* #27232: jni: added pjproject checkout as regular git content

We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/.svn/pristine/42/427fc6955d98a733724206812ae953c31b4c9140.svn-base b/jni/pjproject-android/.svn/pristine/42/427fc6955d98a733724206812ae953c31b4c9140.svn-base
new file mode 100644
index 0000000..f0043ad
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/427fc6955d98a733724206812ae953c31b4c9140.svn-base
@@ -0,0 +1,36 @@
+#
+# PJLIB OS specific configuration for SunOS target. 
+#
+
+#
+# PJLIB_OBJS specified here are object files to be included in PJLIB
+# (the library) for this specific operating system. Object files common 
+# to all operating systems should go in Makefile instead.
+#
+export PJLIB_OBJS += 	addr_resolv_sock.o file_access_unistd.o \
+			file_io_ansi.o guid_simple.o \
+			log_writer_stdout.o os_core_unix.o \
+			os_error_unix.o os_time_unix.o \
+			os_timestamp_common.o os_timestamp_posix.o \
+			pool_policy_malloc.o sock_bsd.o sock_select.o
+
+export PJLIB_OBJS += ioqueue_select.o 
+#export PJLIB_OBJS += ioqueue_epoll.o
+
+#
+# TEST_OBJS are operating system specific object files to be included in
+# the test application.
+#
+export TEST_OBJS +=	main.o
+
+#
+# Additional LDFLAGS for pjlib-test
+#
+export TEST_LDFLAGS += -lm
+
+#
+# TARGETS are make targets in the Makefile, to be executed for this given
+# operating system.
+#
+export TARGETS	    =	pjlib pjlib-test
+
diff --git a/jni/pjproject-android/.svn/pristine/42/4281cddac55f9c81ac54ff43519e1f2dbcf62282.svn-base b/jni/pjproject-android/.svn/pristine/42/4281cddac55f9c81ac54ff43519e1f2dbcf62282.svn-base
new file mode 100644
index 0000000..2e080e3
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/4281cddac55f9c81ac54ff43519e1f2dbcf62282.svn-base
Binary files differ
diff --git a/jni/pjproject-android/.svn/pristine/42/429d1e3bd9d4c52fd0a6cca4633a76061815a78f.svn-base b/jni/pjproject-android/.svn/pristine/42/429d1e3bd9d4c52fd0a6cca4633a76061815a78f.svn-base
new file mode 100644
index 0000000..9f4df39
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/429d1e3bd9d4c52fd0a6cca4633a76061815a78f.svn-base
@@ -0,0 +1,118 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>

+<!DOCTYPE scenario SYSTEM "sipp.dtd">

+

+<!-- 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.,                                    -->

+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->

+<!--                                                                    -->

+<!--                 Sipp default 'uas' scenario.                       -->

+<!--                                                                    -->

+

+<scenario name="Basic UAS responder">

+  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->

+  <!-- are saved and used for following messages sent. Useful to test   -->

+  <!-- against stateful SIP proxies/B2BUAs.                             -->

+  <recv request="INVITE" crlf="true">

+  </recv>

+

+  <!-- The '[last_*]' keyword is replaced automatically by the          -->

+  <!-- specified header if it was present in the last message received  -->

+  <!-- (except if it was a retransmission). If the header was not       -->

+  <!-- present or if no message has been received, the '[last_*]'       -->

+  <!-- keyword is discarded, and all bytes until the end of the line    -->

+  <!-- are also discarded.                                              -->

+  <!--                                                                  -->

+  <!-- If the specified header was present several times in the         -->

+  <!-- message, all occurences are concatenated (CRLF seperated)        -->

+  <!-- to be used in place of the '[last_*]' keyword.                   -->

+

+  <send retrans="500">

+    <![CDATA[

+

+      SIP/2.0 422 Session Timer too small

+      [last_Via:]

+      [last_From:]

+      [last_To:];tag=[call_number]

+      [last_Call-ID:]

+      [last_CSeq:]

+      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 

+      Min-SE:  5400

+      Content-Length: 0

+

+	]]>

+  </send>

+

+  <recv request="ACK"

+        optional="true"

+        rtd="true"

+        crlf="true"> 

+  </recv> 

+ 

+

+  <recv request="INVITE" crlf="true">

+  </recv>

+

+  <send retrans="500"> 

+    <![CDATA[

+

+      SIP/2.0 200 OK

+      [last_Via:]

+      [last_From:]

+      [last_To:];tag=[call_number]

+      [last_Call-ID:]

+      [last_CSeq:]

+      Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

+      Allow-Events: telephone-event

+      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 

+      Supported: replaces

+      Session-Expires:  3600;refresher=uas

+      Require: timer

+      Content-Type: application/sdp

+      Content-Disposition: session;handling=required

+      Content-Length: [len]

+ 

+      v=0

+      o=Some-UserAgent 68 210 IN IP4 [local_ip]

+      s=SIP Call

+      c=IN IP4 [local_ip]

+      t=0 0

+      m=audio 17294 RTP/AVP 0 101

+      c=IN IP4 [local_ip]

+      a=rtpmap:0 PCMU/8000

+      a=rtpmap:101 telephone-event/8000

+      a=fmtp:101 0-16

+      a=ptime:20 

+

+    ]]>

+  </send> 

+  

+  <recv request="ACK"

+        rtd="true"

+        crlf="true"> 

+  </recv> 

+

+

+  <!-- Keep the call open for a while in case the 200 is lost to be     -->

+  <!-- able to retransmit it if we receive the BYE again.               -->

+  <pause milliseconds="4000"/>

+

+

+  <!-- definition of the response time repartition table (unit is ms)   -->

+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

+

+  <!-- definition of the call length repartition table (unit is ms)     -->

+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

+

+</scenario>

+

diff --git a/jni/pjproject-android/.svn/pristine/42/429fdf182f25d2deae23766b82a2b43b46ad5b54.svn-base b/jni/pjproject-android/.svn/pristine/42/429fdf182f25d2deae23766b82a2b43b46ad5b54.svn-base
new file mode 100644
index 0000000..a568462
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/429fdf182f25d2deae23766b82a2b43b46ad5b54.svn-base
@@ -0,0 +1,59 @@
+/*
+   Copyright (C) 2003 Commonwealth Scientific and Industrial Research
+   Organisation (CSIRO) Australia
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   - Neither the name of CSIRO Australia nor the names of its
+   contributors may be used to endorse or promote products derived from
+   this software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
+   PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE ORGANISATION OR
+   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef CONFIG_H
+#define CONFIG_H
+
+/* An inline macro is required for use of the inline keyword as not all C compilers support */
+/* inline.  It is officially C99 and C++ only */
+
+#ifdef __WINS__
+
+#define inline __inline
+
+/* Disable some pointless/stupid warnings */
+
+#pragma warning(disable: 4100) /* unreferenced formal parameter */
+#pragma warning(disable: 4127) /* conditional expression is constant */
+#pragma warning(disable: 4305) /* truncation from '...' to '...' */
+#pragma warning(disable: 4244) /* conversion from '...' to '...', possible loss of data */
+#pragma warning(disable: 4701) /* local variable may be be used without having been initialized */
+
+#endif /* ! __WINS__ */
+
+/* Use only fixed point arithmetic */
+
+#define FIXED_POINT 1
+#define USE_KISS_FFT
+#define EXPORT
+
+#endif /* ! CONFIG_H */
diff --git a/jni/pjproject-android/.svn/pristine/42/42b8f32678efcc3c229449896331a26b9e8660d0.svn-base b/jni/pjproject-android/.svn/pristine/42/42b8f32678efcc3c229449896331a26b9e8660d0.svn-base
new file mode 100644
index 0000000..54eef8d
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/42b8f32678efcc3c229449896331a26b9e8660d0.svn-base
@@ -0,0 +1,340 @@
+/* Copyright (C) 2007 Jean-Marc Valin
+      
+   File: speex_resampler.h
+   Resampling code
+      
+   The design goals of this code are:
+      - Very fast algorithm
+      - Low memory requirement
+      - Good *perceptual* quality (and not best SNR)
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are
+   met:
+
+   1. Redistributions of source code must retain the above copyright notice,
+   this list of conditions and the following disclaimer.
+
+   2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   3. The name of the author may not be used to endorse or promote products
+   derived from this software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
+   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
+   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
+   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef SPEEX_RESAMPLER_H
+#define SPEEX_RESAMPLER_H
+
+#ifdef OUTSIDE_SPEEX
+
+/********* WARNING: MENTAL SANITY ENDS HERE *************/
+
+/* If the resampler is defined outside of Speex, we change the symbol names so that 
+   there won't be any clash if linking with Speex later on. */
+
+/* #define RANDOM_PREFIX your software name here */
+#ifndef RANDOM_PREFIX
+#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
+#endif
+
+#define CAT_PREFIX2(a,b) a ## b
+#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
+      
+#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
+#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
+#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
+#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
+#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
+#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
+#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
+#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
+#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
+#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
+#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
+#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
+#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
+#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
+#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
+#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
+#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
+#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
+#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
+#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
+#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
+#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
+
+#define spx_int16_t short
+#define spx_int32_t int
+#define spx_uint16_t unsigned short
+#define spx_uint32_t unsigned int
+      
+#else /* OUTSIDE_SPEEX */
+
+#include "speex/speex_types.h"
+
+#endif /* OUTSIDE_SPEEX */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define SPEEX_RESAMPLER_QUALITY_MAX 10
+#define SPEEX_RESAMPLER_QUALITY_MIN 0
+#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
+#define SPEEX_RESAMPLER_QUALITY_VOIP 3
+#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
+
+enum {
+   RESAMPLER_ERR_SUCCESS         = 0,
+   RESAMPLER_ERR_ALLOC_FAILED    = 1,
+   RESAMPLER_ERR_BAD_STATE       = 2,
+   RESAMPLER_ERR_INVALID_ARG     = 3,
+   RESAMPLER_ERR_PTR_OVERLAP     = 4,
+   
+   RESAMPLER_ERR_MAX_ERROR
+};
+
+struct SpeexResamplerState_;
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+/** Create a new resampler with integer input and output rates.
+ * @param nb_channels Number of channels to be processed
+ * @param in_rate Input sampling rate (integer number of Hz).
+ * @param out_rate Output sampling rate (integer number of Hz).
+ * @param quality Resampling quality between 0 and 10, where 0 has poor quality
+ * and 10 has very high quality.
+ * @return Newly created resampler state
+ * @retval NULL Error: not enough memory
+ */
+SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, 
+                                          spx_uint32_t in_rate, 
+                                          spx_uint32_t out_rate, 
+                                          int quality,
+                                          int *err);
+
+/** Create a new resampler with fractional input/output rates. The sampling 
+ * rate ratio is an arbitrary rational number with both the numerator and 
+ * denominator being 32-bit integers.
+ * @param nb_channels Number of channels to be processed
+ * @param ratio_num Numerator of the sampling rate ratio
+ * @param ratio_den Denominator of the sampling rate ratio
+ * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
+ * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
+ * @param quality Resampling quality between 0 and 10, where 0 has poor quality
+ * and 10 has very high quality.
+ * @return Newly created resampler state
+ * @retval NULL Error: not enough memory
+ */
+SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, 
+                                               spx_uint32_t ratio_num, 
+                                               spx_uint32_t ratio_den, 
+                                               spx_uint32_t in_rate, 
+                                               spx_uint32_t out_rate, 
+                                               int quality,
+                                               int *err);
+
+/** Destroy a resampler state.
+ * @param st Resampler state
+ */
+void speex_resampler_destroy(SpeexResamplerState *st);
+
+/** Resample a float array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param channel_index Index of the channel to process for the multi-channel 
+ * base (0 otherwise)
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the 
+ * number of samples processed
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written
+ */
+int speex_resampler_process_float(SpeexResamplerState *st, 
+                                   spx_uint32_t channel_index, 
+                                   const float *in, 
+                                   spx_uint32_t *in_len, 
+                                   float *out, 
+                                   spx_uint32_t *out_len);
+
+/** Resample an int array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param channel_index Index of the channel to process for the multi-channel 
+ * base (0 otherwise)
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written
+ */
+int speex_resampler_process_int(SpeexResamplerState *st, 
+                                 spx_uint32_t channel_index, 
+                                 const spx_int16_t *in, 
+                                 spx_uint32_t *in_len, 
+                                 spx_int16_t *out, 
+                                 spx_uint32_t *out_len);
+
+/** Resample an interleaved float array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed. This is all per-channel.
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written.
+ * This is all per-channel.
+ */
+int speex_resampler_process_interleaved_float(SpeexResamplerState *st, 
+                                               const float *in, 
+                                               spx_uint32_t *in_len, 
+                                               float *out, 
+                                               spx_uint32_t *out_len);
+
+/** Resample an interleaved int array. The input and output buffers must *not* overlap.
+ * @param st Resampler state
+ * @param in Input buffer
+ * @param in_len Number of input samples in the input buffer. Returns the number
+ * of samples processed. This is all per-channel.
+ * @param out Output buffer
+ * @param out_len Size of the output buffer. Returns the number of samples written.
+ * This is all per-channel.
+ */
+int speex_resampler_process_interleaved_int(SpeexResamplerState *st, 
+                                             const spx_int16_t *in, 
+                                             spx_uint32_t *in_len, 
+                                             spx_int16_t *out, 
+                                             spx_uint32_t *out_len);
+
+/** Set (change) the input/output sampling rates (integer value).
+ * @param st Resampler state
+ * @param in_rate Input sampling rate (integer number of Hz).
+ * @param out_rate Output sampling rate (integer number of Hz).
+ */
+int speex_resampler_set_rate(SpeexResamplerState *st, 
+                              spx_uint32_t in_rate, 
+                              spx_uint32_t out_rate);
+
+/** Get the current input/output sampling rates (integer value).
+ * @param st Resampler state
+ * @param in_rate Input sampling rate (integer number of Hz) copied.
+ * @param out_rate Output sampling rate (integer number of Hz) copied.
+ */
+void speex_resampler_get_rate(SpeexResamplerState *st, 
+                              spx_uint32_t *in_rate, 
+                              spx_uint32_t *out_rate);
+
+/** Set (change) the input/output sampling rates and resampling ratio 
+ * (fractional values in Hz supported).
+ * @param st Resampler state
+ * @param ratio_num Numerator of the sampling rate ratio
+ * @param ratio_den Denominator of the sampling rate ratio
+ * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
+ * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
+ */
+int speex_resampler_set_rate_frac(SpeexResamplerState *st, 
+                                   spx_uint32_t ratio_num, 
+                                   spx_uint32_t ratio_den, 
+                                   spx_uint32_t in_rate, 
+                                   spx_uint32_t out_rate);
+
+/** Get the current resampling ratio. This will be reduced to the least
+ * common denominator.
+ * @param st Resampler state
+ * @param ratio_num Numerator of the sampling rate ratio copied
+ * @param ratio_den Denominator of the sampling rate ratio copied
+ */
+void speex_resampler_get_ratio(SpeexResamplerState *st, 
+                               spx_uint32_t *ratio_num, 
+                               spx_uint32_t *ratio_den);
+
+/** Set (change) the conversion quality.
+ * @param st Resampler state
+ * @param quality Resampling quality between 0 and 10, where 0 has poor 
+ * quality and 10 has very high quality.
+ */
+int speex_resampler_set_quality(SpeexResamplerState *st, 
+                                 int quality);
+
+/** Get the conversion quality.
+ * @param st Resampler state
+ * @param quality Resampling quality between 0 and 10, where 0 has poor 
+ * quality and 10 has very high quality.
+ */
+void speex_resampler_get_quality(SpeexResamplerState *st, 
+                                 int *quality);
+
+/** Set (change) the input stride.
+ * @param st Resampler state
+ * @param stride Input stride
+ */
+void speex_resampler_set_input_stride(SpeexResamplerState *st, 
+                                      spx_uint32_t stride);
+
+/** Get the input stride.
+ * @param st Resampler state
+ * @param stride Input stride copied
+ */
+void speex_resampler_get_input_stride(SpeexResamplerState *st, 
+                                      spx_uint32_t *stride);
+
+/** Set (change) the output stride.
+ * @param st Resampler state
+ * @param stride Output stride
+ */
+void speex_resampler_set_output_stride(SpeexResamplerState *st, 
+                                      spx_uint32_t stride);
+
+/** Get the output stride.
+ * @param st Resampler state copied
+ * @param stride Output stride
+ */
+void speex_resampler_get_output_stride(SpeexResamplerState *st, 
+                                      spx_uint32_t *stride);
+
+/** Get the latency in input samples introduced by the resampler.
+ * @param st Resampler state
+ */
+int speex_resampler_get_input_latency(SpeexResamplerState *st);
+
+/** Get the latency in output samples introduced by the resampler.
+ * @param st Resampler state
+ */
+int speex_resampler_get_output_latency(SpeexResamplerState *st);
+
+/** Make sure that the first samples to go out of the resamplers don't have 
+ * leading zeros. This is only useful before starting to use a newly created 
+ * resampler. It is recommended to use that when resampling an audio file, as
+ * it will generate a file with the same length. For real-time processing,
+ * it is probably easier not to use this call (so that the output duration
+ * is the same for the first frame).
+ * @param st Resampler state
+ */
+int speex_resampler_skip_zeros(SpeexResamplerState *st);
+
+/** Reset a resampler so a new (unrelated) stream can be processed.
+ * @param st Resampler state
+ */
+int speex_resampler_reset_mem(SpeexResamplerState *st);
+
+/** Returns the English meaning for an error code
+ * @param err Error code
+ * @return English string
+ */
+const char *speex_resampler_strerror(int err);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/jni/pjproject-android/.svn/pristine/42/42c1c750dd66f8ab816414c759ec16eac4ea3e69.svn-base b/jni/pjproject-android/.svn/pristine/42/42c1c750dd66f8ab816414c759ec16eac4ea3e69.svn-base
new file mode 100644
index 0000000..6b83fd4
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/42c1c750dd66f8ab816414c759ec16eac4ea3e69.svn-base
@@ -0,0 +1 @@
+#include "../../../portaudio/src/hostapi/coreaudio/pa_mac_core_blocking.h"
diff --git a/jni/pjproject-android/.svn/pristine/42/42d736aafd19d099e8806b839080aca1ec6d3dcf.svn-base b/jni/pjproject-android/.svn/pristine/42/42d736aafd19d099e8806b839080aca1ec6d3dcf.svn-base
new file mode 100644
index 0000000..10116ff
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/42d736aafd19d099e8806b839080aca1ec6d3dcf.svn-base
@@ -0,0 +1,217 @@
+/* $Id$ */
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA 
+ */
+
+#include <pjmedia.h>
+#include <pjlib-util.h>
+#include <pjlib.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "util.h"
+
+
+/**
+ * \page page_pjmedia_samples_playfile_c Samples: Playing WAV File to Sound Device
+ *
+ * This is a very simple example to use the @ref PJMEDIA_FILE_PLAY and
+ * @ref PJMED_SND_PORT. In this example, we open both the file and sound
+ * device, and connect the two of them, and voila! Sound will be playing
+ * the contents of the file.
+ *
+ * @see page_pjmedia_samples_recfile_c
+ *
+ * This file is pjsip-apps/src/samples/playfile.c
+ *
+ * \includelineno playfile.c
+ */
+
+
+/*
+ * playfile.c
+ *
+ * PURPOSE:
+ *  Play a WAV file to sound player device.
+ *
+ * USAGE:
+ *  playfile FILE.WAV
+ *
+ *  The WAV file could have mono or stereo channels with arbitrary
+ *  sampling rate, but MUST contain uncompressed (i.e. 16bit) PCM.
+ *
+ */
+
+
+/* For logging purpose. */
+#define THIS_FILE   "playfile.c"
+
+
+static const char *desc = 
+" FILE		    						    \n"
+"		    						    \n"
+"  playfile.c	    						    \n"
+"		    						    \n"
+" PURPOSE	    						    \n"
+"		    						    \n"
+"  Demonstrate how to play a WAV file.				    \n"
+"		    						    \n"
+" USAGE		    						    \n"
+"		    						    \n"
+"  playfile FILE.WAV						    \n"
+"		    						    \n"
+"  The WAV file could have mono or stereo channels with arbitrary   \n"
+"  sampling rate, but MUST contain uncompressed (i.e. 16bit) PCM.   \n";
+
+
+/*
+ * main()
+ */
+int main(int argc, char *argv[])
+{
+    pj_caching_pool cp;
+    pjmedia_endpt *med_endpt;
+    pj_pool_t *pool;
+    pjmedia_port *file_port;
+    pjmedia_snd_port *snd_port;
+    char tmp[10];
+    pj_status_t status;
+
+
+    if (argc != 2) {
+    	puts("Error: filename required");
+	puts(desc);
+	return 1;
+    }
+
+
+    /* Must init PJLIB first: */
+    status = pj_init();
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+    /* Must create a pool factory before we can allocate any memory. */
+    pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0);
+
+    /* 
+     * Initialize media endpoint.
+     * This will implicitly initialize PJMEDIA too.
+     */
+    status = pjmedia_endpt_create(&cp.factory, NULL, 1, &med_endpt);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+    /* Create memory pool for our file player */
+    pool = pj_pool_create( &cp.factory,	    /* pool factory	    */
+			   "wav",	    /* pool name.	    */
+			   4000,	    /* init size	    */
+			   4000,	    /* increment size	    */
+			   NULL		    /* callback on error    */
+			   );
+
+    /* Create file media port from the WAV file */
+    status = pjmedia_wav_player_port_create(  pool,	/* memory pool	    */
+					      argv[1],	/* file to play	    */
+					      20,	/* ptime.	    */
+					      0,	/* flags	    */
+					      0,	/* default buffer   */
+					      &file_port/* returned port    */
+					      );
+    if (status != PJ_SUCCESS) {
+	app_perror(THIS_FILE, "Unable to use WAV file", status);
+	return 1;
+    }
+
+    /* Create sound player port. */
+    status = pjmedia_snd_port_create_player( 
+		 pool,				    /* pool		    */
+		 -1,				    /* use default dev.	    */
+		 PJMEDIA_PIA_SRATE(&file_port->info),/* clock rate.	    */
+		 PJMEDIA_PIA_CCNT(&file_port->info),/* # of channels.	    */
+		 PJMEDIA_PIA_SPF(&file_port->info), /* samples per frame.   */
+		 PJMEDIA_PIA_BITS(&file_port->info),/* bits per sample.	    */
+		 0,				    /* options		    */
+		 &snd_port			    /* returned port	    */
+		 );
+    if (status != PJ_SUCCESS) {
+	app_perror(THIS_FILE, "Unable to open sound device", status);
+	return 1;
+    }
+
+    /* Connect file port to the sound player.
+     * Stream playing will commence immediately.
+     */
+    status = pjmedia_snd_port_connect( snd_port, file_port);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+
+    /* 
+     * File should be playing and looping now, using sound device's thread. 
+     */
+
+
+    /* Sleep to allow log messages to flush */
+    pj_thread_sleep(100);
+
+
+    printf("Playing %s..\n", argv[1]);
+    puts("");
+    puts("Press <ENTER> to stop playing and quit");
+
+    if (fgets(tmp, sizeof(tmp), stdin) == NULL) {
+	puts("EOF while reading stdin, will quit now..");
+    }
+
+    
+    /* Start deinitialization: */
+
+    /* Disconnect sound port from file port */
+    status = pjmedia_snd_port_disconnect(snd_port);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+    /* Without this sleep, Windows/DirectSound will repeteadly
+     * play the last frame during destroy.
+     */
+    pj_thread_sleep(100);
+
+    /* Destroy sound device */
+    status = pjmedia_snd_port_destroy( snd_port );
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+    /* Destroy file port */
+    status = pjmedia_port_destroy( file_port );
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+    /* Release application pool */
+    pj_pool_release( pool );
+
+    /* Destroy media endpoint. */
+    pjmedia_endpt_destroy( med_endpt );
+
+    /* Destroy pool factory */
+    pj_caching_pool_destroy( &cp );
+
+    /* Shutdown PJLIB */
+    pj_shutdown();
+
+
+    /* Done. */
+    return 0;
+}
+
diff --git a/jni/pjproject-android/.svn/pristine/42/42e74e5e5bf75b9fc4a9e17beafc3dbc7132ced3.svn-base b/jni/pjproject-android/.svn/pristine/42/42e74e5e5bf75b9fc4a9e17beafc3dbc7132ced3.svn-base
new file mode 100644
index 0000000..a398b8c
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/42/42e74e5e5bf75b9fc4a9e17beafc3dbc7132ced3.svn-base
@@ -0,0 +1,678 @@
+/* $Id$ */
+/*
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+#include <pjmedia-videodev/videodev_imp.h>
+#include <pjmedia-videodev/avi_dev.h>
+#include <pj/assert.h>
+#include <pj/log.h>
+#include <pj/os.h>
+#include <pj/rand.h>
+#include <pjmedia/vid_codec.h>
+
+#if defined(PJMEDIA_VIDEO_DEV_HAS_AVI) && PJMEDIA_VIDEO_DEV_HAS_AVI != 0 && \
+    defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+
+#define THIS_FILE		"avi_dev.c"
+#define DRIVER_NAME		"AVIDev"
+#define DEFAULT_CLOCK_RATE	90000
+#define DEFAULT_WIDTH		640
+#define DEFAULT_HEIGHT		480
+#define DEFAULT_FPS		25
+
+typedef struct avi_dev_strm avi_dev_strm;
+
+/* avi_ device info */
+struct avi_dev_info
+{
+    pjmedia_vid_dev_info	 info;
+
+    pj_pool_t			*pool;
+    pj_str_t			 fpath;
+    pj_str_t			 title;
+    pjmedia_avi_streams		*avi;
+    pjmedia_port                *vid;
+    avi_dev_strm		*strm;
+    pjmedia_vid_codec           *codec;
+    pj_uint8_t                  *enc_buf;
+    pj_size_t                    enc_buf_size;
+};
+
+/* avi_ factory */
+struct avi_factory
+{
+    pjmedia_vid_dev_factory	 base;
+    pj_pool_t			*pool;
+    pj_pool_factory		*pf;
+
+    unsigned			 dev_count;
+    struct avi_dev_info		*dev_info;
+};
+
+/* Video stream. */
+struct avi_dev_strm
+{
+    pjmedia_vid_dev_stream	     base;	    /**< Base stream	    */
+    pjmedia_vid_dev_param	     param;	    /**< Settings	    */
+    pj_pool_t			    *pool;          /**< Memory pool.       */
+    struct avi_dev_info		    *adi;
+
+    pjmedia_vid_dev_cb		     vid_cb;	    /**< Stream callback.   */
+    void			    *user_data;	    /**< Application data.  */
+};
+
+
+/* Prototypes */
+static pj_status_t avi_factory_init(pjmedia_vid_dev_factory *f);
+static pj_status_t avi_factory_destroy(pjmedia_vid_dev_factory *f);
+static pj_status_t avi_factory_refresh(pjmedia_vid_dev_factory *f);
+static unsigned    avi_factory_get_dev_count(pjmedia_vid_dev_factory *f);
+static pj_status_t avi_factory_get_dev_info(pjmedia_vid_dev_factory *f,
+					     unsigned index,
+					     pjmedia_vid_dev_info *info);
+static pj_status_t avi_factory_default_param(pj_pool_t *pool,
+                                              pjmedia_vid_dev_factory *f,
+					      unsigned index,
+					      pjmedia_vid_dev_param *param);
+static pj_status_t avi_factory_create_stream(
+					pjmedia_vid_dev_factory *f,
+					pjmedia_vid_dev_param *param,
+					const pjmedia_vid_dev_cb *cb,
+					void *user_data,
+					pjmedia_vid_dev_stream **p_vid_strm);
+
+static pj_status_t avi_dev_strm_get_param(pjmedia_vid_dev_stream *strm,
+					  pjmedia_vid_dev_param *param);
+static pj_status_t avi_dev_strm_get_cap(pjmedia_vid_dev_stream *strm,
+				        pjmedia_vid_dev_cap cap,
+				        void *value);
+static pj_status_t avi_dev_strm_set_cap(pjmedia_vid_dev_stream *strm,
+				        pjmedia_vid_dev_cap cap,
+				        const void *value);
+static pj_status_t avi_dev_strm_get_frame(pjmedia_vid_dev_stream *strm,
+                                          pjmedia_frame *frame);
+static pj_status_t avi_dev_strm_start(pjmedia_vid_dev_stream *strm);
+static pj_status_t avi_dev_strm_stop(pjmedia_vid_dev_stream *strm);
+static pj_status_t avi_dev_strm_destroy(pjmedia_vid_dev_stream *strm);
+
+static void reset_dev_info(struct avi_dev_info *adi);
+
+/* Operations */
+static pjmedia_vid_dev_factory_op factory_op =
+{
+    &avi_factory_init,
+    &avi_factory_destroy,
+    &avi_factory_get_dev_count,
+    &avi_factory_get_dev_info,
+    &avi_factory_default_param,
+    &avi_factory_create_stream,
+    &avi_factory_refresh
+};
+
+static pjmedia_vid_dev_stream_op stream_op =
+{
+    &avi_dev_strm_get_param,
+    &avi_dev_strm_get_cap,
+    &avi_dev_strm_set_cap,
+    &avi_dev_strm_start,
+    &avi_dev_strm_get_frame,
+    NULL,
+    &avi_dev_strm_stop,
+    &avi_dev_strm_destroy
+};
+
+
+/****************************************************************************
+ * Factory operations
+ */
+
+/* API */
+PJ_DEF(pj_status_t) pjmedia_avi_dev_create_factory(
+				    pj_pool_factory *pf,
+				    unsigned max_dev,
+				    pjmedia_vid_dev_factory **p_ret)
+{
+    struct avi_factory *cf;
+    pj_pool_t *pool;
+    pj_status_t status;
+
+    pool = pj_pool_create(pf, "avidevfc%p", 512, 512, NULL);
+    cf = PJ_POOL_ZALLOC_T(pool, struct avi_factory);
+    cf->pf = pf;
+    cf->pool = pool;
+    cf->dev_count = max_dev;
+    cf->base.op = &factory_op;
+
+    cf->dev_info = (struct avi_dev_info*)
+ 		   pj_pool_calloc(cf->pool, cf->dev_count,
+ 				  sizeof(struct avi_dev_info));
+
+    if (p_ret) {
+	*p_ret = &cf->base;
+    }
+
+    status = pjmedia_vid_register_factory(NULL, &cf->base);
+    if (status != PJ_SUCCESS)
+	return status;
+
+    PJ_LOG(4, (THIS_FILE, "AVI dev factory created with %d virtual device(s)",
+	       cf->dev_count));
+
+    return PJ_SUCCESS;
+}
+
+/* API: init factory */
+static pj_status_t avi_factory_init(pjmedia_vid_dev_factory *f)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+    unsigned i;
+
+    for (i=0; i<cf->dev_count; ++i) {
+	reset_dev_info(&cf->dev_info[i]);
+    }
+
+    return PJ_SUCCESS;
+}
+
+/* API: destroy factory */
+static pj_status_t avi_factory_destroy(pjmedia_vid_dev_factory *f)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+    pj_pool_t *pool = cf->pool;
+
+    cf->pool = NULL;
+    pj_pool_release(pool);
+
+    return PJ_SUCCESS;
+}
+
+/* API: refresh the list of devices */
+static pj_status_t avi_factory_refresh(pjmedia_vid_dev_factory *f)
+{
+    PJ_UNUSED_ARG(f);
+    return PJ_SUCCESS;
+}
+
+/* API: get number of devices */
+static unsigned avi_factory_get_dev_count(pjmedia_vid_dev_factory *f)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+    return cf->dev_count;
+}
+
+/* API: get device info */
+static pj_status_t avi_factory_get_dev_info(pjmedia_vid_dev_factory *f,
+					     unsigned index,
+					     pjmedia_vid_dev_info *info)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+
+    PJ_ASSERT_RETURN(index < cf->dev_count, PJMEDIA_EVID_INVDEV);
+
+    pj_memcpy(info, &cf->dev_info[index].info, sizeof(*info));
+
+    return PJ_SUCCESS;
+}
+
+/* API: create default device parameter */
+static pj_status_t avi_factory_default_param(pj_pool_t *pool,
+                                              pjmedia_vid_dev_factory *f,
+					      unsigned index,
+					      pjmedia_vid_dev_param *param)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+    struct avi_dev_info *di = &cf->dev_info[index];
+
+    PJ_ASSERT_RETURN(index < cf->dev_count, PJMEDIA_EVID_INVDEV);
+
+    PJ_UNUSED_ARG(pool);
+
+    pj_bzero(param, sizeof(*param));
+    param->dir = PJMEDIA_DIR_CAPTURE;
+    param->cap_id = index;
+    param->rend_id = PJMEDIA_VID_INVALID_DEV;
+    param->flags = PJMEDIA_VID_DEV_CAP_FORMAT;
+    param->clock_rate = DEFAULT_CLOCK_RATE;
+    pj_memcpy(&param->fmt, &di->info.fmt[0], sizeof(param->fmt));
+
+    return PJ_SUCCESS;
+}
+
+/* reset dev info */
+static void reset_dev_info(struct avi_dev_info *adi)
+{
+    /* Close avi streams */
+    if (adi->avi) {
+	unsigned i, cnt;
+
+	cnt = pjmedia_avi_streams_get_num_streams(adi->avi);
+	for (i=0; i<cnt; ++i) {
+	    pjmedia_avi_stream *as;
+
+	    as = pjmedia_avi_streams_get_stream(adi->avi, i);
+	    if (as) {
+		pjmedia_port *port;
+		port = pjmedia_avi_stream_get_port(as);
+		pjmedia_port_destroy(port);
+	    }
+	}
+	adi->avi = NULL;
+    }
+
+    if (adi->codec) {
+        pjmedia_vid_codec_close(adi->codec);
+        adi->codec = NULL;
+    }
+
+    if (adi->pool)
+	pj_pool_release(adi->pool);
+
+    pj_bzero(adi, sizeof(*adi));
+
+    /* Fill up with *dummy" device info */
+    pj_ansi_strncpy(adi->info.name, "AVI Player", sizeof(adi->info.name)-1);
+    pj_ansi_strncpy(adi->info.driver, DRIVER_NAME, sizeof(adi->info.driver)-1);
+    adi->info.dir = PJMEDIA_DIR_CAPTURE;
+    adi->info.has_callback = PJ_FALSE;
+}
+
+/* API: release resources */
+PJ_DEF(pj_status_t) pjmedia_avi_dev_free(pjmedia_vid_dev_index id)
+{
+    pjmedia_vid_dev_factory *f;
+    struct avi_factory *cf;
+    unsigned local_idx;
+    struct avi_dev_info *adi;
+    pj_status_t status;
+
+    /* Lookup the factory and local device index */
+    status = pjmedia_vid_dev_get_local_index(id, &f, &local_idx);
+    if (status != PJ_SUCCESS)
+	return status;
+
+    /* The factory must be AVI factory */
+    PJ_ASSERT_RETURN(f->op->init == &avi_factory_init, PJMEDIA_EVID_INVDEV);
+    cf = (struct avi_factory*)f;
+
+    /* Device index should be valid */
+    PJ_ASSERT_RETURN(local_idx <= cf->dev_count, PJ_EBUG);
+    adi = &cf->dev_info[local_idx];
+
+    /* Cannot configure if stream is running */
+    if (adi->strm)
+	return PJ_EBUSY;
+
+    /* Reset */
+    reset_dev_info(adi);
+    return PJ_SUCCESS;
+}
+
+/* API: get param */
+PJ_DEF(pj_status_t) pjmedia_avi_dev_get_param(pjmedia_vid_dev_index id,
+                                              pjmedia_avi_dev_param *prm)
+{
+    pjmedia_vid_dev_factory *f;
+    struct avi_factory *cf;
+    unsigned local_idx;
+    struct avi_dev_info *adi;
+    pj_status_t status;
+
+    /* Lookup the factory and local device index */
+    status = pjmedia_vid_dev_get_local_index(id, &f, &local_idx);
+    if (status != PJ_SUCCESS)
+	return status;
+
+    /* The factory must be factory */
+    PJ_ASSERT_RETURN(f->op->init == &avi_factory_init, PJMEDIA_EVID_INVDEV);
+    cf = (struct avi_factory*)f;
+
+    /* Device index should be valid */
+    PJ_ASSERT_RETURN(local_idx <= cf->dev_count, PJ_EBUG);
+    adi = &cf->dev_info[local_idx];
+
+    pj_bzero(prm, sizeof(*prm));
+    prm->path = adi->fpath;
+    prm->title = adi->title;
+    prm->avi_streams = adi->avi;
+
+    return PJ_SUCCESS;
+}
+
+PJ_DEF(void) pjmedia_avi_dev_param_default(pjmedia_avi_dev_param *p)
+{
+    pj_bzero(p, sizeof(*p));
+}
+
+/* API: configure the AVI */
+PJ_DEF(pj_status_t) pjmedia_avi_dev_alloc( pjmedia_vid_dev_factory *f,
+                                           pjmedia_avi_dev_param *p,
+                                           pjmedia_vid_dev_index *p_id)
+{
+    pjmedia_vid_dev_index id;
+    struct avi_factory *cf = (struct avi_factory*)f;
+    unsigned local_idx;
+    struct avi_dev_info *adi = NULL;
+    pjmedia_format avi_fmt;
+    const pjmedia_video_format_info *vfi;
+    pj_status_t status;
+
+    PJ_ASSERT_RETURN(f && p && p_id, PJ_EINVAL);
+
+    if (p_id)
+	*p_id = PJMEDIA_VID_INVALID_DEV;
+
+    /* Get a free dev */
+    for (local_idx=0; local_idx<cf->dev_count; ++local_idx) {
+	if (cf->dev_info[local_idx].avi == NULL) {
+	    adi = &cf->dev_info[local_idx];
+	    break;
+	}
+    }
+
+    if (!adi)
+	return PJ_ETOOMANY;
+
+    /* Convert local ID to global id */
+    status = pjmedia_vid_dev_get_global_index(&cf->base, local_idx, &id);
+    if (status != PJ_SUCCESS)
+	return status;
+
+    /* Reset */
+    if (adi->pool) {
+	pj_pool_release(adi->pool);
+    }
+    pj_bzero(adi, sizeof(*adi));
+
+    /* Reinit */
+    PJ_ASSERT_RETURN(p->path.slen, PJ_EINVAL);
+    adi->pool = pj_pool_create(cf->pf, "avidi%p", 512, 512, NULL);
+
+
+    /* Open the AVI */
+    pj_strdup_with_null(adi->pool, &adi->fpath, &p->path);
+    status = pjmedia_avi_player_create_streams(adi->pool, adi->fpath.ptr, 0,
+                                               &adi->avi);
+    if (status != PJ_SUCCESS) {
+	goto on_error;
+    }
+
+    adi->vid = pjmedia_avi_streams_get_stream_by_media(adi->avi, 0,
+                                                       PJMEDIA_TYPE_VIDEO);
+    if (!adi->vid) {
+	status = PJMEDIA_EVID_BADFORMAT;
+	PJ_LOG(4,(THIS_FILE, "Error: cannot find video in AVI %s",
+		adi->fpath.ptr));
+	goto on_error;
+    }
+
+    pjmedia_format_copy(&avi_fmt, &adi->vid->info.fmt);
+    vfi = pjmedia_get_video_format_info(NULL, avi_fmt.id);
+    /* Check whether the frame is encoded. */
+    if (!vfi || vfi->bpp == 0) {
+        /* Yes, prepare codec */
+        const pjmedia_vid_codec_info *codec_info;
+        pjmedia_vid_codec_param codec_param;
+	pjmedia_video_apply_fmt_param vafp;
+
+        /* Lookup codec */
+        status = pjmedia_vid_codec_mgr_get_codec_info2(NULL,
+                                                       avi_fmt.id,
+                                                       &codec_info);
+        if (status != PJ_SUCCESS || !codec_info)
+            goto on_error;
+
+        status = pjmedia_vid_codec_mgr_get_default_param(NULL, codec_info,
+                                                         &codec_param);
+        if (status != PJ_SUCCESS)
+            goto on_error;
+
+        /* Open codec */
+        status = pjmedia_vid_codec_mgr_alloc_codec(NULL, codec_info,
+                                                   &adi->codec);
+        if (status != PJ_SUCCESS)
+            goto on_error;
+
+        status = pjmedia_vid_codec_init(adi->codec, adi->pool);
+        if (status != PJ_SUCCESS)
+            goto on_error;
+
+        codec_param.dir = PJMEDIA_DIR_DECODING;
+        codec_param.packing = PJMEDIA_VID_PACKING_WHOLE;
+        status = pjmedia_vid_codec_open(adi->codec, &codec_param);
+        if (status != PJ_SUCCESS)
+            goto on_error;
+
+	/* Allocate buffer */
+        avi_fmt.id = codec_info->dec_fmt_id[0];
+        vfi = pjmedia_get_video_format_info(NULL, avi_fmt.id);
+	pj_bzero(&vafp, sizeof(vafp));
+	vafp.size = avi_fmt.det.vid.size;
+	status = vfi->apply_fmt(vfi, &vafp);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	adi->enc_buf = pj_pool_alloc(adi->pool, vafp.framebytes);
+	adi->enc_buf_size = vafp.framebytes;
+    }
+
+    /* Calculate title */
+    if (p->title.slen) {
+	pj_strdup_with_null(adi->pool, &adi->title, &p->title);
+    } else {
+	char *start = p->path.ptr + p->path.slen;
+	pj_str_t tmp;
+
+	while (start >= p->path.ptr) {
+	    if (*start == '/' || *start == '\\')
+		break;
+	    --start;
+	}
+	tmp.ptr = start + 1;
+	tmp.slen = p->path.ptr + p->path.slen - tmp.ptr;
+	pj_strdup_with_null(adi->pool, &adi->title, &tmp);
+    }
+
+    /* Init device info */
+    pj_ansi_strncpy(adi->info.name, adi->title.ptr, sizeof(adi->info.name)-1);
+    pj_ansi_strncpy(adi->info.driver, DRIVER_NAME, sizeof(adi->info.driver)-1);
+    adi->info.dir = PJMEDIA_DIR_CAPTURE;
+    adi->info.has_callback = PJ_FALSE;
+
+    adi->info.caps = PJMEDIA_VID_DEV_CAP_FORMAT;
+    adi->info.fmt_cnt = 1;
+    pjmedia_format_copy(&adi->info.fmt[0], &avi_fmt);
+
+    /* Set out vars */
+    if (p_id)
+	*p_id = id;
+    p->avi_streams = adi->avi;
+    if (p->title.slen == 0)
+	p->title = adi->title;
+
+    return PJ_SUCCESS;
+
+on_error:
+    if (adi->codec) {
+        pjmedia_vid_codec_close(adi->codec);
+        adi->codec = NULL;
+    }
+    if (adi->pool) {
+	pj_pool_release(adi->pool);
+	adi->pool = NULL;
+    }
+    pjmedia_avi_dev_free(id);
+    return status;
+}
+
+
+/* API: create stream */
+static pj_status_t avi_factory_create_stream(
+					pjmedia_vid_dev_factory *f,
+					pjmedia_vid_dev_param *param,
+					const pjmedia_vid_dev_cb *cb,
+					void *user_data,
+					pjmedia_vid_dev_stream **p_vid_strm)
+{
+    struct avi_factory *cf = (struct avi_factory*)f;
+    pj_pool_t *pool = NULL;
+    struct avi_dev_info *adi;
+    struct avi_dev_strm *strm;
+
+    PJ_ASSERT_RETURN(f && param && p_vid_strm, PJ_EINVAL);
+    PJ_ASSERT_RETURN(param->fmt.type == PJMEDIA_TYPE_VIDEO &&
+		     param->fmt.detail_type == PJMEDIA_FORMAT_DETAIL_VIDEO &&
+                     param->dir == PJMEDIA_DIR_CAPTURE,
+		     PJ_EINVAL);
+
+    /* Device must have been configured with pjmedia_avi_dev_set_param() */
+    adi = &cf->dev_info[param->cap_id];
+    PJ_ASSERT_RETURN(adi->avi != NULL, PJ_EINVALIDOP);
+
+    /* Cannot create while stream is already active */
+    PJ_ASSERT_RETURN(adi->strm==NULL, PJ_EINVALIDOP);
+
+    /* Create and initialize basic stream descriptor */
+    pool = pj_pool_create(cf->pf, "avidev%p", 512, 512, NULL);
+    PJ_ASSERT_RETURN(pool != NULL, PJ_ENOMEM);
+
+    strm = PJ_POOL_ZALLOC_T(pool, struct avi_dev_strm);
+    pj_memcpy(&strm->param, param, sizeof(*param));
+    strm->pool = pool;
+    pj_memcpy(&strm->vid_cb, cb, sizeof(*cb));
+    strm->user_data = user_data;
+    strm->adi = adi;
+
+    pjmedia_format_copy(&param->fmt, &adi->info.fmt[0]);
+
+    /* Done */
+    strm->base.op = &stream_op;
+    adi->strm = strm;
+    *p_vid_strm = &strm->base;
+
+    return PJ_SUCCESS;
+}
+
+/* API: Get stream info. */
+static pj_status_t avi_dev_strm_get_param(pjmedia_vid_dev_stream *s,
+					 pjmedia_vid_dev_param *pi)
+{
+    struct avi_dev_strm *strm = (struct avi_dev_strm*)s;
+
+    PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL);
+
+    pj_memcpy(pi, &strm->param, sizeof(*pi));
+
+    return PJ_SUCCESS;
+}
+
+/* API: get capability */
+static pj_status_t avi_dev_strm_get_cap(pjmedia_vid_dev_stream *s,
+				       pjmedia_vid_dev_cap cap,
+				       void *pval)
+{
+    struct avi_dev_strm *strm = (struct avi_dev_strm*)s;
+
+    PJ_UNUSED_ARG(strm);
+    PJ_UNUSED_ARG(cap);
+    PJ_UNUSED_ARG(pval);
+
+    PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
+
+    return PJMEDIA_EVID_INVCAP;
+}
+
+/* API: set capability */
+static pj_status_t avi_dev_strm_set_cap(pjmedia_vid_dev_stream *s,
+				       pjmedia_vid_dev_cap cap,
+				       const void *pval)
+{
+    struct avi_dev_strm *strm = (struct avi_dev_strm*)s;
+
+    PJ_UNUSED_ARG(strm);
+    PJ_UNUSED_ARG(cap);
+    PJ_UNUSED_ARG(pval);
+
+    PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
+
+    return PJMEDIA_EVID_INVCAP;
+}
+
+/* API: Get frame from stream */
+static pj_status_t avi_dev_strm_get_frame(pjmedia_vid_dev_stream *strm,
+                                         pjmedia_frame *frame)
+{
+    struct avi_dev_strm *stream = (struct avi_dev_strm*)strm;
+    
+    if (stream->adi->codec) {
+        pjmedia_frame enc_frame;
+        pj_status_t status;
+
+        enc_frame.buf = stream->adi->enc_buf;
+        enc_frame.size = stream->adi->enc_buf_size;
+        status = pjmedia_port_get_frame(stream->adi->vid, &enc_frame);
+        if (status != PJ_SUCCESS)
+            return status;
+
+        return pjmedia_vid_codec_decode(stream->adi->codec, 1, &enc_frame,
+                                        (unsigned)frame->size, frame);
+    } else {
+        return pjmedia_port_get_frame(stream->adi->vid, frame);
+    }
+}
+
+/* API: Start stream. */
+static pj_status_t avi_dev_strm_start(pjmedia_vid_dev_stream *strm)
+{
+    struct avi_dev_strm *stream = (struct avi_dev_strm*)strm;
+
+    PJ_UNUSED_ARG(stream);
+
+    PJ_LOG(4, (THIS_FILE, "Starting avi video stream"));
+
+    return PJ_SUCCESS;
+}
+
+/* API: Stop stream. */
+static pj_status_t avi_dev_strm_stop(pjmedia_vid_dev_stream *strm)
+{
+    struct avi_dev_strm *stream = (struct avi_dev_strm*)strm;
+
+    PJ_UNUSED_ARG(stream);
+
+    PJ_LOG(4, (THIS_FILE, "Stopping avi video stream"));
+
+    return PJ_SUCCESS;
+}
+
+
+/* API: Destroy stream. */
+static pj_status_t avi_dev_strm_destroy(pjmedia_vid_dev_stream *strm)
+{
+    struct avi_dev_strm *stream = (struct avi_dev_strm*)strm;
+
+    PJ_ASSERT_RETURN(stream != NULL, PJ_EINVAL);
+
+    avi_dev_strm_stop(strm);
+
+    stream->adi->strm = NULL;
+    stream->adi = NULL;
+    pj_pool_release(stream->pool);
+
+    return PJ_SUCCESS;
+}
+
+#endif	/* PJMEDIA_VIDEO_DEV_HAS_AVI */