsp
View File

@ -42,7 +42,7 @@ RSC=rc.exe
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MD /W4 /GX /Zi /O2 /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../src/pjmedia/portaudio" /I "../src/pjmedia-codec" /I "../../pjnath/include" /I "../../third_party/portaudio/include" /I "../../third_party/speex/include" /D "NDEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /c
# ADD CPP /nologo /MD /W4 /GX /Zi /O2 /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../../pjnath/include" /I "../../third_party/portaudio/include" /I "../../third_party/speex/include" /I "../.." /D "NDEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /c
# SUBTRACT CPP /YX
# ADD BASE RSC /l 0x409 /d "NDEBUG"
# ADD RSC /l 0x409 /d "NDEBUG"
@ -67,7 +67,7 @@ LIB32=link.exe -lib
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
# ADD CPP /nologo /MTd /W4 /Gm /GX /ZI /Od /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../src/pjmedia/portaudio" /I "../src/pjmedia-codec" /I "../../pjnath/include" /I "../../third_party/portaudio/include" /I "../../third_party/speex/include" /D "_DEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /GZ /c
# ADD CPP /nologo /MTd /W4 /Gm /GX /ZI /Od /I "../include" /I "../../pjlib/include" /I "../../pjlib-util/include" /I "../../pjnath/include" /I "../../third_party/portaudio/include" /I "../../third_party/speex/include" /I "../.." /D "_DEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D PJ_WIN32=1 /D PJ_M_I386=1 /FR /FD /GZ /c
# SUBTRACT CPP /YX
# ADD BASE RSC /l 0x409 /d "_DEBUG"
# ADD RSC /l 0x409 /d "_DEBUG"
@ -161,10 +161,6 @@ SOURCE=..\src\pjmedia\mem_player.c
# End Source File
# Begin Source File
SOURCE=..\src\pjmedia\mp3_writer.c
# End Source File
# Begin Source File
SOURCE=..\src\pjmedia\null_port.c
# End Source File
# Begin Source File
@ -181,15 +177,6 @@ SOURCE=..\src\pjmedia\plc_common.c
# End Source File
# Begin Source File
SOURCE=..\src\pjmedia\plc_steveu.c
# ADD CPP /W2
# End Source File
# Begin Source File
SOURCE=..\src\pjmedia\plc_steveu.h
# End Source File
# Begin Source File
SOURCE=..\src\pjmedia\port.c
# End Source File
# Begin Source File
@ -203,7 +190,6 @@ SOURCE=..\src\pjmedia\resample_resample.c
# Begin Source File
SOURCE=..\src\pjmedia\resample_speex.c
# PROP Exclude_From_Build 1
# End Source File
# Begin Source File
@ -339,10 +325,6 @@ SOURCE=..\include\pjmedia\mem_port.h
# End Source File
# Begin Source File
SOURCE=..\include\pjmedia\mp3_port.h
# End Source File
# Begin Source File
SOURCE=..\include\pjmedia\null_port.h
# End Source File
# Begin Source File

530
pjmedia/build/pjmedia.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjmedia"
ProjectGUID="{1434FE6C-89C6-4E13-8150-D5A4B1038AAA}"
ProjectGUID="{7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65}"
RootNamespace="pjmedia"
>
<Platforms>
@ -43,8 +43,8 @@
Name="VCCLCompilerTool"
Optimization="2"
InlineFunctionExpansion="1"
AdditionalIncludeDirectories="../include,../../pjlib/include,../../pjlib-util/include,../src/pjmedia/portaudio,../src/pjmedia-codec,../../pjnath/include"
PreprocessorDefinitions="NDEBUG;PA_NO_ASIO;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1"
AdditionalIncludeDirectories="../include,../../pjlib/include,../../pjlib-util/include,../../pjnath/include,../../third_party/portaudio/include,../../third_party/speex/include,../.."
PreprocessorDefinitions="NDEBUG;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1"
StringPooling="true"
RuntimeLibrary="2"
EnableFunctionLevelLinking="true"
@ -119,8 +119,8 @@
<Tool
Name="VCCLCompilerTool"
Optimization="0"
AdditionalIncludeDirectories="../include,../../pjlib/include,../../pjlib-util/include,../src/pjmedia/portaudio,../src/pjmedia-codec,../../pjnath/include"
PreprocessorDefinitions="_DEBUG;PA_NO_ASIO;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1"
AdditionalIncludeDirectories="../include,../../pjlib/include,../../pjlib-util/include,../../pjnath/include,../../third_party/portaudio/include,../../third_party/speex/include,../.."
PreprocessorDefinitions="_DEBUG;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1"
MinimalRebuild="true"
BasicRuntimeChecks="3"
RuntimeLibrary="1"
@ -571,28 +571,6 @@
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\mp3_writer.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\null_port.c"
>
@ -681,34 +659,6 @@
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\plc_steveu.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="2"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="2"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\plc_steveu.h"
>
</File>
<File
RelativePath="..\src\pjmedia\port.c"
>
@ -731,28 +681,6 @@
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\resample.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\resample_port.c"
>
@ -775,6 +703,14 @@
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\resample_resample.c"
>
</File>
<File
RelativePath="..\src\pjmedia\resample_speex.c"
>
</File>
<File
RelativePath="..\src\pjmedia\rtcp.c"
>
@ -1196,10 +1132,6 @@
RelativePath="..\include\pjmedia\mem_port.h"
>
</File>
<File
RelativePath="..\include\pjmedia\mp3_port.h"
>
</File>
<File
RelativePath="..\include\pjmedia\null_port.h"
>
@ -1293,442 +1225,6 @@
>
</File>
</Filter>
<Filter
Name="PortAudio Files"
>
<File
RelativePath="..\src\pjmedia\portaudio\dsound_wrapper.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\dsound_wrapper.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_allocation.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_allocation.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_converters.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_converters.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_cpuload.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_cpuload.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_dither.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_dither.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_endianness.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_front.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_hostapi.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_process.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_process.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_skeleton.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_stream.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_stream.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_trace.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_trace.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_types.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_util.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_win_ds.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_win_hostapis.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_win_util.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_win_wmme.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_win_wmme.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_x86_plain_converters.c"
>
<FileConfiguration
Name="Release|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
<FileConfiguration
Name="Debug|Win32"
>
<Tool
Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
WarningLevel="3"
/>
</FileConfiguration>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\pa_x86_plain_converters.h"
>
</File>
<File
RelativePath="..\src\pjmedia\portaudio\portaudio.h"
>
</File>
</Filter>
<Filter
Name="Resample Files"
>
<File
RelativePath="..\src\pjmedia\largefilter.h"
>
</File>
<File
RelativePath="..\src\pjmedia\smallfilter.h"
>
</File>
</Filter>
</Files>
<Globals>
</Globals>

2198
pjmedia/build/pjmedia_codec.vcproj
View File

File diff suppressed because it is too large Load Diff

1
pjmedia/include/pjmedia.h
View File

@ -38,7 +38,6 @@
#include <pjmedia/jbuf.h>
#include <pjmedia/master_port.h>
#include <pjmedia/mem_port.h>
#include <pjmedia/mp3_port.h>
#include <pjmedia/null_port.h>
#include <pjmedia/plc.h>
#include <pjmedia/port.h>

52
pjmedia/include/pjmedia/config.h
View File

@ -108,24 +108,57 @@
#endif
/**
/*
* ** THIS MACRO IS DEPRECATED in 0.6. **
* ** See libresample for configuring this **
*
* Include small filter table in resample.
* This adds about 9KB in rdata.
*/
/*
#ifndef PJMEDIA_HAS_SMALL_FILTER
# define PJMEDIA_HAS_SMALL_FILTER 1
#endif
*/
/*
* ** THIS MACRO IS DEPRECATED in 0.6. **
* ** See libresample for configuring this **
*
* Include large filter table in resample.
* This adds about 32KB in rdata.
*/
/*
#ifndef PJMEDIA_HAS_LARGE_FILTER
# define PJMEDIA_HAS_LARGE_FILTER 1
#endif
*/
/**
* Specify whether libresample should be used for the sampling
* rate conversion. This macro and PJMEDIA_HAS_SPEEX_RESAMPLE
* are mutually exclusive.
*
* Default: 1 (Yes)
*/
#ifndef PJMEDIA_HAS_LIBRESAMPLE
# define PJMEDIA_HAS_LIBRESAMPLE 1
#endif
/**
* Include large filter table in resample.
* This adds about 32KB in rdata.
* Specify whether Speex sample rate convertor should be used for the
* sampling rate conversion. This macro and PJMEDIA_HAS_LIBRESAMPLE
* are mutually exclusive.
*
* Default: 0
*/
#ifndef PJMEDIA_HAS_LARGE_FILTER
# define PJMEDIA_HAS_LARGE_FILTER 1
#ifndef PJMEDIA_HAS_SPEEX_RESAMPLE
# define PJMEDIA_HAS_SPEEX_RESAMPLE 0
#endif
/**
* Default file player/writer buffer size.
*/
@ -243,9 +276,14 @@
/**
* Enable Steve Underwood's PLC.
*
* ** This has now been deprecated. If the codec does not have **
* ** PLC, then no PLC will be used for that particular codec. **
*
* Set this to zero, or other link error will occur.
*/
#ifndef PJMEDIA_HAS_STEVEU_PLC
# define PJMEDIA_HAS_STEVEU_PLC PJ_HAS_FLOATING_POINT
# define PJMEDIA_HAS_STEVEU_PLC 0
#endif
@ -275,7 +313,7 @@
* 16 will effectively zero the signal.
*/
#ifndef PJMEDIA_ECHO_SUPPRESS_FACTOR
# define PJMEDIA_ECHO_SUPPRESS_FACTOR 4
# define PJMEDIA_ECHO_SUPPRESS_FACTOR 10
#endif

25
pjmedia/src/pjmedia-codec/gsm.c
View File

@ -34,6 +34,9 @@
*/
#if defined(PJMEDIA_HAS_GSM_CODEC) && PJMEDIA_HAS_GSM_CODEC != 0
/* We removed PLC in 0.6 */
#define PLC_DISABLED 1
/* Prototypes for GSM factory */
static pj_status_t gsm_test_alloc( pjmedia_codec_factory *factory,
@ -72,9 +75,11 @@ static pj_status_t gsm_codec_decode( pjmedia_codec *codec,
const struct pjmedia_frame *input,
unsigned output_buf_len,
struct pjmedia_frame *output);
#if !PLC_DISABLED
static pj_status_t gsm_codec_recover(pjmedia_codec *codec,
unsigned output_buf_len,
struct pjmedia_frame *output);
#endif
/* Definition for GSM codec operations. */
static pjmedia_codec_op gsm_op =
@ -86,7 +91,11 @@ static pjmedia_codec_op gsm_op =
&gsm_codec_parse,
&gsm_codec_encode,
&gsm_codec_decode,
#if !PLC_DISABLED
&gsm_codec_recover
#else
NULL
#endif
};
/* Definition for GSM codec factory operations. */
@ -116,7 +125,9 @@ struct gsm_data
void *encoder;
void *decoder;
pj_bool_t plc_enabled;
#if !PLC_DISABLED
pjmedia_plc *plc;
#endif
pj_bool_t vad_enabled;
pjmedia_silence_det *vad;
pj_timestamp last_tx;
@ -259,7 +270,9 @@ static pj_status_t gsm_default_attr (pjmedia_codec_factory *factory,
attr->setting.frm_per_pkt = 1;
attr->setting.vad = 1;
#if !PLC_DISABLED
attr->setting.plc = 1;
#endif
/* Default all other flag bits disabled. */
@ -320,6 +333,7 @@ static pj_status_t gsm_alloc_codec( pjmedia_codec_factory *factory,
sizeof(struct gsm_data));
codec->codec_data = gsm_data;
#if !PLC_DISABLED
/* Create PLC */
status = pjmedia_plc_create(gsm_codec_factory.pool, 8000,
160, 0, &gsm_data->plc);
@ -327,6 +341,7 @@ static pj_status_t gsm_alloc_codec( pjmedia_codec_factory *factory,
pj_mutex_unlock(gsm_codec_factory.mutex);
return status;
}
#endif
/* Create silence detector */
status = pjmedia_silence_det_create(gsm_codec_factory.pool,
@ -351,7 +366,6 @@ static pj_status_t gsm_dealloc_codec( pjmedia_codec_factory *factory,
pjmedia_codec *codec )
{
struct gsm_data *gsm_data;
pj_int16_t frame[160];
int i;
PJ_ASSERT_RETURN(factory && codec, PJ_EINVAL);
@ -362,13 +376,18 @@ static pj_status_t gsm_dealloc_codec( pjmedia_codec_factory *factory,
/* Close codec, if it's not closed. */
gsm_codec_close(codec);
#if !PLC_DISABLED
/* Clear left samples in the PLC, since codec+plc will be reused
* next time.
*/
for (i=0; i<2; ++i) {
pj_int16_t frame[160];
pjmedia_zero_samples(frame, PJ_ARRAY_SIZE(frame));
pjmedia_plc_save(gsm_data->plc, frame);
}
#else
PJ_UNUSED_ARG(i);
#endif
/* Re-init silence_period */
pj_set_timestamp32(&gsm_data->last_tx, 0, 0);
@ -569,13 +588,16 @@ static pj_status_t gsm_codec_decode( pjmedia_codec *codec,
output->size = 320;
output->type = PJMEDIA_FRAME_TYPE_AUDIO;
#if !PLC_DISABLED
if (gsm_data->plc_enabled)
pjmedia_plc_save( gsm_data->plc, output->buf);
#endif
return PJ_SUCCESS;
}
#if !PLC_DISABLED
/*
* Recover lost frame.
*/
@ -594,6 +616,7 @@ static pj_status_t gsm_codec_recover(pjmedia_codec *codec,
return PJ_SUCCESS;
}
#endif
#endif /* PJMEDIA_HAS_GSM_CODEC */

30
pjmedia/src/pjmedia/g711.c
View File

@ -32,6 +32,9 @@
#if defined(PJMEDIA_HAS_G711_CODEC) && PJMEDIA_HAS_G711_CODEC!=0
/* We removed PLC in 0.6 */
#define PLC_DISABLED 1
#define G711_BPS 64000
#define G711_CODEC_CNT 0 /* number of codec to preallocate in memory */
@ -80,9 +83,11 @@ static pj_status_t g711_decode( pjmedia_codec *codec,
const struct pjmedia_frame *input,
unsigned output_buf_len,
struct pjmedia_frame *output);
#if !PLC_DISABLED
static pj_status_t g711_recover( pjmedia_codec *codec,
unsigned output_buf_len,
struct pjmedia_frame *output);
#endif
/* Definition for G711 codec operations. */
static pjmedia_codec_op g711_op =
@ -94,7 +99,11 @@ static pjmedia_codec_op g711_op =
&g711_parse,
&g711_encode,
&g711_decode,
#if !PLC_DISABLED
&g711_recover
#else
NULL
#endif
};
/* Definition for G711 codec factory operations. */
@ -121,8 +130,10 @@ static struct g711_factory
struct g711_private
{
unsigned pt;
#if !PLC_DISABLED
pj_bool_t plc_enabled;
pjmedia_plc *plc;
#endif
pj_bool_t vad_enabled;
pjmedia_silence_det *vad;
pj_timestamp last_tx;
@ -249,8 +260,10 @@ static pj_status_t g711_default_attr (pjmedia_codec_factory *factory,
/* Set default frames per packet to 2 (or 20ms) */
attr->setting.frm_per_pkt = 2;
#if !PLC_DISABLED
/* Enable plc by default. */
attr->setting.plc = 1;
#endif
/* Enable VAD by default. */
attr->setting.vad = 1;
@ -317,6 +330,7 @@ static pj_status_t g711_alloc_codec( pjmedia_codec_factory *factory,
/* Set the payload type */
codec_priv->pt = id->pt;
#if !PLC_DISABLED
/* Create PLC, always with 10ms ptime */
status = pjmedia_plc_create(g711_factory.pool, 8000, 80,
0, &codec_priv->plc);
@ -324,6 +338,7 @@ static pj_status_t g711_alloc_codec( pjmedia_codec_factory *factory,
pj_mutex_unlock(g711_factory.mutex);
return status;
}
#endif
/* Create VAD */
status = pjmedia_silence_det_create(g711_factory.pool,
@ -357,7 +372,6 @@ static pj_status_t g711_dealloc_codec(pjmedia_codec_factory *factory,
pjmedia_codec *codec )
{
struct g711_private *priv = codec->codec_data;
pj_int16_t frame[SAMPLES_PER_FRAME];
int i;
PJ_ASSERT_RETURN(factory==&g711_factory.base, PJ_EINVAL);
@ -368,13 +382,19 @@ static pj_status_t g711_dealloc_codec(pjmedia_codec_factory *factory,
return PJ_EINVALIDOP;
}
#if !PLC_DISABLED
/* Clear left samples in the PLC, since codec+plc will be reused
* next time.
*/
for (i=0; i<2; ++i) {
pj_int16_t frame[SAMPLES_PER_FRAME];
pjmedia_zero_samples(frame, PJ_ARRAY_SIZE(frame));
pjmedia_plc_save(priv->plc, frame);
}
#else
PJ_UNUSED_ARG(i);
PJ_UNUSED_ARG(priv);
#endif
/* Lock mutex. */
pj_mutex_lock(g711_factory.mutex);
@ -402,7 +422,9 @@ static pj_status_t g711_open(pjmedia_codec *codec,
{
struct g711_private *priv = codec->codec_data;
priv->pt = attr->info.pt;
#if !PLC_DISABLED
priv->plc_enabled = (attr->setting.plc != 0);
#endif
priv->vad_enabled = (attr->setting.vad != 0);
return PJ_SUCCESS;
}
@ -422,7 +444,9 @@ static pj_status_t g711_modify(pjmedia_codec *codec,
if (attr->info.pt != priv->pt)
return PJMEDIA_EINVALIDPT;
#if !PLC_DISABLED
priv->plc_enabled = (attr->setting.plc != 0);
#endif
priv->vad_enabled = (attr->setting.vad != 0);
return PJ_SUCCESS;
@ -561,12 +585,15 @@ static pj_status_t g711_decode(pjmedia_codec *codec,
output->type = PJMEDIA_FRAME_TYPE_AUDIO;
output->size = (input->size << 1);
#if !PLC_DISABLED
if (priv->plc_enabled)
pjmedia_plc_save( priv->plc, output->buf);
#endif
return PJ_SUCCESS;
}
#if !PLC_DISABLED
static pj_status_t g711_recover( pjmedia_codec *codec,
unsigned output_buf_len,
struct pjmedia_frame *output)
@ -584,6 +611,7 @@ static pj_status_t g711_recover( pjmedia_codec *codec,
return PJ_SUCCESS;
}
#endif
#endif /* PJMEDIA_HAS_G711_CODEC */

526
pjmedia/src/pjmedia/resample_resample.c
View File

@ -17,451 +17,21 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/*
* Based on:
* resample-1.8.tar.gz from the
* Digital Audio Resampling Home Page located at
* http://www-ccrma.stanford.edu/~jos/resample/.
*
* SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN
*
* Snippet from the resample.1 man page:
*
* HISTORY
*
* The first version of this software was written by Julius O. Smith III
* <jos@ccrma.stanford.edu> at CCRMA <http://www-ccrma.stanford.edu> in
* 1981. It was called SRCONV and was written in SAIL for PDP-10
* compatible machines. The algorithm was first published in
*
* Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate
* Conversion Method,'' Proceedings (2): 19.4.1-19.4.4, IEEE Conference
* on Acoustics, Speech, and Signal Processing, San Diego, March 1984.
*
* An expanded tutorial based on this paper is available at the Digital
* Audio Resampling Home Page given above.
*
* Circa 1988, the SRCONV program was translated from SAIL to C by
* Christopher Lee Fraley working with Roger Dannenberg at CMU.
*
* Since then, the C version has been maintained by jos.
*
* Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>.
*
* The resample program is free software distributed in accordance
* with the Lesser GNU Public License (LGPL). There is NO warranty; not
* even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*/
/* PJMEDIA modification:
* - remove resample(), just use SrcUp, SrcUD, and SrcLinear directly.
* - move FilterUp() and FilterUD() from filterkit.c
* - move stddefs.h and resample.h to this file.
* - const correctness.
*/
#include <pjmedia/resample.h>
#include <pjmedia/errno.h>
#include <pj/assert.h>
#include <pj/log.h>
#include <pj/pool.h>
#if defined(PJMEDIA_HAS_LIBRESAMPLE) && PJMEDIA_HAS_LIBRESAMPLE != 0
#include <third_party/resample/include/resamplesubs.h>
#define THIS_FILE "resample.c"
/*
* Taken from stddefs.h
*/
#ifndef PI
#define PI (3.14159265358979232846)
#endif
#ifndef PI2
#define PI2 (6.28318530717958465692)
#endif
#define D2R (0.01745329348) /* (2*pi)/360 */
#define R2D (57.29577951) /* 360/(2*pi) */
#ifndef MAX
#define MAX(x,y) ((x)>(y) ?(x):(y))
#endif
#ifndef MIN
#define MIN(x,y) ((x)<(y) ?(x):(y))
#endif
#ifndef ABS
#define ABS(x) ((x)<0 ?(-(x)):(x))
#endif
#ifndef SGN
#define SGN(x) ((x)<0 ?(-1):((x)==0?(0):(1)))
#endif
typedef char RES_BOOL;
typedef short RES_HWORD;
typedef int RES_WORD;
typedef unsigned short RES_UHWORD;
typedef unsigned int RES_UWORD;
#define MAX_HWORD (32767)
#define MIN_HWORD (-32768)
#ifdef DEBUG
#define INLINE
#else
#define INLINE inline
#endif
/*
* Taken from resample.h
*
* The configuration constants below govern
* the number of bits in the input sample and filter coefficients, the
* number of bits to the right of the binary-point for fixed-point math, etc.
*
*/
/* Conversion constants */
#define Nhc 8
#define Na 7
#define Np (Nhc+Na)
#define Npc (1<<Nhc)
#define Amask ((1<<Na)-1)
#define Pmask ((1<<Np)-1)
#define Nh 16
#define Nb 16
#define Nhxn 14
#define Nhg (Nh-Nhxn)
#define NLpScl 13
/* Description of constants:
*
* Npc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Npc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Npc must be a power of 2 due to the details of the current
* implementation. The default value of 512 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit accuracy in filter coefficients.
*
* Nhc - is log base 2 of Npc.
*
* Na - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Np - is Na + Nhc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. Np must be less than 16 in
* this implementation.
*
* Nh - is the number of bits in the filter coefficients. The sum of Nh and
* the number of bits in the input data (typically 16) cannot exceed 32.
* Thus Nh should be 16. The largest filter coefficient should nearly
* fill 16 bits (32767).
*
* Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
* exceed 32.
*
* Nhxn - is the number of bits to right shift after multiplying each input
* sample times a filter coefficient. It can be as great as Nh and as
* small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
* accumulation. If Nhxn=0, the accumulation will soon overflow 32 bits.
*
* Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
*
* NLpScl - is the number of bits allocated to the unity-gain normalization
* factor. The output of the lowpass filter is multiplied by LpScl and
* then right-shifted NLpScl bits. To avoid overflow, we must have
* Nb+Nhg+NLpScl < 32.
*/
#ifdef _MSC_VER
# pragma warning(push, 3)
//# pragma warning(disable: 4245) // Conversion from uint to ushort
# pragma warning(disable: 4244) // Conversion from double to uint
# pragma warning(disable: 4146) // unary minus operator applied to unsigned type, result still unsigned
# pragma warning(disable: 4761) // integral size mismatch in argument; conversion supplied
#endif
#if defined(PJMEDIA_HAS_SMALL_FILTER) && PJMEDIA_HAS_SMALL_FILTER!=0
# include "smallfilter.h"
#else
# define SMALL_FILTER_NMULT 0
# define SMALL_FILTER_SCALE 0
# define SMALL_FILTER_NWING 0
# define SMALL_FILTER_IMP NULL
# define SMALL_FILTER_IMPD NULL
#endif
#if defined(PJMEDIA_HAS_LARGE_FILTER) && PJMEDIA_HAS_LARGE_FILTER!=0
# include "largefilter.h"
#else
# define LARGE_FILTER_NMULT 0
# define LARGE_FILTER_SCALE 0
# define LARGE_FILTER_NWING 0
# define LARGE_FILTER_IMP NULL
# define LARGE_FILTER_IMPD NULL
#endif
#undef INLINE
#define INLINE
#define HAVE_FILTER 0
#ifndef NULL
# define NULL 0
#endif
static INLINE RES_HWORD WordToHword(RES_WORD v, int scl)
{
RES_HWORD out;
RES_WORD llsb = (1<<(scl-1));
v += llsb; /* round */
v >>= scl;
if (v>MAX_HWORD) {
v = MAX_HWORD;
} else if (v < MIN_HWORD) {
v = MIN_HWORD;
}
out = (RES_HWORD) v;
return out;
}
/* Sampling rate conversion using linear interpolation for maximum speed.
*/
static int
SrcLinear(const RES_HWORD X[], RES_HWORD Y[], double pFactor, RES_UHWORD nx)
{
RES_HWORD iconst;
RES_UWORD time = 0;
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v,x1,x2;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
iconst = (time) & Pmask;
xp = &X[(time)>>Np]; /* Ptr to current input sample */
x1 = *xp++;
x2 = *xp;
x1 *= ((1<<Np)-iconst);
x2 *= iconst;
v = x1 + x2;
*Y++ = WordToHword(v,Np); /* Deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return number of output samples */
}
static RES_WORD FilterUp(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc)
{
const RES_HWORD *Hp;
const RES_HWORD *Hdp = NULL;
const RES_HWORD *End;
RES_HWORD a = 0;
RES_WORD v, t;
v=0;
Hp = &Imp[Ph>>Na];
End = &Imp[Nwing];
if (Interp) {
Hdp = &ImpD[Ph>>Na];
a = Ph & Amask;
}
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp += Npc; /* first sample, so we must also */
Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
}
}
if (Interp)
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
Hdp += Npc; /* Filter coeff differences step */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
static RES_WORD FilterUD(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc, RES_UHWORD dhb)
{
RES_HWORD a;
const RES_HWORD *Hp, *Hdp, *End;
RES_WORD v, t;
RES_UWORD Ho;
v=0;
Ho = (Ph*(RES_UWORD)dhb)>>Np;
End = &Imp[Nwing];
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
Ho += dhb; /* ...then we've already skipped the */
} /* first sample, so we must also */
/* skip ahead in Imp[] and ImpD[] */
if (Interp)
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
Hdp = &ImpD[Ho>>Na]; /* get interp (lower Na) bits from diff table*/
a = Ho & Amask; /* a is logically between 0 and 1 */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
/* Sampling rate up-conversion only subroutine;
* Slightly faster than down-conversion;
*/
static int SrcUp(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
/* Perform left-wing inner product */
v = 0;
v = FilterUp(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),-1);
/* Perform right-wing inner product */
v += FilterUp(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),1);
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* Sampling rate conversion subroutine */
static int SrcUD(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dh; /* Step through filter impulse response */
double dt; /* Step through input signal */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
RES_UWORD dhb, dtb; /* Fixed-point versions of Dh,Dt */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
dh = MIN(Npc, pFactor*Npc); /* Filter sampling period */
dhb = dh*(1<<Na) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
v = FilterUD(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),
-1, dhb); /* Perform left-wing inner product */
v += FilterUD(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),
1, dhb); /* Perform right-wing inner product */
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* ***************************************************************************
*
* PJMEDIA RESAMPLE
*
* ***************************************************************************
*/
struct pjmedia_resample
{
@ -503,31 +73,6 @@ PJ_DEF(pj_status_t) pjmedia_resample_create( pj_pool_t *pool,
//high_quality = 0;
}
#if !defined(PJMEDIA_HAS_LARGE_FILTER) || PJMEDIA_HAS_LARGE_FILTER==0
/*
* If large filter is excluded in the build, then prevent application
* from using it.
*/
if (high_quality && large_filter) {
large_filter = PJ_FALSE;
PJ_LOG(5,(THIS_FILE,
"Resample uses small filter because large filter is "
"disabled"));
}
#endif
#if !defined(PJMEDIA_HAS_SMALL_FILTER) || PJMEDIA_HAS_SMALL_FILTER==0
/*
* If small filter is excluded in the build and application wants to
* use it, then drop to linear conversion.
*/
if (high_quality && large_filter == 0) {
high_quality = PJ_FALSE;
PJ_LOG(4,(THIS_FILE,
"Resample uses linear because small filter is disabled"));
}
#endif
resample->factor = rate_out * 1.0 / rate_in;
resample->large_filter = large_filter;
resample->high_quality = high_quality;
@ -540,13 +85,7 @@ PJ_DEF(pj_status_t) pjmedia_resample_create( pj_pool_t *pool,
* of Macadamian dot com.
* resample->xoff = large_filter ? 32 : 6;
*/
if (large_filter)
resample->xoff = (LARGE_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/resample->factor);
else
resample->xoff = (SMALL_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/resample->factor);
resample->xoff = res_GetXOFF(resample->factor, (char)large_filter);
size = (samples_per_frame + 2*resample->xoff) * sizeof(pj_int16_t);
resample->buffer = pj_pool_alloc(pool, size);
@ -649,53 +188,18 @@ PJ_DEF(void) pjmedia_resample_run( pjmedia_resample *resample,
*/
dst_buf = resample->buffer + resample->xoff*2;
pjmedia_copy_samples(dst_buf, input, resample->frame_size);
if (resample->factor >= 1) {
if (resample->large_filter) {
SrcUp(resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
LARGE_FILTER_NWING, LARGE_FILTER_SCALE,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
PJ_TRUE);
} else {
SrcUp(resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
SMALL_FILTER_NWING, SMALL_FILTER_SCALE,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
PJ_TRUE);
}
} else {
if (resample->large_filter) {
SrcUD( resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
LARGE_FILTER_NWING,
LARGE_FILTER_SCALE * resample->factor + 0.5,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
PJ_TRUE);
} else {
SrcUD( resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
SMALL_FILTER_NWING,
SMALL_FILTER_SCALE * resample->factor + 0.5,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
PJ_TRUE);
}
}
res_Resample(resample->buffer + resample->xoff, output,
resample->factor, (pj_uint16_t)resample->frame_size,
(char)resample->large_filter, (char)PJ_TRUE);
dst_buf = resample->buffer;
src_buf = input + resample->frame_size - resample->xoff*2;
pjmedia_copy_samples(dst_buf, src_buf, resample->xoff * 2);
} else {
SrcLinear( input, output, resample->factor, resample->frame_size);
res_SrcLinear( input, output, resample->factor,
(pj_uint16_t)resample->frame_size);
}
}
@ -709,3 +213,11 @@ PJ_DEF(void) pjmedia_resample_destroy(pjmedia_resample *resample)
{
PJ_UNUSED_ARG(resample);
}
#else /* PJMEDIA_HAS_LIBRESAMPLE */
int pjmedia_resample_resample_excluded;
#endif /* PJMEDIA_HAS_LIBRESAMPLE */

11
pjmedia/src/pjmedia/resample_speex.c
View File

@ -16,16 +16,17 @@
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <pjmedia/resample.h>
#include <pjmedia/errno.h>
#include <pj/assert.h>
#include <pj/log.h>
#include <pj/pool.h>
#if defined(PJMEDIA_HAS_SPEEX_RESAMPLE) && PJMEDIA_HAS_SPEEX_RESAMPLE != 0
#include <speex/speex_resampler.h>
#define THIS_FILE "resample.c"
#define THIS_FILE "resample_speex.c"
struct pjmedia_resample
@ -117,3 +118,9 @@ PJ_DEF(void) pjmedia_resample_destroy(pjmedia_resample *resample)
}
}
#else /* PJMEDIA_HAS_SPEEX_RESAMPLE */
int pjmedia_resample_speex_excluded;
#endif /* PJMEDIA_HAS_SPEEX_RESAMPLE */

54
pjproject-vs8.sln
View File

@ -28,29 +28,39 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "pjsip_ua", "pjsip\build\pjs
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "pjsua", "pjsip-apps\build\pjsua.vcproj", "{8310649E-A25E-4AF0-91E8-9E3CC659BB89}"
ProjectSection(ProjectDependencies) = postProject
{2BB84911-C1B4-4747-B93D-36AA82CC5031} = {2BB84911-C1B4-4747-B93D-36AA82CC5031}
{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4} = {A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}
{E53AA5FF-B737-40AA-BD13-387EFA99023D} = {E53AA5FF-B737-40AA-BD13-387EFA99023D}
{9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37} = {9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37}
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D} = {B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}
{DA0E03ED-53A7-4050-8A85-90541C5509F8} = {DA0E03ED-53A7-4050-8A85-90541C5509F8}
{B8719FD5-E8A6-4A36-943C-891D07F5DD21} = {B8719FD5-E8A6-4A36-943C-891D07F5DD21}
{4B5945CD-0CB3-49AA-A7FF-7612D93F82C0} = {4B5945CD-0CB3-49AA-A7FF-7612D93F82C0}
{855DC8C0-D3E9-4A2E-AE47-116605A7BC9B} = {855DC8C0-D3E9-4A2E-AE47-116605A7BC9B}
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9} = {4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA} = {3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}
{7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65} = {7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65}
{6794B975-4E84-4F49-B2DC-C31F2224E03E} = {6794B975-4E84-4F49-B2DC-C31F2224E03E}
{FE07F272-AE7F-4549-9E9F-EF9B80CB1693} = {FE07F272-AE7F-4549-9E9F-EF9B80CB1693}
{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4} = {A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}
{2BB84911-C1B4-4747-B93D-36AA82CC5031} = {2BB84911-C1B4-4747-B93D-36AA82CC5031}
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "pjsua_lib", "pjsip\build\pjsua_lib.vcproj", "{9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "sample_debug", "pjsip-apps\build\sample_debug.vcproj", "{A0F1AA62-0F6F-420D-B09A-AC04B6862821}"
ProjectSection(ProjectDependencies) = postProject
{2BB84911-C1B4-4747-B93D-36AA82CC5031} = {2BB84911-C1B4-4747-B93D-36AA82CC5031}
{FE07F272-AE7F-4549-9E9F-EF9B80CB1693} = {FE07F272-AE7F-4549-9E9F-EF9B80CB1693}
{7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65} = {7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65}
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9} = {4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}
{6794B975-4E84-4F49-B2DC-C31F2224E03E} = {6794B975-4E84-4F49-B2DC-C31F2224E03E}
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA} = {3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}
{855DC8C0-D3E9-4A2E-AE47-116605A7BC9B} = {855DC8C0-D3E9-4A2E-AE47-116605A7BC9B}
{4B5945CD-0CB3-49AA-A7FF-7612D93F82C0} = {4B5945CD-0CB3-49AA-A7FF-7612D93F82C0}
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D} = {B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}
{E53AA5FF-B737-40AA-BD13-387EFA99023D} = {E53AA5FF-B737-40AA-BD13-387EFA99023D}
{9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37} = {9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37}
{DA0E03ED-53A7-4050-8A85-90541C5509F8} = {DA0E03ED-53A7-4050-8A85-90541C5509F8}
{B8719FD5-E8A6-4A36-943C-891D07F5DD21} = {B8719FD5-E8A6-4A36-943C-891D07F5DD21}
{4B5945CD-0CB3-49AA-A7FF-7612D93F82C0} = {4B5945CD-0CB3-49AA-A7FF-7612D93F82C0}
{855DC8C0-D3E9-4A2E-AE47-116605A7BC9B} = {855DC8C0-D3E9-4A2E-AE47-116605A7BC9B}
{7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65} = {7FDE3880-A4AB-49E3-B439-EBEF0A0C7A65}
{FE07F272-AE7F-4549-9E9F-EF9B80CB1693} = {FE07F272-AE7F-4549-9E9F-EF9B80CB1693}
{2BB84911-C1B4-4747-B93D-36AA82CC5031} = {2BB84911-C1B4-4747-B93D-36AA82CC5031}
EndProjectSection
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "samples", "pjsip-apps\build\samples.vcproj", "{E378A1FC-0C9C-4462-860F-7E60BC1BF84E}"
@ -79,6 +89,16 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "test_pjsip", "pjsip\build\t
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "pjnath", "pjnath\build\pjnath.vcproj", "{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libgsmcodec", "third_party\build\gsm\libgsmcodec.vcproj", "{E53AA5FF-B737-40AA-BD13-387EFA99023D}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libilbccodec", "third_party\build\ilbc\libilbccodec.vcproj", "{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libportaudio", "third_party\build\portaudio\libportaudio.vcproj", "{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libresample", "third_party\build\resample\libresample.vcproj", "{6794B975-4E84-4F49-B2DC-C31F2224E03E}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libspeex", "third_party\build\speex\libspeex.vcproj", "{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}"
EndProject
Global
GlobalSection(SolutionConfigurationPlatforms) = preSolution
Debug|Win32 = Debug|Win32
@ -145,6 +165,26 @@ Global
{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}.Debug|Win32.Build.0 = Debug|Win32
{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}.Release|Win32.ActiveCfg = Release|Win32
{A5D9AA24-08ED-48B9-BD65-F0A25E96BFC4}.Release|Win32.Build.0 = Release|Win32
{E53AA5FF-B737-40AA-BD13-387EFA99023D}.Debug|Win32.ActiveCfg = Debug|Win32
{E53AA5FF-B737-40AA-BD13-387EFA99023D}.Debug|Win32.Build.0 = Debug|Win32
{E53AA5FF-B737-40AA-BD13-387EFA99023D}.Release|Win32.ActiveCfg = Release|Win32
{E53AA5FF-B737-40AA-BD13-387EFA99023D}.Release|Win32.Build.0 = Release|Win32
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}.Debug|Win32.ActiveCfg = Debug|Win32
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}.Debug|Win32.Build.0 = Debug|Win32
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}.Release|Win32.ActiveCfg = Release|Win32
{B5FE16F8-3EDB-4110-BD80-B4238CC01E8D}.Release|Win32.Build.0 = Release|Win32
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}.Debug|Win32.ActiveCfg = Debug|Win32
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}.Debug|Win32.Build.0 = Debug|Win32
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}.Release|Win32.ActiveCfg = Release|Win32
{3CF9FFA9-8387-4635-9D1B-E7944CBEFEAA}.Release|Win32.Build.0 = Release|Win32
{6794B975-4E84-4F49-B2DC-C31F2224E03E}.Debug|Win32.ActiveCfg = Debug|Win32
{6794B975-4E84-4F49-B2DC-C31F2224E03E}.Debug|Win32.Build.0 = Debug|Win32
{6794B975-4E84-4F49-B2DC-C31F2224E03E}.Release|Win32.ActiveCfg = Release|Win32
{6794B975-4E84-4F49-B2DC-C31F2224E03E}.Release|Win32.Build.0 = Release|Win32
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}.Debug|Win32.ActiveCfg = Debug|Win32
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}.Debug|Win32.Build.0 = Debug|Win32
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}.Release|Win32.ActiveCfg = Release|Win32
{4B059DBA-CD9C-4D0F-BE8C-FFB4EFD498E9}.Release|Win32.Build.0 = Release|Win32
EndGlobalSection
GlobalSection(SolutionProperties) = preSolution
HideSolutionNode = FALSE

30
pjproject.dsw
View File

@ -39,6 +39,18 @@ Package=<4>
###############################################################################
Project: "libresample"=".\THIRD_PARTY\BUILD\RESAMPLE\libresample.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "libspeex"=".\third_party\build\speex\libspeex.dsp" - Package Owner=<4>
Package=<5>
@ -227,6 +239,9 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name libspeex
End Project Dependency
Begin Project Dependency
Project_Dep_Name libresample
End Project Dependency
}}}
###############################################################################
@ -275,6 +290,21 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name pjsua_lib
End Project Dependency
Begin Project Dependency
Project_Dep_Name libgsmcodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libilbccodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libportaudio
End Project Dependency
Begin Project Dependency
Project_Dep_Name libresample
End Project Dependency
Begin Project Dependency
Project_Dep_Name libspeex
End Project Dependency
}}}
###############################################################################

12
pjsip-apps/build/Samples-vc.mak
View File

@ -18,9 +18,19 @@ PJSIP_UA_LIB = ..\..\pjsip\lib\pjsip-ua-$(TARGET)$(LIBEXT)
PJSIP_SIMPLE_LIB = ..\..\pjsip\lib\pjsip-simple-$(TARGET)$(LIBEXT)
PJSUA_LIB_LIB = ..\..\pjsip\lib\pjsua-lib-$(TARGET)$(LIBEXT)
GSM_LIB = ..\..\third_party\lib\libgsmcodec-$(TARGET)$(LIBEXT)
ILBC_LIB = ..\..\third_party\lib\libilbccodec-$(TARGET)$(LIBEXT)
PORTAUDIO_LIB = ..\..\third_party\lib\libportaudio-$(TARGET)$(LIBEXT)
RESAMPLE_LIB = ..\..\third_party\lib\libresample-$(TARGET)$(LIBEXT)
SPEEX_LIB = ..\..\third_party\lib\libspeex-$(TARGET)$(LIBEXT)
THIRD_PARTY_LIBS = $(GSM_LIB) $(ILBC_LIB) $(PORTAUDIO_LIB) $(RESAMPLE_LIB) \
$(SPEEX_LIB)
LIBS = $(PJSUA_LIB_LIB) $(PJSIP_UA_LIB) $(PJSIP_SIMPLE_LIB) \
$(PJSIP_LIB) $(PJMEDIA_CODEC_LIB) $(PJMEDIA_LIB) $(PJNATH_LIB) \
$(PJLIB_UTIL_LIB) $(PJLIB_LIB)
$(PJLIB_UTIL_LIB) $(PJLIB_LIB) \
$(THIRD_PARTY_LIBS)
CFLAGS = /DPJ_WIN32=1 /DPJ_M_I386=1 \
$(BUILD_FLAGS) \

105
pjsip-apps/build/pjsip_apps.dsw
View File

@ -3,6 +3,66 @@ Microsoft Developer Studio Workspace File, Format Version 6.00
###############################################################################
Project: "libgsmcodec"="..\..\third_party\build\gsm\libgsmcodec.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "libilbccodec"="..\..\third_party\build\ilbc\libilbccodec.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "libportaudio"="..\..\third_party\build\portaudio\libportaudio.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "libresample"="..\..\third_party\build\resample\libresample.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "libspeex"="..\..\third_party\build\speex\libspeex.dsp" - Package Owner=<4>
Package=<5>
{{{
}}}
Package=<4>
{{{
}}}
###############################################################################
Project: "pjlib"="..\..\pjlib\build\pjlib.dsp" - Package Owner=<4>
Package=<5>
@ -134,6 +194,21 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name pjnath
End Project Dependency
Begin Project Dependency
Project_Dep_Name libgsmcodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libilbccodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libportaudio
End Project Dependency
Begin Project Dependency
Project_Dep_Name libresample
End Project Dependency
Begin Project Dependency
Project_Dep_Name libspeex
End Project Dependency
}}}
###############################################################################
@ -185,6 +260,21 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name pjnath
End Project Dependency
Begin Project Dependency
Project_Dep_Name libgsmcodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libilbccodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libportaudio
End Project Dependency
Begin Project Dependency
Project_Dep_Name libresample
End Project Dependency
Begin Project Dependency
Project_Dep_Name libspeex
End Project Dependency
}}}
###############################################################################
@ -221,6 +311,21 @@ Package=<4>
Begin Project Dependency
Project_Dep_Name pjsua_lib
End Project Dependency
Begin Project Dependency
Project_Dep_Name libgsmcodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libilbccodec
End Project Dependency
Begin Project Dependency
Project_Dep_Name libportaudio
End Project Dependency
Begin Project Dependency
Project_Dep_Name libresample
End Project Dependency
Begin Project Dependency
Project_Dep_Name libspeex
End Project Dependency
}}}
###############################################################################

2
pjsip-apps/build/pjsua.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjsua"
ProjectGUID="{19EBB464-60E7-47C5-96D8-E012B8CF3A41}"
ProjectGUID="{8310649E-A25E-4AF0-91E8-9E3CC659BB89}"
RootNamespace="pjsua"
>
<Platforms>

2
pjsip-apps/build/sample_debug.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="sample_debug"
ProjectGUID="{E13CECAA-8A8E-48CC-841F-5C4B845F0ECB}"
ProjectGUID="{A0F1AA62-0F6F-420D-B09A-AC04B6862821}"
RootNamespace="sample_debug"
>
<Platforms>

2
pjsip-apps/build/samples.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="samples"
ProjectGUID="{3CE5B675-EF25-434E-8AC2-23F0D76D1719}"
ProjectGUID="{E378A1FC-0C9C-4462-860F-7E60BC1BF84E}"
RootNamespace="samples"
Keyword="MakeFileProj"
>

4
pjsip-apps/src/samples/pjsip-perf.c
View File

@ -751,13 +751,13 @@ static pj_status_t create_app(void)
*/
static pj_status_t init_sip()
{
pj_status_t status;
pj_status_t status = -1;
/* Add UDP/TCP transport. */
{
pj_sockaddr_in addr;
pjsip_host_port addrname;
const char *transport_type;
const char *transport_type = NULL;
pj_bzero(&addr, sizeof(addr));
addr.sin_family = PJ_AF_INET;

2
pjsip/build/pjsip_core.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjsip_core"
ProjectGUID="{98E41390-0F17-4501-BB27-B7A933D06806}"
ProjectGUID="{2BB84911-C1B4-4747-B93D-36AA82CC5031}"
>
<Platforms>
<Platform

2
pjsip/build/pjsip_simple.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjsip_simple"
ProjectGUID="{F40BFAB7-A013-4B6F-9AC4-676199004F0F}"
ProjectGUID="{4B5945CD-0CB3-49AA-A7FF-7612D93F82C0}"
>
<Platforms>
<Platform

2
pjsip/build/pjsip_ua.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjsip_ua"
ProjectGUID="{5672ED37-5A78-4393-94D0-7CFED60DCC44}"
ProjectGUID="{B8719FD5-E8A6-4A36-943C-891D07F5DD21}"
>
<Platforms>
<Platform

2
pjsip/build/pjsua_lib.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="pjsua_lib"
ProjectGUID="{8D13F110-93F9-4AE7-A717-DF25111B05DA}"
ProjectGUID="{9CA0FDFB-2172-41FC-B7F1-5CE915EDCB37}"
RootNamespace="pjsua_lib"
>
<Platforms>

2
pjsip/build/test_pjsip.vcproj
View File

@ -3,7 +3,7 @@
ProjectType="Visual C++"
Version="8.00"
Name="test_pjsip"
ProjectGUID="{AF5815F7-AACB-4798-8FBD-37BF18CAC15C}"
ProjectGUID="{B3F7D4E9-702F-4EB4-ADA8-098D0A83D770}"
>
<Platforms>
<Platform

13
pjsip/include/pjsua-lib/pjsua.h
View File

@ -3916,21 +3916,18 @@ PJ_DECL(pj_status_t) pjsua_player_destroy(pjsua_player_id id);
/**
* Create a file recorder, and automatically connect this recorder to
* the conference bridge. The recorder currently supports recording WAV file,
* and on Windows, MP3 file. The type of the recorder to use is determined
* by the extension of the file (e.g. ".wav" or ".mp3").
* the conference bridge. The recorder currently supports recording WAV file.
* The type of the recorder to use is determined by the extension of the file
* (e.g. ".wav").
*
* @param filename Output file name. The function will determine the
* default format to be used based on the file extension.
* Currently ".wav" is supported on all platforms, and
* also ".mp3" is support on Windows.
* Currently ".wav" is supported on all platforms.
* @param enc_type Optionally specify the type of encoder to be used to
* compress the media, if the file can support different
* encodings. This value must be zero for now.
* @param enc_param Optionally specify codec specific parameter to be
* passed to the file writer. For .MP3 recorder, this
* can point to pjmedia_mp3_encoder_option structure to
* specify additional settings for the .mp3 recorder.
* passed to the file writer.
* For .WAV recorder, this value must be NULL.
* @param max_size Maximum file size. Specify zero or -1 to remove size
* limitation. This value must be zero or -1 for now.

8
pjsip/src/pjsua-lib/pjsua_media.c
View File

@ -1370,14 +1370,8 @@ PJ_DEF(pj_status_t) pjsua_recorder_create( const pj_str_t *filename,
pjsua_var.mconf_cfg.samples_per_frame,
pjsua_var.mconf_cfg.bits_per_sample,
options, 0, &port);
} else if (file_format == FMT_MP3) {
status = pjmedia_mp3_writer_port_create(pjsua_var.pool, path,
pjsua_var.media_cfg.clock_rate,
pjsua_var.mconf_cfg.channel_count,
pjsua_var.mconf_cfg.samples_per_frame,
pjsua_var.mconf_cfg.bits_per_sample,
enc_param, &port);
} else {
PJ_UNUSED_ARG(enc_param);
port = NULL;
status = PJ_ENOTSUP;
}

1
third_party/README.txt vendored
View File

@ -16,3 +16,4 @@ portaudio: SVN -r1186
gsm: gsm-1.0.12
ilbc: from RFC
plc_steveu: Steve Underwood's PLC
resample: lib-resample, I think version 1.7

22
third_party/build/gsm/libgsmcodec.dsp vendored
View File

@ -32,17 +32,17 @@ RSC=rc.exe
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 0
# PROP BASE Output_Dir "output\gsm-i386-vc6-release"
# PROP BASE Intermediate_Dir "output\gsm-i386-vc6-release"
# PROP BASE Output_Dir "output\gsm-i386-win32-vc6-release"
# PROP BASE Intermediate_Dir "output\gsm-i386-win32-vc6-release"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 0
# PROP Output_Dir "output\gsm-i386-vc6-release"
# PROP Intermediate_Dir "output\gsm-i386-vc6-release"
# PROP Output_Dir "output\gsm-i386-win32-vc6-release"
# PROP Intermediate_Dir "output\gsm-i386-win32-vc6-release"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MT /GX /O2 /I "." /I "../../gsm/inc" /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MD /GX /O2 /I "." /I "../../gsm/inc" /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD BASE RSC /l 0x409 /d "NDEBUG"
# ADD RSC /l 0x409 /d "NDEBUG"
BSC32=bscmake.exe
@ -50,19 +50,19 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libgsmcodec-i386-vc6-release.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libgsmcodec-i386-win32-vc6-release.lib"
!ELSEIF "$(CFG)" == "libgsmcodec - Win32 Debug"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 1
# PROP BASE Output_Dir "output\gsm-i386-vc6-debug"
# PROP BASE Intermediate_Dir "output\gsm-i386-vc6-debug"
# PROP BASE Output_Dir "output\gsm-i386-win32-vc6-debug"
# PROP BASE Intermediate_Dir "output\gsm-i386-win32-vc6-debug"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 1
# PROP Output_Dir "output\gsm-i386-vc6-debug"
# PROP Intermediate_Dir "output\gsm-i386-vc6-debug"
# PROP Output_Dir "output\gsm-i386-win32-vc6-debug"
# PROP Intermediate_Dir "output\gsm-i386-win32-vc6-debug"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
@ -74,7 +74,7 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libgsmcodec-i386-vc6-debug.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libgsmcodec-i386-win32-vc6-debug.lib"
!ENDIF

26
third_party/build/ilbc/libilbccodec.dsp vendored
View File

@ -32,17 +32,17 @@ RSC=rc.exe
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 0
# PROP BASE Output_Dir "output\ilbc-i386-vc6-release"
# PROP BASE Intermediate_Dir "output\ilbc-i386-vc6-release"
# PROP BASE Output_Dir "output\ilbc-i386-win32-vc6-release"
# PROP BASE Intermediate_Dir "output\ilbc-i386-win32-vc6-release"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 0
# PROP Output_Dir "output\ilbc-i386-vc6-release"
# PROP Intermediate_Dir "output\ilbc-i386-vc6-release"
# PROP Output_Dir "output\ilbc-i386-win32-vc6-release"
# PROP Intermediate_Dir "output\ilbc-i386-win32-vc6-release"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MD /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD BASE RSC /l 0x409 /d "NDEBUG"
# ADD RSC /l 0x409 /d "NDEBUG"
BSC32=bscmake.exe
@ -50,19 +50,19 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libilbccodec-i386-vc6-release.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libilbccodec-i386-win32-vc6-release.lib"
!ELSEIF "$(CFG)" == "libilbccodec - Win32 Debug"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 1
# PROP BASE Output_Dir "output\ilbc-i386-vc6-debug"
# PROP BASE Intermediate_Dir "output\ilbc-i386-vc6-debug"
# PROP BASE Output_Dir "output\ilbc-i386-win32-vc6-debug"
# PROP BASE Intermediate_Dir "output\ilbc-i386-win32-vc6-debug"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 1
# PROP Output_Dir "output\ilbc-i386-vc6-debug"
# PROP Intermediate_Dir "output\ilbc-i386-vc6-debug"
# PROP Output_Dir "output\ilbc-i386-win32-vc6-debug"
# PROP Intermediate_Dir "output\ilbc-i386-win32-vc6-debug"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
@ -74,7 +74,7 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libilbccodec-i386-vc6-debug.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libilbccodec-i386-win32-vc6-debug.lib"
!ENDIF
@ -151,10 +151,6 @@ SOURCE=..\..\ilbc\iLBC_encode.c
# End Source File
# Begin Source File
SOURCE=..\..\ilbc\iLBC_test.c
# End Source File
# Begin Source File
SOURCE=..\..\ilbc\LPCdecode.c
# End Source File
# Begin Source File

22
third_party/build/portaudio/libportaudio.dsp vendored
View File

@ -32,17 +32,17 @@ RSC=rc.exe
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 0
# PROP BASE Output_Dir "output\pa-i386-vc6-release"
# PROP BASE Intermediate_Dir "output\pa-i386-vc6-release"
# PROP BASE Output_Dir "output\pa-i386-win32-vc6-release"
# PROP BASE Intermediate_Dir "output\pa-i386-win32-vc6-release"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 0
# PROP Output_Dir "output\pa-i386-vc6-release"
# PROP Intermediate_Dir "output\pa-i386-vc6-release"
# PROP Output_Dir "output\pa-i386-win32-vc6-release"
# PROP Intermediate_Dir "output\pa-i386-win32-vc6-release"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MT /W3 /GX /O2 /I "..\..\portaudio\src\common" /I "..\..\portaudio\include" /D "NDEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D "PA_ENABLE_DEBUG_OUTPUT" /D "_CRT_SECURE_NO_DEPRECATE" /D "PA_NO_ASIO" /YX /FD /c
# ADD CPP /nologo /MD /W3 /GX /O2 /I "..\..\portaudio\src\common" /I "..\..\portaudio\include" /D "NDEBUG" /D "WIN32" /D "_MBCS" /D "_LIB" /D "PA_ENABLE_DEBUG_OUTPUT" /D "_CRT_SECURE_NO_DEPRECATE" /D "PA_NO_ASIO" /YX /FD /c
# ADD BASE RSC /l 0x409 /d "NDEBUG"
# ADD RSC /l 0x409 /d "NDEBUG"
BSC32=bscmake.exe
@ -50,19 +50,19 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libportaudio-i386-vc6-release.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libportaudio-i386-win32-vc6-release.lib"
!ELSEIF "$(CFG)" == "libportaudio - Win32 Debug"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 1
# PROP BASE Output_Dir "output\pa-i386-vc6-debug"
# PROP BASE Intermediate_Dir "output\pa-i386-vc6-debug"
# PROP BASE Output_Dir "output\pa-i386-win32-vc6-debug"
# PROP BASE Intermediate_Dir "output\pa-i386-win32-vc6-debug"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 1
# PROP Output_Dir "output\pa-i386-vc6-debug"
# PROP Intermediate_Dir "output\pa-i386-vc6-debug"
# PROP Output_Dir "output\pa-i386-win32-vc6-debug"
# PROP Intermediate_Dir "output\pa-i386-win32-vc6-debug"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
@ -74,7 +74,7 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libportaudio-i386-vc6-debug.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libportaudio-i386-win32-vc6-debug.lib"
!ENDIF

3
third_party/build/resample/config.h vendored Normal file
View File

@ -0,0 +1,3 @@
#define RESAMPLE_HAS_SMALL_FILTER 1
#define RESAMPLE_HAS_LARGE_FILTER 1

118
third_party/build/resample/libresample.dsp vendored Normal file
View File

@ -0,0 +1,118 @@
# Microsoft Developer Studio Project File - Name="libresample" - Package Owner=<4>
# Microsoft Developer Studio Generated Build File, Format Version 6.00
# ** DO NOT EDIT **
# TARGTYPE "Win32 (x86) Static Library" 0x0104
CFG=libresample - Win32 Debug
!MESSAGE This is not a valid makefile. To build this project using NMAKE,
!MESSAGE use the Export Makefile command and run
!MESSAGE
!MESSAGE NMAKE /f "libresample.mak".
!MESSAGE
!MESSAGE You can specify a configuration when running NMAKE
!MESSAGE by defining the macro CFG on the command line. For example:
!MESSAGE
!MESSAGE NMAKE /f "libresample.mak" CFG="libresample - Win32 Debug"
!MESSAGE
!MESSAGE Possible choices for configuration are:
!MESSAGE
!MESSAGE "libresample - Win32 Release" (based on "Win32 (x86) Static Library")
!MESSAGE "libresample - Win32 Debug" (based on "Win32 (x86) Static Library")
!MESSAGE
# Begin Project
# PROP AllowPerConfigDependencies 0
# PROP Scc_ProjName ""
# PROP Scc_LocalPath ""
CPP=cl.exe
RSC=rc.exe
!IF "$(CFG)" == "libresample - Win32 Release"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 0
# PROP BASE Output_Dir "output\libresample-i386-win32-vc6-release"
# PROP BASE Intermediate_Dir "output\libresample-i386-win32-vc6-release"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 0
# PROP Output_Dir "output\libresample-i386-win32-vc6-release"
# PROP Intermediate_Dir "output\libresample-i386-win32-vc6-release"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD CPP /nologo /MD /W3 /GX /O2 /I "." /I "../../resample/include" /D "WIN32" /D "NDEBUG" /D "_MBCS" /D "_LIB" /YX /FD /c
# ADD BASE RSC /l 0x409 /d "NDEBUG"
# ADD RSC /l 0x409 /d "NDEBUG"
BSC32=bscmake.exe
# ADD BASE BSC32 /nologo
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libresample-i386-win32-vc6-release.lib"
!ELSEIF "$(CFG)" == "libresample - Win32 Debug"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 1
# PROP BASE Output_Dir "output\libresample-i386-win32-vc6-debug"
# PROP BASE Intermediate_Dir "output\libresample-i386-win32-vc6-debug"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 1
# PROP Output_Dir "output\libresample-i386-win32-vc6-debug"
# PROP Intermediate_Dir "output\libresample-i386-win32-vc6-debug"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
# ADD CPP /nologo /MTd /W3 /Gm /GX /ZI /Od /I "." /I "../../resample/include" /D "WIN32" /D "_DEBUG" /D "_MBCS" /D "_LIB" /YX /FD /GZ /c
# ADD BASE RSC /l 0x409 /d "_DEBUG"
# ADD RSC /l 0x409 /d "_DEBUG"
BSC32=bscmake.exe
# ADD BASE BSC32 /nologo
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libresample-i386-win32-vc6-debug.lib"
!ENDIF
# Begin Target
# Name "libresample - Win32 Release"
# Name "libresample - Win32 Debug"
# Begin Group "Source Files"
# PROP Default_Filter "cpp;c;cxx;rc;def;r;odl;idl;hpj;bat"
# Begin Source File
SOURCE=..\..\resample\src\largefilter.h
# End Source File
# Begin Source File
SOURCE=..\..\resample\src\resample.h
# End Source File
# Begin Source File
SOURCE=..\..\resample\src\resamplesubs.c
# End Source File
# Begin Source File
SOURCE=..\..\resample\src\smallfilter.h
# End Source File
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SOURCE=..\..\resample\src\stddefs.h
# End Source File
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# PROP Default_Filter "h;hpp;hxx;hm;inl"
# Begin Source File
SOURCE=..\..\resample\include\resamplesubs.h
# End Source File
# End Group
# End Target
# End Project

24
third_party/build/speex/libspeex.dsp vendored
View File

@ -32,17 +32,17 @@ RSC=rc.exe
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 0
# PROP BASE Output_Dir "output\libspeex-i386-vc6-release"
# PROP BASE Intermediate_Dir "output\libspeex-i386-vc6-release"
# PROP BASE Output_Dir "output\libspeex-i386-win32-vc6-release"
# PROP BASE Intermediate_Dir "output\libspeex-i386-win32-vc6-release"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 0
# PROP Output_Dir "output\libspeex-i386-vc6-release"
# PROP Intermediate_Dir "output\libspeex-i386-vc6-release"
# PROP Output_Dir "output\libspeex-i386-win32-vc6-release"
# PROP Intermediate_Dir "output\libspeex-i386-win32-vc6-release"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c
# ADD CPP /nologo /MT /GX /Ox /Ot /Og /Oi /Ob2 /I "." /I "../../speex/include" /I "../../../pjlib/include" /D inline=__inline /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVE_CONFIG_H" /YX /FD /c
# ADD CPP /nologo /MD /GX /Ox /Ot /Og /Oi /Ob2 /I "." /I "../../speex/include" /I "../../../pjlib/include" /D inline=__inline /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVE_CONFIG_H" /YX /FD /c
# ADD BASE RSC /l 0x809 /d "NDEBUG"
# ADD RSC /l 0x809 /d "NDEBUG"
BSC32=bscmake.exe
@ -50,23 +50,23 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libspeex-i386-vc6-release.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libspeex-i386-win32-vc6-release.lib"
!ELSEIF "$(CFG)" == "libspeex - Win32 Debug"
# PROP BASE Use_MFC 0
# PROP BASE Use_Debug_Libraries 1
# PROP BASE Output_Dir "output\libspeex-i386-vc6-debug"
# PROP BASE Intermediate_Dir "output\libspeex-i386-vc6-debug"
# PROP BASE Output_Dir "output\libspeex-i386-win32-vc6-debug"
# PROP BASE Intermediate_Dir "output\libspeex-i386-win32-vc6-debug"
# PROP BASE Target_Dir ""
# PROP Use_MFC 0
# PROP Use_Debug_Libraries 1
# PROP Output_Dir "output\libspeex-i386-vc6-debug"
# PROP Intermediate_Dir "output\libspeex-i386-vc6-debug"
# PROP Output_Dir "output\libspeex-i386-win32-vc6-debug"
# PROP Intermediate_Dir "output\libspeex-i386-win32-vc6-debug"
# PROP Target_Dir ""
F90=df.exe
# ADD BASE CPP /nologo /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /GZ /c
# ADD CPP /nologo /MTd /W3 /GX /ZI /Od /Ob0 /I "." /I "../../speex/include" /I "../../../pjlib/include" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "HAVE_CONFIG_H" /YX /FD /c
# ADD CPP /nologo /MTd /W3 /GX /ZI /Od /I "." /I "../../speex/include" /I "../../../pjlib/include" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "HAVE_CONFIG_H" /YX /FD /c
# ADD BASE RSC /l 0x809 /d "_DEBUG"
# ADD RSC /l 0x809 /d "_DEBUG"
BSC32=bscmake.exe
@ -74,7 +74,7 @@ BSC32=bscmake.exe
# ADD BSC32 /nologo
LIB32=link.exe -lib
# ADD BASE LIB32 /nologo
# ADD LIB32 /nologo /out:"..\..\lib\libspeex-i386-vc6-debug.lib"
# ADD LIB32 /nologo /out:"..\..\lib\libspeex-i386-win32-vc6-debug.lib"
!ENDIF

0
pjmedia/include/pjmedia/mp3_port.h → third_party/mp3/mp3_port.h vendored
View File

338
third_party/plc_steveu/plc_steveu.c vendored
View File

@ -1,338 +0,0 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* plc.c
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2004 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* This version may be optionally licenced under the GNU LGPL licence.
* This version is disclaimed to DIGIUM for inclusion in the Asterisk project.
*/
/*! \file */
#include <pjmedia/types.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <limits.h>
#include "plc_steveu.h"
#if !defined(FALSE)
#define FALSE 0
#endif
#if !defined(TRUE)
#define TRUE (!FALSE)
#endif
#ifndef INT16_MAX
#define INT16_MAX (32767)
#endif
#ifndef INT16_MIN
#define INT16_MIN (-32767-1)
#endif
//#define PJ_HAS_RINT 1
/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
#define ms_to_samples(t) (((t)*SAMPLE_RATE)/1000)
#if defined(PJ_HAS_RINT) && PJ_HAS_RINT!=0
#define RINT(d) rint(d)
#else
double RINT(double d)
{
double f = floor(d);
double c = ceil(d);
if (c-d > d-f)
return f;
else if (c-d < d-f)
return c;
else if (d >= 0) {
if (f/2==f)
return f;
else
return c;
} else {
if (c/2==c)
return c;
else
return f;
}
}
#endif
PJ_INLINE(pj_int16_t) fsaturate(double damp)
{
if (damp > 32767.0)
return INT16_MAX;
else if (damp < -32768.0)
return INT16_MIN;
else {
return (pj_int16_t) RINT(damp);
}
}
static void save_history(plc_state_t *s, pj_int16_t *buf, int len)
{
if (len >= PLC_HISTORY_LEN)
{
/* Just keep the last part of the new data, starting at the beginning of the buffer */
memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(pj_int16_t)*PLC_HISTORY_LEN);
s->buf_ptr = 0;
return;
}
if (s->buf_ptr + len > PLC_HISTORY_LEN)
{
/* Wraps around - must break into two sections */
memcpy(s->history + s->buf_ptr, buf, sizeof(pj_int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
len -= (PLC_HISTORY_LEN - s->buf_ptr);
memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(pj_int16_t)*len);
s->buf_ptr = len;
return;
}
/* Can use just one section */
memcpy(s->history + s->buf_ptr, buf, sizeof(pj_int16_t)*len);
s->buf_ptr += len;
}
/*- End of function --------------------------------------------------------*/
static void normalise_history(plc_state_t *s)
{
pj_int16_t tmp[PLC_HISTORY_LEN];
if (s->buf_ptr == 0)
return;
memcpy(tmp, s->history, sizeof(pj_int16_t)*s->buf_ptr);
memcpy(s->history, s->history + s->buf_ptr, sizeof(pj_int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(pj_int16_t)*s->buf_ptr);
s->buf_ptr = 0;
}
/*- End of function --------------------------------------------------------*/
PJ_INLINE(int) amdf_pitch(int min_pitch, int max_pitch, pj_int16_t amp[], int len)
{
int i;
int j;
int acc;
int min_acc;
int pitch;
pitch = min_pitch;
min_acc = INT_MAX;
for (i = max_pitch; i <= min_pitch; i++)
{
acc = 0;
for (j = 0; j < len; j++)
acc += abs(amp[i + j] - amp[j]);
if (acc < min_acc)
{
min_acc = acc;
pitch = i;
}
}
return pitch;
}
/*- End of function --------------------------------------------------------*/
int plc_rx(plc_state_t *s, pj_int16_t amp[], int len)
{
int i;
/*int overlap_len;*/
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
if (s->missing_samples)
{
/* Although we have a real signal, we need to smooth it to fit well
with the synthetic signal we used for the previous block */
/* The start of the real data is overlapped with the next 1/4 cycle
of the synthetic data. */
pitch_overlap = s->pitch >> 2;
if (pitch_overlap > len)
pitch_overlap = len;
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
if (gain < 0.0)
gain = 0.0;
new_step = 1.0/pitch_overlap;
old_step = new_step*gain;
new_weight = new_step;
old_weight = (1.0 - new_step)*gain;
for (i = 0; i < pitch_overlap; i++)
{
amp[i] = fsaturate(old_weight*s->pitchbuf[s->pitch_offset] + new_weight*amp[i]);
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->missing_samples = 0;
}
save_history(s, amp, len);
return len;
}
/*- End of function --------------------------------------------------------*/
int plc_fillin(plc_state_t *s, pj_int16_t amp[], int len)
{
/*pj_int16_t tmp[PLC_PITCH_OVERLAP_MAX];*/
int i;
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
pj_int16_t *orig_amp;
int orig_len;
orig_amp = amp;
orig_len = len;
if (s->missing_samples == 0)
{
/* As the gap in real speech starts we need to assess the last known pitch,
and prepare the synthetic data we will use for fill-in */
normalise_history(s);
s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
/* We overlap a 1/4 wavelength */
pitch_overlap = s->pitch >> 2;
/* Cook up a single cycle of pitch, using a single of the real signal with 1/4
cycle OLA'ed to make the ends join up nicely */
/* The first 3/4 of the cycle is a simple copy */
for (i = 0; i < s->pitch - pitch_overlap; i++)
s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
/* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
new_step = 1.0/pitch_overlap;
new_weight = new_step;
for ( ; i < s->pitch; i++)
{
s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i]*(1.0 - new_weight) + s->history[PLC_HISTORY_LEN - 2*s->pitch + i]*new_weight;
new_weight += new_step;
}
/* We should now be ready to fill in the gap with repeated, decaying cycles
of what is in pitchbuf */
/* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
it into the previous real data. To avoid the need to introduce a delay
in the stream, reverse the last 1/4 wavelength, and OLA with that. */
gain = 1.0;
new_step = 1.0/pitch_overlap;
old_step = new_step;
new_weight = new_step;
old_weight = 1.0 - new_step;
for (i = 0; i < pitch_overlap; i++)
{
amp[i] = fsaturate(old_weight*s->history[PLC_HISTORY_LEN - 1 - i] + new_weight*s->pitchbuf[i]);
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->pitch_offset = i;
}
else
{
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
i = 0;
}
for ( ; gain > 0.0 && i < len; i++)
{
amp[i] = (pj_int16_t)(s->pitchbuf[s->pitch_offset]*gain);
gain = gain - ATTENUATION_INCREMENT;
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
}
for ( ; i < len; i++)
amp[i] = 0;
s->missing_samples += orig_len;
save_history(s, amp, len);
return len;
}
/*- End of function --------------------------------------------------------*/
plc_state_t *plc_init(plc_state_t *s)
{
memset(s, 0, sizeof(*s));
return s;
}
/*- End of function --------------------------------------------------------*/
/*
* PJMEDIA specifics
*/
#include <pj/assert.h>
#include <pj/pool.h>
#include <pj/log.h>
#define THIS_FILE "plc_steveu.c"
struct steveu_plc
{
plc_state_t state;
unsigned samples_per_frame;
};
void* pjmedia_plc_steveu_create(pj_pool_t *pool, unsigned c, unsigned f)
{
struct steveu_plc *splc;
PJ_ASSERT_RETURN(c==8000, NULL);
PJ_UNUSED_ARG(c);
splc = pj_pool_alloc(pool, sizeof(struct steveu_plc));
plc_init(&splc->state);
splc->samples_per_frame = f;
return splc;
}
void pjmedia_plc_steveu_save(void *obj, pj_int16_t *samples)
{
struct steveu_plc *splc = obj;
plc_rx(&splc->state, samples, splc->samples_per_frame);
}
void pjmedia_plc_steveu_generate(void *obj, pj_int16_t *samples)
{
struct steveu_plc *splc = obj;
//PJ_LOG(5,(THIS_FILE, "PLC: generating lost frame"));
plc_fillin(&splc->state, samples, splc->samples_per_frame);
}
/*- End of file ------------------------------------------------------------*/

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third_party/plc_steveu/plc_steveu.h vendored
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@ -1,153 +0,0 @@
/*! \file
* \brief SpanDSP - a series of DSP components for telephony
*
* plc.h
*
* \author Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2004 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* This version may be optionally licenced under the GNU LGPL licence.
*
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
*/
#if !defined(_PLC_H_)
#define _PLC_H_
/*! \page plc_page Packet loss concealment
\section plc_page_sec_1 What does it do?
The packet loss concealment module provides a suitable synthetic fill-in signal,
to minimise the audible effect of lost packets in VoIP applications. It is not
tied to any particular codec, and could be used with almost any codec which does not
specify its own procedure for packet loss concealment.
Where a codec specific concealment procedure exists, the algorithm is usually built
around knowledge of the characteristics of the particular codec. It will, therefore,
generally give better results for that particular codec than this generic concealer will.
\section plc_page_sec_2 How does it work?
While good packets are being received, the plc_rx() routine keeps a record of the trailing
section of the known speech signal. If a packet is missed, plc_fillin() is called to produce
a synthetic replacement for the real speech signal. The average mean difference function
(AMDF) is applied to the last known good signal, to determine its effective pitch.
Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech
will be repeated over and over until the real speech resumes. However, several refinements
are needed to obtain smooth pleasant sounding results.
- The two ends of the stored cycle of speech will not always fit together smoothly. This can
cause roughness, or even clicks, at the joins between cycles. To soften this, the
1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last
1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e.
in total, the last 5/4 pitch periods of real speech are used).
- The start of the synthetic speech will not always fit together smoothly with the tail of
real speech passed on before the erasure was identified. Ideally, we would like to modify
the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However,
it is too late for that. We could have delayed the real speech a little, but that would
require more buffer manipulation, and hurt the efficiency of the no-lost-packets case
(which we hope is the dominant case). Instead we use a degenerate form of OLA to modify
the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed,
and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result
seems quite acceptable.
- As we progress into the erasure, the chances of the synthetic signal being anything like
correct steadily fall. Therefore, the volume of the synthesized signal is made to decay
linearly, such that after 50ms of missing audio it is reduced to silence.
- When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the
start of the real speech. If the erasure is small, this smoothes the transition. If the erasure
is long, and the synthetic signal has faded to zero, the blending softens the start up of the
real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset.
\section plc_page_sec_3 How do I use it?
Before audio is processed, call plc_init() to create an instance of the packet loss
concealer. For each received audio packet that is acceptable (i.e. not including those being
dropped for being too late) call plc_rx() to record the content of the packet. Note this may
modify the packet a little after a period of packet loss, to blend real synthetic data smoothly.
When a real packet is not available in time, call plc_fillin() to create a sythetic substitute.
That's it!
*/
/*! Minimum allowed pitch (66 Hz) */
#define PLC_PITCH_MIN 120
/*! Maximum allowed pitch (200 Hz) */
#define PLC_PITCH_MAX 40
/*! Maximum pitch OLA window */
#define PLC_PITCH_OVERLAP_MAX (PLC_PITCH_MIN >> 2)
/*! The length over which the AMDF function looks for similarity (20 ms) */
#define CORRELATION_SPAN 160
/*! History buffer length. The buffer much also be at leat 1.25 times
PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for
the pitch assessment. */
#define PLC_HISTORY_LEN (CORRELATION_SPAN + PLC_PITCH_MIN)
typedef struct
{
/*! Consecutive erased samples */
int missing_samples;
/*! Current offset into pitch period */
int pitch_offset;
/*! Pitch estimate */
int pitch;
/*! Buffer for a cycle of speech */
float pitchbuf[PLC_PITCH_MIN];
/*! History buffer */
pj_int16_t history[PLC_HISTORY_LEN];
/*! Current pointer into the history buffer */
int buf_ptr;
} plc_state_t;
#ifdef __cplusplus
extern "C" {
#endif
/*! Process a block of received audio samples.
\brief Process a block of received audio samples.
\param s The packet loss concealer context.
\param amp The audio sample buffer.
\param len The number of samples in the buffer.
\return The number of samples in the buffer. */
int plc_rx(plc_state_t *s, pj_int16_t amp[], int len);
/*! Fill-in a block of missing audio samples.
\brief Fill-in a block of missing audio samples.
\param s The packet loss concealer context.
\param amp The audio sample buffer.
\param len The number of samples to be synthesised.
\return The number of samples synthesized. */
int plc_fillin(plc_state_t *s, pj_int16_t amp[], int len);
/*! Process a block of received V.29 modem audio samples.
\brief Process a block of received V.29 modem audio samples.
\param s The packet loss concealer context.
\return A pointer to the he packet loss concealer context. */
plc_state_t *plc_init(plc_state_t *s);
#ifdef __cplusplus
}
#endif
#endif
/*- End of file ------------------------------------------------------------*/

463
third_party/resample/COPYING vendored Normal file
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@ -0,0 +1,463 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
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18
third_party/resample/include/resamplesubs.h vendored Normal file
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#ifndef __RESAMPLESUBS_H__
#define __RESAMPLESUBS_H__
typedef char RES_BOOL;
typedef short RES_HWORD;
typedef int RES_WORD;
typedef unsigned short RES_UHWORD;
typedef unsigned int RES_UWORD;
int res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
double pFactor, RES_UHWORD nx);
int res_Resample(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_BOOL LargeF, RES_BOOL Interp);
int res_GetXOFF(double pFactor, RES_BOOL LargeF);
#endif

695
third_party/resample/resample.c vendored
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@ -1,695 +0,0 @@
/* $Id$ */
/*
* Based on:
* resample-1.8.tar.gz from the
* Digital Audio Resampling Home Page located at
* http://www-ccrma.stanford.edu/~jos/resample/.
*
* SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN
*
* Snippet from the resample.1 man page:
*
* HISTORY
*
* The first version of this software was written by Julius O. Smith III
* <jos@ccrma.stanford.edu> at CCRMA <http://www-ccrma.stanford.edu> in
* 1981. It was called SRCONV and was written in SAIL for PDP-10
* compatible machines. The algorithm was first published in
*
* Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate
* Conversion Method,'' Proceedings (2): 19.4.1-19.4.4, IEEE Conference
* on Acoustics, Speech, and Signal Processing, San Diego, March 1984.
*
* An expanded tutorial based on this paper is available at the Digital
* Audio Resampling Home Page given above.
*
* Circa 1988, the SRCONV program was translated from SAIL to C by
* Christopher Lee Fraley working with Roger Dannenberg at CMU.
*
* Since then, the C version has been maintained by jos.
*
* Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>.
*
* The resample program is free software distributed in accordance
* with the Lesser GNU Public License (LGPL). There is NO warranty; not
* even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*/
/* PJMEDIA modification:
* - remove resample(), just use SrcUp, SrcUD, and SrcLinear directly.
* - move FilterUp() and FilterUD() from filterkit.c
* - move stddefs.h and resample.h to this file.
* - const correctness.
*/
#include <pjmedia/resample.h>
#include <pjmedia/errno.h>
#include <pj/assert.h>
#include <pj/log.h>
#include <pj/pool.h>
#define THIS_FILE "resample.c"
/*
* Taken from stddefs.h
*/
#ifndef PI
#define PI (3.14159265358979232846)
#endif
#ifndef PI2
#define PI2 (6.28318530717958465692)
#endif
#define D2R (0.01745329348) /* (2*pi)/360 */
#define R2D (57.29577951) /* 360/(2*pi) */
#ifndef MAX
#define MAX(x,y) ((x)>(y) ?(x):(y))
#endif
#ifndef MIN
#define MIN(x,y) ((x)<(y) ?(x):(y))
#endif
#ifndef ABS
#define ABS(x) ((x)<0 ?(-(x)):(x))
#endif
#ifndef SGN
#define SGN(x) ((x)<0 ?(-1):((x)==0?(0):(1)))
#endif
typedef char RES_BOOL;
typedef short RES_HWORD;
typedef int RES_WORD;
typedef unsigned short RES_UHWORD;
typedef unsigned int RES_UWORD;
#define MAX_HWORD (32767)
#define MIN_HWORD (-32768)
#ifdef DEBUG
#define INLINE
#else
#define INLINE inline
#endif
/*
* Taken from resample.h
*
* The configuration constants below govern
* the number of bits in the input sample and filter coefficients, the
* number of bits to the right of the binary-point for fixed-point math, etc.
*
*/
/* Conversion constants */
#define Nhc 8
#define Na 7
#define Np (Nhc+Na)
#define Npc (1<<Nhc)
#define Amask ((1<<Na)-1)
#define Pmask ((1<<Np)-1)
#define Nh 16
#define Nb 16
#define Nhxn 14
#define Nhg (Nh-Nhxn)
#define NLpScl 13
/* Description of constants:
*
* Npc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Npc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Npc must be a power of 2 due to the details of the current
* implementation. The default value of 512 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit accuracy in filter coefficients.
*
* Nhc - is log base 2 of Npc.
*
* Na - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Np - is Na + Nhc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. Np must be less than 16 in
* this implementation.
*
* Nh - is the number of bits in the filter coefficients. The sum of Nh and
* the number of bits in the input data (typically 16) cannot exceed 32.
* Thus Nh should be 16. The largest filter coefficient should nearly
* fill 16 bits (32767).
*
* Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
* exceed 32.
*
* Nhxn - is the number of bits to right shift after multiplying each input
* sample times a filter coefficient. It can be as great as Nh and as
* small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
* accumulation. If Nhxn=0, the accumulation will soon overflow 32 bits.
*
* Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
*
* NLpScl - is the number of bits allocated to the unity-gain normalization
* factor. The output of the lowpass filter is multiplied by LpScl and
* then right-shifted NLpScl bits. To avoid overflow, we must have
* Nb+Nhg+NLpScl < 32.
*/
#ifdef _MSC_VER
# pragma warning(push, 3)
//# pragma warning(disable: 4245) // Conversion from uint to ushort
# pragma warning(disable: 4244) // Conversion from double to uint
# pragma warning(disable: 4146) // unary minus operator applied to unsigned type, result still unsigned
# pragma warning(disable: 4761) // integral size mismatch in argument; conversion supplied
#endif
#if defined(PJMEDIA_HAS_SMALL_FILTER) && PJMEDIA_HAS_SMALL_FILTER!=0
# include "smallfilter.h"
#else
# define SMALL_FILTER_NMULT 0
# define SMALL_FILTER_SCALE 0
# define SMALL_FILTER_NWING 0
# define SMALL_FILTER_IMP NULL
# define SMALL_FILTER_IMPD NULL
#endif
#if defined(PJMEDIA_HAS_LARGE_FILTER) && PJMEDIA_HAS_LARGE_FILTER!=0
# include "largefilter.h"
#else
# define LARGE_FILTER_NMULT 0
# define LARGE_FILTER_SCALE 0
# define LARGE_FILTER_NWING 0
# define LARGE_FILTER_IMP NULL
# define LARGE_FILTER_IMPD NULL
#endif
#undef INLINE
#define INLINE
#define HAVE_FILTER 0
#ifndef NULL
# define NULL 0
#endif
static INLINE RES_HWORD WordToHword(RES_WORD v, int scl)
{
RES_HWORD out;
RES_WORD llsb = (1<<(scl-1));
v += llsb; /* round */
v >>= scl;
if (v>MAX_HWORD) {
v = MAX_HWORD;
} else if (v < MIN_HWORD) {
v = MIN_HWORD;
}
out = (RES_HWORD) v;
return out;
}
/* Sampling rate conversion using linear interpolation for maximum speed.
*/
static int
SrcLinear(const RES_HWORD X[], RES_HWORD Y[], double pFactor, RES_UHWORD nx)
{
RES_HWORD iconst;
RES_UWORD time = 0;
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v,x1,x2;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
iconst = (time) & Pmask;
xp = &X[(time)>>Np]; /* Ptr to current input sample */
x1 = *xp++;
x2 = *xp;
x1 *= ((1<<Np)-iconst);
x2 *= iconst;
v = x1 + x2;
*Y++ = WordToHword(v,Np); /* Deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return number of output samples */
}
static RES_WORD FilterUp(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc)
{
const RES_HWORD *Hp;
const RES_HWORD *Hdp = NULL;
const RES_HWORD *End;
RES_HWORD a = 0;
RES_WORD v, t;
v=0;
Hp = &Imp[Ph>>Na];
End = &Imp[Nwing];
if (Interp) {
Hdp = &ImpD[Ph>>Na];
a = Ph & Amask;
}
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp += Npc; /* first sample, so we must also */
Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
}
}
if (Interp)
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
Hdp += Npc; /* Filter coeff differences step */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
static RES_WORD FilterUD(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc, RES_UHWORD dhb)
{
RES_HWORD a;
const RES_HWORD *Hp, *Hdp, *End;
RES_WORD v, t;
RES_UWORD Ho;
v=0;
Ho = (Ph*(RES_UWORD)dhb)>>Np;
End = &Imp[Nwing];
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
Ho += dhb; /* ...then we've already skipped the */
} /* first sample, so we must also */
/* skip ahead in Imp[] and ImpD[] */
if (Interp)
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
Hdp = &ImpD[Ho>>Na]; /* get interp (lower Na) bits from diff table*/
a = Ho & Amask; /* a is logically between 0 and 1 */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
/* Sampling rate up-conversion only subroutine;
* Slightly faster than down-conversion;
*/
static int SrcUp(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
/* Perform left-wing inner product */
v = 0;
v = FilterUp(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),-1);
/* Perform right-wing inner product */
v += FilterUp(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),1);
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* Sampling rate conversion subroutine */
static int SrcUD(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dh; /* Step through filter impulse response */
double dt; /* Step through input signal */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
RES_UWORD dhb, dtb; /* Fixed-point versions of Dh,Dt */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
dh = MIN(Npc, pFactor*Npc); /* Filter sampling period */
dhb = dh*(1<<Na) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
v = FilterUD(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),
-1, dhb); /* Perform left-wing inner product */
v += FilterUD(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),
1, dhb); /* Perform right-wing inner product */
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* ***************************************************************************
*
* PJMEDIA RESAMPLE
*
* ***************************************************************************
*/
struct pjmedia_resample
{
double factor; /* Conversion factor = rate_out / rate_in. */
pj_bool_t large_filter; /* Large filter? */
pj_bool_t high_quality; /* Not fast? */
unsigned xoff; /* History and lookahead size, in samples */
unsigned frame_size; /* Samples per frame. */
pj_int16_t *buffer; /* Input buffer. */
};
PJ_DEF(pj_status_t) pjmedia_resample_create( pj_pool_t *pool,
pj_bool_t high_quality,
pj_bool_t large_filter,
unsigned channel_count,
unsigned rate_in,
unsigned rate_out,
unsigned samples_per_frame,
pjmedia_resample **p_resample)
{
pjmedia_resample *resample;
PJ_ASSERT_RETURN(pool && p_resample && rate_in &&
rate_out && samples_per_frame, PJ_EINVAL);
resample = pj_pool_alloc(pool, sizeof(pjmedia_resample));
PJ_ASSERT_RETURN(resample, PJ_ENOMEM);
PJ_UNUSED_ARG(channel_count);
/*
* If we're downsampling, always use the fast algorithm since it seems
* to yield the same quality.
*/
if (rate_out < rate_in) {
//no this is not a good idea. It sounds pretty good with speech,
//but very poor with background noise etc.
//high_quality = 0;
}
#if !defined(PJMEDIA_HAS_LARGE_FILTER) || PJMEDIA_HAS_LARGE_FILTER==0
/*
* If large filter is excluded in the build, then prevent application
* from using it.
*/
if (high_quality && large_filter) {
large_filter = PJ_FALSE;
PJ_LOG(5,(THIS_FILE,
"Resample uses small filter because large filter is "
"disabled"));
}
#endif
#if !defined(PJMEDIA_HAS_SMALL_FILTER) || PJMEDIA_HAS_SMALL_FILTER==0
/*
* If small filter is excluded in the build and application wants to
* use it, then drop to linear conversion.
*/
if (high_quality && large_filter == 0) {
high_quality = PJ_FALSE;
PJ_LOG(4,(THIS_FILE,
"Resample uses linear because small filter is disabled"));
}
#endif
resample->factor = rate_out * 1.0 / rate_in;
resample->large_filter = large_filter;
resample->high_quality = high_quality;
resample->frame_size = samples_per_frame;
if (high_quality) {
unsigned size;
/* This is a bug in xoff calculation, thanks Stephane Lussier
* of Macadamian dot com.
* resample->xoff = large_filter ? 32 : 6;
*/
if (large_filter)
resample->xoff = (LARGE_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/resample->factor);
else
resample->xoff = (SMALL_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/resample->factor);
size = (samples_per_frame + 2*resample->xoff) * sizeof(pj_int16_t);
resample->buffer = pj_pool_alloc(pool, size);
PJ_ASSERT_RETURN(resample->buffer, PJ_ENOMEM);
pjmedia_zero_samples(resample->buffer, resample->xoff*2);
} else {
resample->xoff = 0;
}
*p_resample = resample;
PJ_LOG(5,(THIS_FILE, "resample created: %s qualiy, %s filter, in/out "
"rate=%d/%d",
(high_quality?"high":"low"),
(large_filter?"large":"small"),
rate_in, rate_out));
return PJ_SUCCESS;
}
PJ_DEF(void) pjmedia_resample_run( pjmedia_resample *resample,
const pj_int16_t *input,
pj_int16_t *output )
{
PJ_ASSERT_ON_FAIL(resample, return);
if (resample->high_quality) {
pj_int16_t *dst_buf;
const pj_int16_t *src_buf;
/* Okay chaps, here's how we do resampling.
*
* The original resample algorithm requires xoff samples *before* the
* input buffer as history, and another xoff samples *after* the
* end of the input buffer as lookahead. Since application can only
* supply framesize buffer on each run, PJMEDIA needs to arrange the
* buffer to meet these requirements.
*
* So here comes the trick.
*
* First of all, because of the history and lookahead requirement,
* resample->buffer need to accomodate framesize+2*xoff samples in its
* buffer. This is done when the buffer is created.
*
* On the first run, the input frame (supplied by application) is
* copied to resample->buffer at 2*xoff position. The first 2*xoff
* samples are initially zeroed (in the initialization). The resample
* algorithm then invoked at resample->buffer+xoff ONLY, thus giving
* it one xoff at the beginning as zero, and one xoff at the end
* as the end of the original input. The resample algorithm will see
* that the first xoff samples in the input as zero.
*
* So here's the layout of resample->buffer on the first run.
*
* run 0
* +------+------+--------------+
* | 0000 | 0000 | frame0... |
* +------+------+--------------+
* ^ ^ ^ ^
* 0 xoff 2*xoff size+2*xoff
*
* (Note again: resample algorithm is called at resample->buffer+xoff)
*
* At the end of the run, 2*xoff samples from the end of
* resample->buffer are copied to the beginning of resample->buffer.
* The first xoff part of this will be used as history for the next
* run, and the second xoff part of this is actually the start of
* resampling for the next run.
*
* And the first run completes, the function returns.
*
*
* On the next run, the input frame supplied by application is again
* copied at 2*xoff position in the resample->buffer, and the
* resample algorithm is again invoked at resample->buffer+xoff
* position. So effectively, the resample algorithm will start its
* operation on the last xoff from the previous frame, and gets the
* history from the last 2*xoff of the previous frame, and the look-
* ahead from the last xoff of current frame.
*
* So on this run, the buffer layout is:
*
* run 1
* +------+------+--------------+
* | frm0 | frm0 | frame1... |
* +------+------+--------------+
* ^ ^ ^ ^
* 0 xoff 2*xoff size+2*xoff
*
* As you can see from above diagram, the resampling algorithm is
* actually called from the last xoff part of previous frame (frm0).
*
* And so on the process continues for the next frame, and the next,
* and the next, ...
*
*/
dst_buf = resample->buffer + resample->xoff*2;
pjmedia_copy_samples(dst_buf, input, resample->frame_size);
if (resample->factor >= 1) {
if (resample->large_filter) {
SrcUp(resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
LARGE_FILTER_NWING, LARGE_FILTER_SCALE,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
PJ_TRUE);
} else {
SrcUp(resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
SMALL_FILTER_NWING, SMALL_FILTER_SCALE,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
PJ_TRUE);
}
} else {
if (resample->large_filter) {
SrcUD( resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
LARGE_FILTER_NWING,
LARGE_FILTER_SCALE * resample->factor + 0.5,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD,
PJ_TRUE);
} else {
SrcUD( resample->buffer + resample->xoff, output,
resample->factor, resample->frame_size,
SMALL_FILTER_NWING,
SMALL_FILTER_SCALE * resample->factor + 0.5,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD,
PJ_TRUE);
}
}
dst_buf = resample->buffer;
src_buf = input + resample->frame_size - resample->xoff*2;
pjmedia_copy_samples(dst_buf, src_buf, resample->xoff * 2);
} else {
SrcLinear( input, output, resample->factor, resample->frame_size);
}
}
PJ_DEF(unsigned) pjmedia_resample_get_input_size(pjmedia_resample *resample)
{
PJ_ASSERT_RETURN(resample != NULL, 0);
return resample->frame_size;
}
PJ_DEF(void) pjmedia_resample_destroy(pjmedia_resample *resample)
{
PJ_UNUSED_ARG(resample);
}

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/*
* The configuration constants below govern
* the number of bits in the input sample and filter coefficients, the
* number of bits to the right of the binary-point for fixed-point math, etc.
*
*/
/* Conversion constants */
#define Nhc 8
#define Na 7
#define Np (Nhc+Na)
#define Npc (1<<Nhc)
#define Amask ((1<<Na)-1)
#define Pmask ((1<<Np)-1)
#define Nh 16
#define Nb 16
#define Nhxn 14
#define Nhg (Nh-Nhxn)
#define NLpScl 13
/* Description of constants:
*
* Npc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Npc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Npc must be a power of 2 due to the details of the current
* implementation. The default value of 512 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit accuracy in filter coefficients.
*
* Nhc - is log base 2 of Npc.
*
* Na - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Np - is Na + Nhc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. Np must be less than 16 in
* this implementation.
*
* Nh - is the number of bits in the filter coefficients. The sum of Nh and
* the number of bits in the input data (typically 16) cannot exceed 32.
* Thus Nh should be 16. The largest filter coefficient should nearly
* fill 16 bits (32767).
*
* Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
* exceed 32.
*
* Nhxn - is the number of bits to right shift after multiplying each input
* sample times a filter coefficient. It can be as great as Nh and as
* small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
* accumulation. If Nhxn=0, the accumulation will soon overflow 32 bits.
*
* Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
*
* NLpScl - is the number of bits allocated to the unity-gain normalization
* factor. The output of the lowpass filter is multiplied by LpScl and
* then right-shifted NLpScl bits. To avoid overflow, we must have
* Nb+Nhg+NLpScl < 32.
*/

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/* $Id$ */
/*
* Digital Audio Resampling Home Page located at
* http://www-ccrma.stanford.edu/~jos/resample/.
*
* SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN
*
* Snippet from the resample.1 man page:
*
* HISTORY
*
* The first version of this software was written by Julius O. Smith III
* <jos@ccrma.stanford.edu> at CCRMA <http://www-ccrma.stanford.edu> in
* 1981. It was called SRCONV and was written in SAIL for PDP-10
* compatible machines. The algorithm was first published in
*
* Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate
* Conversion Method,'' Proceedings (2): 19.4.1-19.4.4, IEEE Conference
* on Acoustics, Speech, and Signal Processing, San Diego, March 1984.
*
* An expanded tutorial based on this paper is available at the Digital
* Audio Resampling Home Page given above.
*
* Circa 1988, the SRCONV program was translated from SAIL to C by
* Christopher Lee Fraley working with Roger Dannenberg at CMU.
*
* Since then, the C version has been maintained by jos.
*
* Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>.
*
* The resample program is free software distributed in accordance
* with the Lesser GNU Public License (LGPL). There is NO warranty; not
* even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*/
/* PJMEDIA modification:
* - remove resample(), just use SrcUp, SrcUD, and SrcLinear directly.
* - move FilterUp() and FilterUD() from filterkit.c
* - move stddefs.h and resample.h to this file.
* - const correctness.
*/
#include <resamplesubs.h>
#include "config.h"
#include "stddefs.h"
#include "resample.h"
#ifdef _MSC_VER
# pragma warning(push, 3)
//# pragma warning(disable: 4245) // Conversion from uint to ushort
# pragma warning(disable: 4244) // Conversion from double to uint
# pragma warning(disable: 4146) // unary minus operator applied to unsigned type, result still unsigned
# pragma warning(disable: 4761) // integral size mismatch in argument; conversion supplied
#endif
#if defined(RESAMPLE_HAS_SMALL_FILTER) && RESAMPLE_HAS_SMALL_FILTER!=0
# include "smallfilter.h"
#else
# define SMALL_FILTER_NMULT 0
# define SMALL_FILTER_SCALE 0
# define SMALL_FILTER_NWING 0
# define SMALL_FILTER_IMP NULL
# define SMALL_FILTER_IMPD NULL
#endif
#if defined(RESAMPLE_HAS_LARGE_FILTER) && RESAMPLE_HAS_LARGE_FILTER!=0
# include "largefilter.h"
#else
# define LARGE_FILTER_NMULT 0
# define LARGE_FILTER_SCALE 0
# define LARGE_FILTER_NWING 0
# define LARGE_FILTER_IMP NULL
# define LARGE_FILTER_IMPD NULL
#endif
#undef INLINE
#define INLINE
#define HAVE_FILTER 0
#ifndef NULL
# define NULL 0
#endif
static INLINE RES_HWORD WordToHword(RES_WORD v, int scl)
{
RES_HWORD out;
RES_WORD llsb = (1<<(scl-1));
v += llsb; /* round */
v >>= scl;
if (v>MAX_HWORD) {
v = MAX_HWORD;
} else if (v < MIN_HWORD) {
v = MIN_HWORD;
}
out = (RES_HWORD) v;
return out;
}
/* Sampling rate conversion using linear interpolation for maximum speed.
*/
static int
SrcLinear(const RES_HWORD X[], RES_HWORD Y[], double pFactor, RES_UHWORD nx)
{
RES_HWORD iconst;
RES_UWORD time = 0;
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v,x1,x2;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
iconst = (time) & Pmask;
xp = &X[(time)>>Np]; /* Ptr to current input sample */
x1 = *xp++;
x2 = *xp;
x1 *= ((1<<Np)-iconst);
x2 *= iconst;
v = x1 + x2;
*Y++ = WordToHword(v,Np); /* Deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return number of output samples */
}
static RES_WORD FilterUp(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc)
{
const RES_HWORD *Hp;
const RES_HWORD *Hdp = NULL;
const RES_HWORD *End;
RES_HWORD a = 0;
RES_WORD v, t;
v=0;
Hp = &Imp[Ph>>Na];
End = &Imp[Nwing];
if (Interp) {
Hdp = &ImpD[Ph>>Na];
a = Ph & Amask;
}
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp += Npc; /* first sample, so we must also */
Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
}
}
if (Interp)
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
Hdp += Npc; /* Filter coeff differences step */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while (Hp < End) {
t = *Hp; /* Get filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & (1<<(Nhxn-1))) /* Round, if needed */
t += (1<<(Nhxn-1));
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Hp += Npc; /* Filter coeff step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
static RES_WORD FilterUD(const RES_HWORD Imp[], const RES_HWORD ImpD[],
RES_UHWORD Nwing, RES_BOOL Interp,
const RES_HWORD *Xp, RES_HWORD Ph, RES_HWORD Inc, RES_UHWORD dhb)
{
RES_HWORD a;
const RES_HWORD *Hp, *Hdp, *End;
RES_WORD v, t;
RES_UWORD Ho;
v=0;
Ho = (Ph*(RES_UWORD)dhb)>>Np;
End = &Imp[Nwing];
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (Ph == 0) /* If the phase is zero... */
Ho += dhb; /* ...then we've already skipped the */
} /* first sample, so we must also */
/* skip ahead in Imp[] and ImpD[] */
if (Interp)
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
Hdp = &ImpD[Ho>>Na]; /* get interp (lower Na) bits from diff table*/
a = Ho & Amask; /* a is logically between 0 and 1 */
t += (((RES_WORD)*Hdp)*a)>>Na; /* t is now interp'd filter coeff */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
else
while ((Hp = &Imp[Ho>>Na]) < End) {
t = *Hp; /* Get IR sample */
t *= *Xp; /* Mult coeff by input sample */
if (t & 1<<(Nhxn-1)) /* Round, if needed */
t += 1<<(Nhxn-1);
t >>= Nhxn; /* Leave some guard bits, but come back some */
v += t; /* The filter output */
Ho += dhb; /* IR step */
Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */
}
return(v);
}
/* Sampling rate up-conversion only subroutine;
* Slightly faster than down-conversion;
*/
static int SrcUp(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dt; /* Step through input signal */
RES_UWORD dtb; /* Fixed-point version of Dt */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
/* Perform left-wing inner product */
v = 0;
v = FilterUp(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),-1);
/* Perform right-wing inner product */
v += FilterUp(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),1);
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
/* Sampling rate conversion subroutine */
static int SrcUD(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_UHWORD pNwing, RES_UHWORD pLpScl,
const RES_HWORD pImp[], const RES_HWORD pImpD[], RES_BOOL Interp)
{
const RES_HWORD *xp;
RES_HWORD *Ystart, *Yend;
RES_WORD v;
double dh; /* Step through filter impulse response */
double dt; /* Step through input signal */
RES_UWORD time = 0;
RES_UWORD endTime; /* When time reaches EndTime, return to user */
RES_UWORD dhb, dtb; /* Fixed-point versions of Dh,Dt */
dt = 1.0/pFactor; /* Output sampling period */
dtb = dt*(1<<Np) + 0.5; /* Fixed-point representation */
dh = MIN(Npc, pFactor*Npc); /* Filter sampling period */
dhb = dh*(1<<Na) + 0.5; /* Fixed-point representation */
Ystart = Y;
Yend = Ystart + (unsigned)(nx * pFactor);
endTime = time + (1<<Np)*(RES_WORD)nx;
while (time < endTime)
{
xp = &X[time>>Np]; /* Ptr to current input sample */
v = FilterUD(pImp, pImpD, pNwing, Interp, xp, (RES_HWORD)(time&Pmask),
-1, dhb); /* Perform left-wing inner product */
v += FilterUD(pImp, pImpD, pNwing, Interp, xp+1, (RES_HWORD)((-time)&Pmask),
1, dhb); /* Perform right-wing inner product */
v >>= Nhg; /* Make guard bits */
v *= pLpScl; /* Normalize for unity filter gain */
*Y++ = WordToHword(v,NLpScl); /* strip guard bits, deposit output */
time += dtb; /* Move to next sample by time increment */
}
return (Y - Ystart); /* Return the number of output samples */
}
int res_SrcLinear(const RES_HWORD X[], RES_HWORD Y[],
double pFactor, RES_UHWORD nx)
{
return SrcLinear(X, Y, pFactor, nx);
}
int res_Resample(const RES_HWORD X[], RES_HWORD Y[], double pFactor,
RES_UHWORD nx, RES_BOOL LargeF, RES_BOOL Interp)
{
if (pFactor >= 1) {
if (LargeF)
return SrcUp(X, Y, pFactor, nx,
LARGE_FILTER_NWING, LARGE_FILTER_SCALE,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD, Interp);
else
return SrcUp(X, Y, pFactor, nx,
SMALL_FILTER_NWING, SMALL_FILTER_SCALE,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD, Interp);
} else {
if (LargeF)
return SrcUD(X, Y, pFactor, nx,
LARGE_FILTER_NWING, LARGE_FILTER_SCALE * pFactor + 0.5,
LARGE_FILTER_IMP, LARGE_FILTER_IMPD, Interp);
else
return SrcUD(X, Y, pFactor, nx,
SMALL_FILTER_NWING, SMALL_FILTER_SCALE * pFactor + 0.5,
SMALL_FILTER_IMP, SMALL_FILTER_IMPD, Interp);
}
}
int res_GetXOFF(double pFactor, RES_BOOL LargeF)
{
if (LargeF)
return (LARGE_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/pFactor);
else
return (SMALL_FILTER_NMULT + 1) / 2.0 *
MAX(1.0, 1.0/pFactor);
}

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#ifndef PI
#define PI (3.14159265358979232846)
#endif
#ifndef PI2
#define PI2 (6.28318530717958465692)
#endif
#define D2R (0.01745329348) /* (2*pi)/360 */
#define R2D (57.29577951) /* 360/(2*pi) */
#ifndef MAX
#define MAX(x,y) ((x)>(y) ?(x):(y))
#endif
#ifndef MIN
#define MIN(x,y) ((x)<(y) ?(x):(y))
#endif
#ifndef ABS
#define ABS(x) ((x)<0 ?(-(x)):(x))
#endif
#ifndef SGN
#define SGN(x) ((x)<0 ?(-1):((x)==0?(0):(1)))
#endif
#define MAX_HWORD (32767)
#define MIN_HWORD (-32768)