diff options
44 files changed, 1458 insertions, 4515 deletions
diff --git a/pjlib-util/build/pjlib_util.vcproj b/pjlib-util/build/pjlib_util.vcproj index e104a346..7ca537c3 100644 --- a/pjlib-util/build/pjlib_util.vcproj +++ b/pjlib-util/build/pjlib_util.vcproj @@ -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>
diff --git a/pjlib/build/pjlib.vcproj b/pjlib/build/pjlib.vcproj index 223137dd..92183780 100644 --- a/pjlib/build/pjlib.vcproj +++ b/pjlib/build/pjlib.vcproj @@ -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>
diff --git a/pjmedia/build/pjmedia.dsp b/pjmedia/build/pjmedia.dsp index 6846ffe5..801b96a0 100644 --- a/pjmedia/build/pjmedia.dsp +++ b/pjmedia/build/pjmedia.dsp @@ -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
diff --git a/pjmedia/build/pjmedia.vcproj b/pjmedia/build/pjmedia.vcproj index 8a4dd066..03343089 100644 --- a/pjmedia/build/pjmedia.vcproj +++ b/pjmedia/build/pjmedia.vcproj @@ -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"
@@ -572,28 +572,6 @@ </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"
>
<FileConfiguration
@@ -682,7 +660,7 @@ </FileConfiguration>
</File>
<File
- RelativePath="..\src\pjmedia\plc_steveu.c"
+ RelativePath="..\src\pjmedia\port.c"
>
<FileConfiguration
Name="Release|Win32"
@@ -691,7 +669,6 @@ Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
- WarningLevel="2"
/>
</FileConfiguration>
<FileConfiguration
@@ -701,16 +678,11 @@ Name="VCCLCompilerTool"
AdditionalIncludeDirectories=""
PreprocessorDefinitions=""
- WarningLevel="2"
/>
</FileConfiguration>
</File>
<File
- RelativePath="..\src\pjmedia\plc_steveu.h"
- >
- </File>
- <File
- RelativePath="..\src\pjmedia\port.c"
+ RelativePath="..\src\pjmedia\resample_port.c"
>
<FileConfiguration
Name="Release|Win32"
@@ -732,48 +704,12 @@ </FileConfiguration>
</File>
<File
- RelativePath="..\src\pjmedia\resample.c"
+ RelativePath="..\src\pjmedia\resample_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"
+ RelativePath="..\src\pjmedia\resample_speex.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\rtcp.c"
@@ -1197,10 +1133,6 @@ >
</File>
<File
- RelativePath="..\include\pjmedia\mp3_port.h"
- >
- </File>
- <File
RelativePath="..\include\pjmedia\null_port.h"
>
</File>
@@ -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>
diff --git a/pjmedia/build/pjmedia_codec.vcproj b/pjmedia/build/pjmedia_codec.vcproj index 661f59e0..e8e293bf 100644 --- a/pjmedia/build/pjmedia_codec.vcproj +++ b/pjmedia/build/pjmedia_codec.vcproj @@ -3,7 +3,7 @@ ProjectType="Visual C++"
Version="8.00"
Name="pjmedia_codec"
- ProjectGUID="{0545EB9B-CE7C-4EDC-86E8-FB46A8DC1236}"
+ ProjectGUID="{855DC8C0-D3E9-4A2E-AE47-116605A7BC9B}"
>
<Platforms>
<Platform
@@ -41,7 +41,7 @@ <Tool
Name="VCCLCompilerTool"
Optimization="0"
- AdditionalIncludeDirectories="../include,../../pjlib/include,../src/pjmedia-codec"
+ AdditionalIncludeDirectories="../include,../../pjlib/include,../../third_party/speex/include"
PreprocessorDefinitions="_DEBUG;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1;HAVE_CONFIG_H"
MinimalRebuild="true"
BasicRuntimeChecks="3"
@@ -118,7 +118,7 @@ Name="VCCLCompilerTool"
Optimization="2"
InlineFunctionExpansion="1"
- AdditionalIncludeDirectories="../include,../../pjlib/include,../src/pjmedia-codec"
+ AdditionalIncludeDirectories="../include,../../pjlib/include,../../third_party/speex/include"
PreprocessorDefinitions="NDEBUG;WIN32;_LIB;PJ_WIN32=1;PJ_M_I386=1;HAVE_CONFIG_H"
StringPooling="true"
RuntimeLibrary="2"
@@ -296,2198 +296,6 @@ >
</File>
</Filter>
- <Filter
- Name="GSM 06.10 Library"
- >
- <File
- RelativePath="..\src\pjmedia-codec\gsm\add.c"
- >
- <FileConfiguration
- Name="Debug|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- </File>
- <File
- RelativePath="..\src\pjmedia-codec\gsm\code.c"
- >
- <FileConfiguration
- Name="Debug|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- </File>
- <File
- RelativePath="..\src\pjmedia-codec\gsm\config.h"
- >
- </File>
- <File
- RelativePath="..\src\pjmedia-codec\gsm\debug.c"
- >
- <FileConfiguration
- Name="Debug|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- </File>
- <File
- RelativePath="..\src\pjmedia-codec\gsm\decode.c"
- >
- <FileConfiguration
- Name="Debug|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories=""
- PreprocessorDefinitions=""
- WarningLevel="2"
- />
- </FileConfiguration>
- </File>
- <File
- RelativePath="..\src\pjmedia-codec\gsm\gsm.h"
- >
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<Globals>
</Globals>
diff --git a/pjmedia/include/pjmedia.h b/pjmedia/include/pjmedia.h index 6db9bbe7..4a104512 100644 --- a/pjmedia/include/pjmedia.h +++ b/pjmedia/include/pjmedia.h @@ -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> diff --git a/pjmedia/include/pjmedia/config.h b/pjmedia/include/pjmedia/config.h index 2d64e0ec..23e1318d 100644 --- a/pjmedia/include/pjmedia/config.h +++ b/pjmedia/include/pjmedia/config.h @@ -108,22 +108,55 @@ #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 + + +/** + * 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_SPEEX_RESAMPLE +# define PJMEDIA_HAS_SPEEX_RESAMPLE 0 +#endif + /** @@ -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 diff --git a/pjmedia/src/pjmedia-codec/gsm.c b/pjmedia/src/pjmedia-codec/gsm.c index 34f538b2..d21956ae 100644 --- a/pjmedia/src/pjmedia-codec/gsm.c +++ b/pjmedia/src/pjmedia-codec/gsm.c @@ -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 */ diff --git a/pjmedia/src/pjmedia/g711.c b/pjmedia/src/pjmedia/g711.c index 38306d0b..da9b7586 100644 --- a/pjmedia/src/pjmedia/g711.c +++ b/pjmedia/src/pjmedia/g711.c @@ -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 */ diff --git a/pjmedia/src/pjmedia/resample_resample.c b/pjmedia/src/pjmedia/resample_resample.c index fbb4a727..0dc755d9 100644 --- a/pjmedia/src/pjmedia/resample_resample.c +++ b/pjmedia/src/pjmedia/resample_resample.c @@ -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> -#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; -} +#if defined(PJMEDIA_HAS_LIBRESAMPLE) && PJMEDIA_HAS_LIBRESAMPLE != 0 -/* 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 */ -} +#include <third_party/resample/include/resamplesubs.h> -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 */ -} +#define THIS_FILE "resample.c" -/* 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 */ + diff --git a/pjmedia/src/pjmedia/resample_speex.c b/pjmedia/src/pjmedia/resample_speex.c index 447d3f6a..fcc42115 100644 --- a/pjmedia/src/pjmedia/resample_speex.c +++ b/pjmedia/src/pjmedia/resample_speex.c @@ -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 */ + diff --git a/pjproject-vs8.sln b/pjproject-vs8.sln index b84f8c49..4ad7a91e 100644 --- a/pjproject-vs8.sln +++ b/pjproject-vs8.sln @@ -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
diff --git a/pjproject.dsw b/pjproject.dsw index 3b9cd1c1..f68a5350 100644 --- a/pjproject.dsw +++ b/pjproject.dsw @@ -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
}}}
###############################################################################
diff --git a/pjsip-apps/build/Samples-vc.mak b/pjsip-apps/build/Samples-vc.mak index 18d5d65c..bd8d6d4f 100644 --- a/pjsip-apps/build/Samples-vc.mak +++ b/pjsip-apps/build/Samples-vc.mak @@ -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) \ diff --git a/pjsip-apps/build/pjsip_apps.dsw b/pjsip-apps/build/pjsip_apps.dsw index d97b3c70..ebc52ccd 100644 --- a/pjsip-apps/build/pjsip_apps.dsw +++ b/pjsip-apps/build/pjsip_apps.dsw @@ -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
}}}
###############################################################################
diff --git a/pjsip-apps/build/pjsua.vcproj b/pjsip-apps/build/pjsua.vcproj index d11e57cc..ae598229 100644 --- a/pjsip-apps/build/pjsua.vcproj +++ b/pjsip-apps/build/pjsua.vcproj @@ -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>
diff --git a/pjsip-apps/build/sample_debug.vcproj b/pjsip-apps/build/sample_debug.vcproj index 39a09317..1fa16da9 100644 --- a/pjsip-apps/build/sample_debug.vcproj +++ b/pjsip-apps/build/sample_debug.vcproj @@ -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>
diff --git a/pjsip-apps/build/samples.vcproj b/pjsip-apps/build/samples.vcproj index 0d872689..986dacbd 100644 --- a/pjsip-apps/build/samples.vcproj +++ b/pjsip-apps/build/samples.vcproj @@ -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"
>
diff --git a/pjsip-apps/src/samples/pjsip-perf.c b/pjsip-apps/src/samples/pjsip-perf.c index 264c2065..e3d7fc43 100644 --- a/pjsip-apps/src/samples/pjsip-perf.c +++ b/pjsip-apps/src/samples/pjsip-perf.c @@ -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; diff --git a/pjsip/build/pjsip_core.vcproj b/pjsip/build/pjsip_core.vcproj index 8f9c92da..33fbd5c9 100644 --- a/pjsip/build/pjsip_core.vcproj +++ b/pjsip/build/pjsip_core.vcproj @@ -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
diff --git a/pjsip/build/pjsip_simple.vcproj b/pjsip/build/pjsip_simple.vcproj index 13ef2763..3beea61d 100644 --- a/pjsip/build/pjsip_simple.vcproj +++ b/pjsip/build/pjsip_simple.vcproj @@ -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
diff --git a/pjsip/build/pjsip_ua.vcproj b/pjsip/build/pjsip_ua.vcproj index 0e4531db..f3a1ea69 100644 --- a/pjsip/build/pjsip_ua.vcproj +++ b/pjsip/build/pjsip_ua.vcproj @@ -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
diff --git a/pjsip/build/pjsua_lib.vcproj b/pjsip/build/pjsua_lib.vcproj index 96d8605f..23f268d0 100644 --- a/pjsip/build/pjsua_lib.vcproj +++ b/pjsip/build/pjsua_lib.vcproj @@ -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>
diff --git a/pjsip/build/test_pjsip.vcproj b/pjsip/build/test_pjsip.vcproj index 3a19827e..91208c48 100644 --- a/pjsip/build/test_pjsip.vcproj +++ b/pjsip/build/test_pjsip.vcproj @@ -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
diff --git a/pjsip/include/pjsua-lib/pjsua.h b/pjsip/include/pjsua-lib/pjsua.h index 76a27d33..10c19290 100644 --- a/pjsip/include/pjsua-lib/pjsua.h +++ b/pjsip/include/pjsua-lib/pjsua.h @@ -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. diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c index 785fb18f..66efe68c 100644 --- a/pjsip/src/pjsua-lib/pjsua_media.c +++ b/pjsip/src/pjsua-lib/pjsua_media.c @@ -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; } diff --git a/third_party/README.txt b/third_party/README.txt index 9518bcfc..b53b7afc 100644 --- a/third_party/README.txt +++ b/third_party/README.txt @@ -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 diff --git a/third_party/build/gsm/libgsmcodec.dsp b/third_party/build/gsm/libgsmcodec.dsp index f7ca6f06..6f070cce 100644 --- a/third_party/build/gsm/libgsmcodec.dsp +++ b/third_party/build/gsm/libgsmcodec.dsp @@ -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
diff --git a/third_party/build/ilbc/libilbccodec.dsp b/third_party/build/ilbc/libilbccodec.dsp index 8726aac6..679667b7 100644 --- a/third_party/build/ilbc/libilbccodec.dsp +++ b/third_party/build/ilbc/libilbccodec.dsp @@ -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
diff --git a/third_party/build/portaudio/libportaudio.dsp b/third_party/build/portaudio/libportaudio.dsp index 57a105e9..688f4131 100644 --- a/third_party/build/portaudio/libportaudio.dsp +++ b/third_party/build/portaudio/libportaudio.dsp @@ -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
diff --git a/third_party/build/resample/config.h b/third_party/build/resample/config.h new file mode 100644 index 00000000..fde89374 --- /dev/null +++ b/third_party/build/resample/config.h @@ -0,0 +1,3 @@ +#define RESAMPLE_HAS_SMALL_FILTER 1 +#define RESAMPLE_HAS_LARGE_FILTER 1 + diff --git a/third_party/build/resample/libresample.dsp b/third_party/build/resample/libresample.dsp new file mode 100644 index 00000000..82aab151 --- /dev/null +++ b/third_party/build/resample/libresample.dsp @@ -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"
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+# 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"
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+# 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
+# Begin Source File
+
+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
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+# End Target
+# End Project
diff --git a/third_party/build/speex/libspeex.dsp b/third_party/build/speex/libspeex.dsp index b36818f3..a769c1ed 100644 --- a/third_party/build/speex/libspeex.dsp +++ b/third_party/build/speex/libspeex.dsp @@ -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
diff --git a/pjmedia/include/pjmedia/mp3_port.h b/third_party/mp3/mp3_port.h index a4fc7e5f..a4fc7e5f 100644 --- a/pjmedia/include/pjmedia/mp3_port.h +++ b/third_party/mp3/mp3_port.h diff --git a/third_party/plc_steveu/plc_steveu.c b/third_party/plc_steveu/plc_steveu.c deleted file mode 100644 index 3326b748..00000000 --- a/third_party/plc_steveu/plc_steveu.c +++ /dev/null @@ -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 ------------------------------------------------------------*/ - diff --git a/third_party/plc_steveu/plc_steveu.h b/third_party/plc_steveu/plc_steveu.h deleted file mode 100644 index 483e774b..00000000 --- a/third_party/plc_steveu/plc_steveu.h +++ /dev/null @@ -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 ------------------------------------------------------------*/ - diff --git a/third_party/resample/COPYING b/third_party/resample/COPYING new file mode 100644 index 00000000..4ccd6ccf --- /dev/null +++ b/third_party/resample/COPYING @@ -0,0 +1,463 @@ + + GNU LESSER GENERAL PUBLIC LICENSE + Version 2.1, February 1999 + + Copyright (C) 1991, 1999 Free Software Foundation, Inc. + 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + Everyone is permitted to copy and distribute verbatim copies + of this license document, but changing it is not allowed. + +[This is the first released version of the Lesser GPL. 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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); -} - - diff --git a/third_party/resample/largefilter.h b/third_party/resample/src/largefilter.h index 1c3f7c0a..1c3f7c0a 100644 --- a/third_party/resample/largefilter.h +++ b/third_party/resample/src/largefilter.h diff --git a/third_party/resample/src/resample.h b/third_party/resample/src/resample.h new file mode 100644 index 00000000..35212512 --- /dev/null +++ b/third_party/resample/src/resample.h @@ -0,0 +1,67 @@ +/* + * 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. + */ + diff --git a/third_party/resample/src/resamplesubs.c b/third_party/resample/src/resamplesubs.c new file mode 100644 index 00000000..ae78add8 --- /dev/null +++ b/third_party/resample/src/resamplesubs.c @@ -0,0 +1,365 @@ +/* $Id$ */ +/* + * 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); +} + diff --git a/third_party/resample/smallfilter.h b/third_party/resample/src/smallfilter.h index 396fcb14..396fcb14 100644 --- a/third_party/resample/smallfilter.h +++ b/third_party/resample/src/smallfilter.h diff --git a/third_party/resample/src/stddefs.h b/third_party/resample/src/stddefs.h new file mode 100644 index 00000000..83497705 --- /dev/null +++ b/third_party/resample/src/stddefs.h @@ -0,0 +1,29 @@ +#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) + |