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Remove bitrot related to voip testing

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This commit is contained in:
Jeremy Lakeman 2014-04-07 15:48:10 +09:30
parent 025726dcab
commit 1228df2b70
5 changed files with 2 additions and 350 deletions
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31
INSTALL.md
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@ -170,37 +170,6 @@ Configure
The [doc/Servald-Configuration](./doc/Servald-Configuration.md) document The [doc/Servald-Configuration](./doc/Servald-Configuration.md) document
describes the configuration of Serval DNA in detail. describes the configuration of Serval DNA in detail.
Voice call test
---------------
If the following packages are present then `./configure` will set the
`HAVE_VOIPTEST` macro and build **servald** with its `phone` command available
for performing voice call testing:
* [Port audio](http://www.portaudio.com)
* [Secret Rabbit Code](http://www.mega-nerd.com/SRC/) (a.k.a. Sample Rate
Convert) by Erik de Castro Lopo
* [SpanDSP](http://www.soft-switch.org/) by Steve Underwood
* [Codec2](http://www.rowetel.com/blog/?page_id=452) by Dave Rowe of Rowetel
The Codec2 source code can be fetched using [Subversion][]:
$ cd $HOME/src
$ svn checkout https://freetel.svn.sourceforge.net/svnroot/freetel/codec2 codec2
$
There are command-line options to control the `HAVE_VOIPTEST` macro:
* To force `HAVE_VOIPTEST` to be set, and fail if the necessary packages are
not present, use:
$ ./configure --enable-voiptest
* To force `HAVE_VOIPTEST` to be un-set (and not check for the presence of the
above packages), use:
$ ./configure --disable-voiptest
About the examples About the examples
------------------ ------------------

15
Makefile.in
View File

@ -16,17 +16,6 @@ SERVALD_SRCS = \
$(SERVAL_CLIENT_SOURCES) \ $(SERVAL_CLIENT_SOURCES) \
$(SERVAL_DAEMON_SOURCES) $(SERVAL_DAEMON_SOURCES)
HAVE_ALSA= @HAVE_ALSA@
ifeq (HAVE_ALSA,1)
SERVALD_SRCS+= audio_alsa.c
endif
HAVE_VOIPTEST= @HAVE_VOIPTEST@
ifeq ($(HAVE_VOIPTEST), 1)
SERVALD_SRCS+= pa_phone.c
VOIPTEST_CFLAGS=-DHAVE_VOIPTEST=1
endif
MONITOR_CLIENT_SRCS = \ MONITOR_CLIENT_SRCS = \
$(SERVAL_CLIENT_SOURCES) \ $(SERVAL_CLIENT_SOURCES) \
monitor-client.c monitor-client.c
@ -41,9 +30,9 @@ SERVAL_DAEMON_OBJS= $(SERVAL_DAEMON_SOURCES:.c=.o)
MONITOR_CLIENT_OBJS= $(MONITOR_CLIENT_SRCS:.c=.o) MONITOR_CLIENT_OBJS= $(MONITOR_CLIENT_SRCS:.c=.o)
MDP_CLIENT_OBJS= $(MDP_CLIENT_SRCS:.c=.o) MDP_CLIENT_OBJS= $(MDP_CLIENT_SRCS:.c=.o)
LDFLAGS=@LDFLAGS@ @LIBS@ @PORTAUDIO_LIBS@ @SRC_LIBS@ @SPANDSP_LIBS@ @CODEC2_LIBS@ @PTHREAD_LIBS@ LDFLAGS=@LDFLAGS@ @LIBS@ @PTHREAD_LIBS@
CFLAGS= -Isqlite-amalgamation-3070900 @CPPFLAGS@ @CFLAGS@ @PORTAUDIO_CFLAGS@ @SRC_CFLAGS@ @SPANDSP_CFLAGS@ @PTHREAD_CFLAGS@ $(VOIPTEST_CFLAGS) -Inacl/include CFLAGS= -Isqlite-amalgamation-3070900 @CPPFLAGS@ @CFLAGS@ @PTHREAD_CFLAGS@ -Inacl/include
CFLAGS+=-DSYSCONFDIR="\"$(sysconfdir)\"" -DLOCALSTATEDIR="\"$(localstatedir)\"" CFLAGS+=-DSYSCONFDIR="\"$(sysconfdir)\"" -DLOCALSTATEDIR="\"$(localstatedir)\""
CFLAGS+=-fPIC CFLAGS+=-fPIC
CFLAGS+=-Wall -Wno-unused-value CFLAGS+=-Wall -Wno-unused-value

4
commandline.c
View File

@ -3135,9 +3135,5 @@ struct cli_schema command_line_options[]={
"Listen for incoming connections"}, "Listen for incoming connections"},
{app_msp_connection,{"msp", "connect", "[--once]", "[--forward=<local_port>]", "<sid>", "<port>", NULL}, 0, {app_msp_connection,{"msp", "connect", "[--once]", "[--forward=<local_port>]", "<sid>", "<port>", NULL}, 0,
"Connect to a remote party"}, "Connect to a remote party"},
#ifdef HAVE_VOIPTEST
{app_pa_phone,{"phone",NULL}, 0,
"Run phone test application"},
#endif
{NULL,{NULL},0,NULL} {NULL,{NULL},0,NULL}
}; };

25
configure.in
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@ -38,26 +38,6 @@ dnl Specify default Rhizome store directory
AC_ARG_VAR([RHIZOME_STORE_PATH], [default Rhizome store directory]) AC_ARG_VAR([RHIZOME_STORE_PATH], [default Rhizome store directory])
AS_IF([test "x$RHIZOME_STORE_PATH" != x], [AC_DEFINE_UNQUOTED([RHIZOME_STORE_PATH], ["$RHIZOME_STORE_PATH"], [default Rhizome store directory])]) AS_IF([test "x$RHIZOME_STORE_PATH" != x], [AC_DEFINE_UNQUOTED([RHIZOME_STORE_PATH], ["$RHIZOME_STORE_PATH"], [default Rhizome store directory])])
dnl VoIP test app
AC_ARG_ENABLE(voiptest,
AS_HELP_STRING([--enable-voiptest], [Require VoIP test program (default: only build if dependencies are present)])
AS_HELP_STRING([--disable-voiptest], [Disable VoIP test program (don't test for dependencies)])dnl'
)
have_voip=0
AS_IF([test "x$enable_voiptest" != "xno"], [
have_voip=1
PKG_CHECK_MODULES([PORTAUDIO], [portaudio-2.0],,[have_voip=0])
PKG_CHECK_MODULES([SRC], [samplerate],,[have_voip=0])
PKG_CHECK_MODULES([SPANDSP], [spandsp],,[have_voip=0])
AC_CHECK_HEADER([codec2.h],, [have_voip=0])
AC_CHECK_LIB([codec2], [codec2_create], [AC_SUBST([CODEC2_LIBS], -lcodec2)], [have_voip=0])
])
AS_IF([test "x$enable_voiptest" = "xyes" -a "x$have_voip" != "x1" ], [
AC_MSG_ERROR([Missing VoIP dependencies])
])
AC_SUBST([HAVE_VOIPTEST], $have_voip)
dnl Check for programs. dnl Check for programs.
AC_PROG_CC AC_PROG_CC
@ -141,11 +121,6 @@ AC_CHECK_HEADERS(
#endif #endif
]) ])
dnl Check for ALSA
AC_CHECK_HEADER([alsa/asoundlib.h], [have_alsa=1], [have_alsa=0])
AS_IF([test x"$have_alsa" = "x1"], [AC_DEFINE([HAVE_ALSA_ASOUNDLIB_H])])
AS_IF([test x"$have_alsa" = "x1"], [AC_SUBST(HAVE_ALSA,1)], [AC_SUBST(HAVE_ALSA,0)])
dnl Lazy way of checking for Linux dnl Lazy way of checking for Linux
AS_IF([test "x$ac_cv_header_linux_if_h" = xyes], [AC_DEFINE([USE_ABSTRACT_NAMESPACE])]) AS_IF([test "x$ac_cv_header_linux_if_h" = xyes], [AC_DEFINE([USE_ABSTRACT_NAMESPACE])])

277
pa_phone.c
View File

@ -1,277 +0,0 @@
/*
Serval DNA Portaudio phone interface
Copyright (C) 2012 Serval Project Inc.
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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include <codec2.h>
#include <spandsp.h>
#include "fifo.h"
#include <portaudio.h>
#include <pthread.h>
#include <samplerate.h>
#include "serval.h"
/* Defines */
#define IN_FRAMES 128
#define NUM_BUFS (8)
#define ECHO_LEN (128)
#define ADAPT_MODE (ECHO_CAN_USE_ADAPTION | ECHO_CAN_USE_NLP | ECHO_CAN_USE_CNG)
#define CODEC2_BYTES_PER_FRAME ((CODEC2_BITS_PER_FRAME + 7) / 8)
/* Prototypes */
typedef struct {
PaStream *stream;
SRC_STATE *src;
pthread_mutex_t mtx; /* Mutex for frobbing queues */
/* Incoming samples after decompression
* Written with result of recvfrom + codec2_decode
* Read by sample rate converter
*/
struct fifo *incoming;
int incoverflow;
/* Samples after rate conversion
* Written by sample rate converter
* Read by PA callback
*/
struct fifo *incrate;
int underrun;
/* Outgoing samples
* Written by PA callback
* Read by codec2_encode + sendto
*/
struct fifo *outgoing;
int overrun;
echo_can_state_t *echocan; /* Echo canceller state */
void *codec2; /* Codec2 state */
} PaCtx;
static void freectx(PaCtx *ctx);
static int patestCallback(const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags,
void *userData);
static PaCtx *pa_phone_setup(void);
/* Declarations */
int app_pa_phone(const struct cli_parsed *parsed, void *context)
{
PaCtx *ctx;
if ((ctx = pa_phone_setup()) == NULL)
return -1;
freectx(ctx);
return 0;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int
patestCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData) {
PaCtx *ctx;
int16_t *in, *out;
int avail, amt;
ctx = (PaCtx *)userData;
out = (int16_t *)outputBuffer;
in = (int16_t *)inputBuffer;
pthread_mutex_lock(&ctx->mtx);
amt = framesPerBuffer * sizeof(out[0]);
/* Copy out samples to be played */
if ((avail = fifo_get(ctx->incrate, (uint8_t *)out, amt)) < amt) {
/* Zero out samples there are no data for */
bzero(out + (avail / sizeof(out[0])), amt - avail);
ctx->underrun += (amt - avail) / sizeof(out[0]);
}
/* Copy in samples to be recorded */
if ((avail = fifo_put(ctx->outgoing, (uint8_t *)in, amt)) < amt) {
/* Zero out samples there are no data for */
bzero(in + (avail / sizeof(out[0])), amt - avail);
ctx->overrun += (amt - avail) / sizeof(out[0]);
}
#if 1
/* Run the echo canceller */
for (int ofs = 0; ofs < framesPerBuffer; ofs++)
out[ofs] = echo_can_update(ctx->echocan, in[ofs], out[ofs]);
#endif
pthread_mutex_unlock(&ctx->mtx);
return paContinue;
}
static PaCtx *
pa_phone_setup(void) {
PaCtx *ctx;
int err, i, srcerr;
PaError err2;
err = paNoError;
err2 = 0;
if ((ctx = calloc(1, sizeof(PaCtx))) == NULL) {
WHY("Unable to allocate PA context");
err2 = 1;
goto error;
}
/* Init mutex */
if (pthread_mutex_init(&ctx->mtx, NULL) != 0) {
WHYF("Unable to init mutex: %s\n", strerror(errno));
err2 = 1;
goto error;
}
/* Allocate FIFOs */
i = IN_FRAMES * 10 * sizeof(int16_t);
printf("Allocating %d byte FIFOs\n", i);
if ((ctx->incoming = fifo_alloc(i)) == NULL) {
WHY("Unable to allocate incoming FIFO\n");
err2 = 1;
goto error;
}
if ((ctx->incrate = fifo_alloc(i)) == NULL) {
WHY("Unable to allocate incoming SRC FIFO\n");
err2 = 1;
goto error;
}
if ((ctx->outgoing = fifo_alloc(i)) == NULL) {
WHY("Unable to allocate outgoing FIFO\n");
err2 = 1;
goto error;
}
/* Init sample rate converter */
if ((ctx->src = src_new(SRC_SINC_BEST_QUALITY, 1, &srcerr)) == NULL) {
WHYF("Unable to init sample rate converter: %d\n", srcerr);
err2 = 1;
goto error;
}
/* Init echo canceller */
if ((ctx->echocan = echo_can_init(ECHO_LEN, ADAPT_MODE)) == NULL) {
WHY("Unable to init echo canceller\n");
err2 = 1;
goto error;
}
/* Init codec2 */
if ((ctx->codec2 = codec2_create()) == NULL) {
WHY("Unable to init codec2\n");
err2 = 1;
goto error;
}
/* Initialize Port Audio library */
if ((err = Pa_Initialize()) != paNoError)
goto error;
/* Open an audio I/O stream. */
if ((err = Pa_OpenDefaultStream(&ctx->stream,
1, /* input channels */
1, /* output channels */
paInt16,
SAMPLE_RATE,
IN_FRAMES, /* frames per buffer */
patestCallback,
&ctx)) != paNoError)
goto error;
/* Start stream */
if ((err = Pa_StartStream(ctx->stream)) != paNoError)
goto error;
/* Close down stream, PA, etc */
/* XXX: hangs in pthread_join on Ubuntu 10.04 */
#ifndef linux
if ((err = Pa_StopStream(ctx->stream)) != paNoError)
goto error;
#endif
/* Do stuff */
if ((err = Pa_CloseStream(ctx->stream)) != paNoError)
goto error;
error:
Pa_Terminate();
/* Free things */
freectx(ctx);
if (err != paNoError)
WHYF("Port audio error: %s\n", Pa_GetErrorText(err));
return NULL;
}
static void
freectx(PaCtx *ctx) {
/* Destroy mutex */
pthread_mutex_destroy(&ctx->mtx);
/* Free SRC resources */
if (ctx->src != NULL)
src_delete(ctx->src);
/* Free echo caneller */
if (ctx->echocan != NULL)
echo_can_free(ctx->echocan);
/* Free codec2 */
if (ctx->codec2 != NULL)
codec2_destroy(ctx->codec2);
/* Free FIFOs */
if (ctx->incoming != NULL)
fifo_free(ctx->incoming);
if (ctx->incrate != NULL)
fifo_free(ctx->incrate);
if (ctx->outgoing != NULL)
fifo_free(ctx->outgoing);
}
/*
* Local variables:
* c-basic-offset: 2
* End:
*/