[asterisk-dev] Questions about app_jack.c [solved]
Fabien COMTE
fabien.comte at ercogener.com
Tue Oct 6 04:28:34 CDT 2009
Hi,
http://issues.digium.com is not responding.
Bug was that queue_voice_frame was not called in right place.
Optimization are about removing / moving some useless memset.
Fabien
This is the patch :
*** app_jack_original.c 2009-10-06 11:19:38.000000000 +0200
--- app_jack.c 2009-10-06 10:20:15.000000000 +0200
***************
*** 247,260 ****
}
for (i = 0; i < total_out_buf_used; i++)
! s_buf[i] = f_buf[i] * (SHRT_MAX / 1.0);
write_len = total_out_buf_used * sizeof(int16_t);
} else {
/* No resampling needed */
for (i = 0; i < nframes; i++)
! s_buf[i] = in_buf[i] * (SHRT_MAX / 1.0);
}
res = jack_ringbuffer_write(jack_data->input_rb, (const char *)
s_buf, write_len);
--- 247,260 ----
}
for (i = 0; i < total_out_buf_used; i++)
! s_buf[i] = ((float) f_buf[i]) * ((float) SHRT_MAX);
write_len = total_out_buf_used * sizeof(int16_t);
} else {
/* No resampling needed */
for (i = 0; i < nframes; i++)
! s_buf[i] = ((float) in_buf[i]) * ((float) SHRT_MAX);
}
res = jack_ringbuffer_write(jack_data->input_rb, (const char *)
s_buf, write_len);
***************
*** 370,376 ****
ast_channel_unlock(chan);
}
! if (!(jack_data->output_rb =
jack_ringbuffer_create(RINGBUFFER_SIZE)))
return -1;
if (!(jack_data->input_rb =
jack_ringbuffer_create(RINGBUFFER_SIZE)))
--- 370,376 ----
ast_channel_unlock(chan);
}
! if (!(jack_data->output_rb = jack_ringbuffer_create(RINGBUFFER_SIZE
* 2)))
return -1;
if (!(jack_data->input_rb =
jack_ringbuffer_create(RINGBUFFER_SIZE)))
***************
*** 492,511 ****
int16_t *s_buf = f->data.ptr;
size_t res;
! memset(f_buf, 0, sizeof(f_buf));
if (!jack_data->output_resample_factor)
alloc_resampler(jack_data, 0);
! if (jack_data->output_resampler) {
float in_buf[f->samples];
int total_in_buf_used = 0;
int total_out_buf_used = 0;
! memset(in_buf, 0, sizeof(in_buf));
for (i = 0; i < f->samples; i++)
! in_buf[i] = s_buf[i] * (1.0 / SHRT_MAX);
while (total_in_buf_used < ARRAY_LEN(in_buf)) {
int in_buf_used;
--- 492,513 ----
int16_t *s_buf = f->data.ptr;
size_t res;
! // memset(f_buf, 0, sizeof(f_buf)); // not needed if no resample
if (!jack_data->output_resample_factor)
alloc_resampler(jack_data, 0);
! if (jack_data->output_resampler) {
float in_buf[f->samples];
int total_in_buf_used = 0;
int total_out_buf_used = 0;
! memset(f_buf, 0, sizeof(f_buf));
!
! // memset(in_buf, 0, sizeof(in_buf)); // not needed (see the
next for)
for (i = 0; i < f->samples; i++)
! in_buf[i] = ((float) s_buf[i]) / ((float) SHRT_MAX);
while (total_in_buf_used < ARRAY_LEN(in_buf)) {
int in_buf_used;
***************
*** 536,542 ****
/* No resampling needed */
for (i = 0; i < f->samples; i++)
! f_buf[i] = s_buf[i] * (1.0 / SHRT_MAX);
f_buf_used = f->samples;
}
--- 538,544 ----
/* No resampling needed */
for (i = 0; i < f->samples; i++)
! f_buf[i] = ((float) s_buf[i]) / ((float) SHRT_MAX);
f_buf_used = f->samples;
}
***************
*** 797,811 ****
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
return 0;
- if (direction != AST_AUDIOHOOK_DIRECTION_READ)
- return 0;
-
if (frame->frametype != AST_FRAME_VOICE)
return 0;
! if (frame->subclass != AST_FORMAT_SLINEAR) {
ast_log(LOG_WARNING, "Expected frame in SLINEAR for the
audiohook, but got format %d\n",
! frame->subclass);
return 0;
}
--- 799,811 ----
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
return 0;
if (frame->frametype != AST_FRAME_VOICE)
return 0;
! if (frame->subclass != AST_FORMAT_SLINEAR)
! {
ast_log(LOG_WARNING, "Expected frame in SLINEAR for the
audiohook, but got format %d\n",
! frame->subclass);
return 0;
}
***************
*** 819,827 ****
jack_data = datastore->data;
! queue_voice_frame(jack_data, frame);
!
! handle_jack_audio(chan, jack_data, frame);
ast_channel_unlock(chan);
--- 819,832 ----
jack_data = datastore->data;
! if (direction != AST_AUDIOHOOK_DIRECTION_READ)
! {
! queue_voice_frame(jack_data, frame);
! }
! else
! {
! handle_jack_audio(chan, jack_data, frame);
! }
ast_channel_unlock(chan);
-----Message d'origine-----
De : asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] De la part de Kai Hoerner
Envoyé : mardi 6 octobre 2009 11:11
À : Asterisk Developers Mailing List
Objet : Re: [asterisk-dev] Questions about app_jack.c [solved]
Hi Fabien,
please make a diff between revision 140568 and your new file to reflect
the changes and upload this to the http://issues.digium.com tracker so
the community has a real benefit from your efforts.
thanks in advance,
kaii
Fabien COMTE schrieb:
> Hello,
> I corrected a bug and did some little optimizations in app_jack.c.
>
> It works great now.
>
> I propose this new file based on revision 140568.
>
> Fabien
>
> /*
> * Asterisk -- An open source telephony toolkit.
> *
> * Copyright (C) 2007 - 2008, Russell Bryant
> *
> * Russell Bryant <russell at digium.com>
> *
> * See http://www.asterisk.org for more information about
> * the Asterisk project. Please do not directly contact
> * any of the maintainers of this project for assistance;
> * the project provides a web site, mailing lists and IRC
> * channels for your use.
> *
> * This program is free software, distributed under the terms of
> * the GNU General Public License Version 2. See the LICENSE file
> * at the top of the source tree.
> */
>
> /*!
> * \file
> * \brief Jack Application
> *
> * \author Russell Bryant <russell at digium.com>
> *
> * This is an application to connect an Asterisk channel to an input
> * and output jack port so that the audio can be processed through
> * another application, or to play audio from another application.
> *
> * \arg http://www.jackaudio.org/
> *
> * \note To install libresample, check it out of the following repository:
> * <code>$ svn co
> http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
> *
> * \ingroup applications
> */
>
> /*** MODULEINFO
> <depend>jack</depend>
> <depend>resample</depend>
> ***/
>
> #include "asterisk.h"
>
> ASTERISK_FILE_VERSION(__FILE__, "$Revision: 140568 $")
>
> #include <limits.h>
>
> #include <jack/jack.h>
> #include <jack/ringbuffer.h>
>
> #include <libresample.h>
>
> #include "asterisk/module.h"
> #include "asterisk/channel.h"
> #include "asterisk/strings.h"
> #include "asterisk/lock.h"
> #include "asterisk/app.h"
> #include "asterisk/pbx.h"
> #include "asterisk/audiohook.h"
>
> #define RESAMPLE_QUALITY 1
>
> #define RINGBUFFER_SIZE 16384
>
> /*! \brief Common options between the Jack() app and JACK_HOOK() function
*/
> #define COMMON_OPTIONS \
> " s(<name>) - Connect to the specified jack server name.\n" \
> " i(<name>) - Connect the output port that gets created to the
> specified\n" \
> " jack input port.\n" \
> " o(<name>) - Connect the input port that gets created to the
> specified\n" \
> " jack output port.\n" \
> " n - Do not automatically start the JACK server if it is not
> already\n" \
> " running.\n" \
> " c(<name>) - By default, Asterisk will use the channel name for the
jack
> client\n" \
> " name. Use this option to specify a custom client
name.\n"
>
> static char *jack_app = "JACK";
> static char *jack_synopsis =
> "JACK (Jack Audio Connection Kit) Application";
> static char *jack_desc =
> "JACK([options])\n"
> " When this application is executed, two jack ports will be created; one
> input\n"
> "and one output. Other applications can be hooked up to these ports to
> access\n"
> "the audio coming from, or being sent to the channel.\n"
> " Valid options:\n"
> COMMON_OPTIONS
> "";
>
> struct jack_data {
> AST_DECLARE_STRING_FIELDS(
> AST_STRING_FIELD(server_name);
> AST_STRING_FIELD(client_name);
> AST_STRING_FIELD(connect_input_port);
> AST_STRING_FIELD(connect_output_port);
> );
> jack_client_t *client;
> jack_port_t *input_port;
> jack_port_t *output_port;
> jack_ringbuffer_t *input_rb;
> jack_ringbuffer_t *output_rb;
> void *output_resampler;
> double output_resample_factor;
> void *input_resampler;
> double input_resample_factor;
> unsigned int stop:1;
> unsigned int has_audiohook:1;
> unsigned int no_start_server:1;
> /*! Only used with JACK_HOOK */
> struct ast_audiohook audiohook;
> };
>
> static const struct {
> jack_status_t status;
> const char *str;
> } jack_status_table[] = {
> { JackFailure, "Failure" },
> { JackInvalidOption, "Invalid Option" },
> { JackNameNotUnique, "Name Not Unique" },
> { JackServerStarted, "Server Started" },
> { JackServerFailed, "Server Failed" },
> { JackServerError, "Server Error" },
> { JackNoSuchClient, "No Such Client" },
> { JackLoadFailure, "Load Failure" },
> { JackInitFailure, "Init Failure" },
> { JackShmFailure, "Shared Memory Access Failure" },
> { JackVersionError, "Version Mismatch" },
> };
>
> static const char *jack_status_to_str(jack_status_t status)
> {
> int i;
>
> for (i = 0; i < ARRAY_LEN(jack_status_table); i++) {
> if (jack_status_table[i].status == status)
> return jack_status_table[i].str;
> }
>
> return "Unknown Error";
> }
>
> static void log_jack_status(const char *prefix, jack_status_t status)
> {
> struct ast_str *str = ast_str_alloca(512);
> int i, first = 0;
>
> for (i = 0; i < (sizeof(status) * 8); i++) {
> if (!(status & (1 << i)))
> continue;
>
> if (!first) {
> ast_str_set(&str, 0, "%s", jack_status_to_str((1 <<
> i)));
> first = 1;
> } else
> ast_str_append(&str, 0, ", %s",
> jack_status_to_str((1 << i)));
> }
>
> ast_log(LOG_NOTICE, "%s: %s\n", prefix, str->str);
> }
>
> static int alloc_resampler(struct jack_data *jack_data, int input)
> {
> double from_srate, to_srate, jack_srate;
> void **resampler;
> double *resample_factor;
>
> if (input && jack_data->input_resampler)
> return 0;
>
> if (!input && jack_data->output_resampler)
> return 0;
>
> jack_srate = jack_get_sample_rate(jack_data->client);
>
> /* XXX Hard coded 8 kHz */
>
> to_srate = input ? 8000.0 : jack_srate;
> from_srate = input ? jack_srate : 8000.0;
>
> resample_factor = input ? &jack_data->input_resample_factor :
> &jack_data->output_resample_factor;
>
> if (from_srate == to_srate) {
> /* Awesome! The jack sample rate is the same as ours.
> * Resampling isn't needed. */
> *resample_factor = 1.0;
> return 0;
> }
>
> *resample_factor = to_srate / from_srate;
>
> resampler = input ? &jack_data->input_resampler :
> &jack_data->output_resampler;
>
> if (!(*resampler = resample_open(RESAMPLE_QUALITY,
> *resample_factor, *resample_factor))) {
> ast_log(LOG_ERROR, "Failed to open %s resampler\n",
> input ? "input" : "output");
> return -1;
> }
>
> return 0;
> }
>
> /*!
> * \brief Handle jack input port
> *
> * Read nframes number of samples from the input buffer, resample it
> * if necessary, and write it into the appropriate ringbuffer.
> */
> static void handle_input(void *buf, jack_nframes_t nframes,
> struct jack_data *jack_data)
> {
> short s_buf[nframes];
> float *in_buf = buf;
> size_t res;
> int i;
> size_t write_len = sizeof(s_buf);
>
> if (jack_data->input_resampler) {
> int total_in_buf_used = 0;
> int total_out_buf_used = 0;
> float f_buf[nframes + 1];
>
> memset(f_buf, 0, sizeof(f_buf));
>
> while (total_in_buf_used < nframes) {
> int in_buf_used;
> int out_buf_used;
>
> out_buf_used =
> resample_process(jack_data->input_resampler,
> jack_data->input_resample_factor,
> &in_buf[total_in_buf_used], nframes -
> total_in_buf_used,
> 0, &in_buf_used,
> &f_buf[total_out_buf_used], ARRAY_LEN(f_buf)
> - total_out_buf_used);
>
> if (out_buf_used < 0)
> break;
>
> total_out_buf_used += out_buf_used;
> total_in_buf_used += in_buf_used;
>
> if (total_out_buf_used == ARRAY_LEN(f_buf)) {
> ast_log(LOG_ERROR, "Output buffer filled ...
> need to increase its size, "
> "nframes '%d', total_out_buf_used
> '%d'\n", nframes, total_out_buf_used);
> break;
> }
> }
>
> for (i = 0; i < total_out_buf_used; i++)
> s_buf[i] = ((float) f_buf[i]) * ((float) SHRT_MAX);
>
> write_len = total_out_buf_used * sizeof(int16_t);
> } else {
> /* No resampling needed */
>
> for (i = 0; i < nframes; i++)
> s_buf[i] = ((float) in_buf[i]) * ((float) SHRT_MAX);
> }
>
> res = jack_ringbuffer_write(jack_data->input_rb, (const char *)
> s_buf, write_len);
> if (res != write_len) {
> ast_debug(2, "Tried to write %d bytes to the ringbuffer, but
> only wrote %d\n",
> (int) sizeof(s_buf), (int) res);
> }
> }
>
> /*!
> * \brief Handle jack output port
> *
> * Read nframes number of samples from the ringbuffer and write it out to
> the
> * output port buffer.
> */
> static void handle_output(void *buf, jack_nframes_t nframes,
> struct jack_data *jack_data)
> {
> size_t res, len;
>
> len = nframes * sizeof(float);
>
> res = jack_ringbuffer_read(jack_data->output_rb, buf, len);
>
> if (len != res) {
> ast_debug(2, "Wanted %d bytes to send to the output port, "
> "but only got %d\n", (int) len, (int) res);
> }
> }
>
> static int jack_process(jack_nframes_t nframes, void *arg)
> {
> struct jack_data *jack_data = arg;
> void *input_port_buf, *output_port_buf;
>
> if (!jack_data->input_resample_factor)
> alloc_resampler(jack_data, 1);
>
> input_port_buf = jack_port_get_buffer(jack_data->input_port,
> nframes);
> handle_input(input_port_buf, nframes, jack_data);
>
> output_port_buf = jack_port_get_buffer(jack_data->output_port,
> nframes);
> handle_output(output_port_buf, nframes, jack_data);
>
> return 0;
> }
>
> static void jack_shutdown(void *arg)
> {
> struct jack_data *jack_data = arg;
>
> jack_data->stop = 1;
> }
>
> static struct jack_data *destroy_jack_data(struct jack_data *jack_data)
> {
> if (jack_data->input_port) {
> jack_port_unregister(jack_data->client,
> jack_data->input_port);
> jack_data->input_port = NULL;
> }
>
> if (jack_data->output_port) {
> jack_port_unregister(jack_data->client,
> jack_data->output_port);
> jack_data->output_port = NULL;
> }
>
> if (jack_data->client) {
> jack_client_close(jack_data->client);
> jack_data->client = NULL;
> }
>
> if (jack_data->input_rb) {
> jack_ringbuffer_free(jack_data->input_rb);
> jack_data->input_rb = NULL;
> }
>
> if (jack_data->output_rb) {
> jack_ringbuffer_free(jack_data->output_rb);
> jack_data->output_rb = NULL;
> }
>
> if (jack_data->output_resampler) {
> resample_close(jack_data->output_resampler);
> jack_data->output_resampler = NULL;
> }
>
> if (jack_data->input_resampler) {
> resample_close(jack_data->input_resampler);
> jack_data->input_resampler = NULL;
> }
>
> if (jack_data->has_audiohook)
> ast_audiohook_destroy(&jack_data->audiohook);
>
> ast_string_field_free_memory(jack_data);
>
> ast_free(jack_data);
>
> return NULL;
> }
>
> static int init_jack_data(struct ast_channel *chan, struct jack_data
> *jack_data)
> {
> const char *client_name;
> jack_status_t status = 0;
> jack_options_t jack_options = JackNullOption;
>
> if (!ast_strlen_zero(jack_data->client_name)) {
> client_name = jack_data->client_name;
> } else {
> ast_channel_lock(chan);
> client_name = ast_strdupa(chan->name);
> ast_channel_unlock(chan);
> }
>
> if (!(jack_data->output_rb = jack_ringbuffer_create(RINGBUFFER_SIZE
> * 2)))
> return -1;
>
> if (!(jack_data->input_rb =
> jack_ringbuffer_create(RINGBUFFER_SIZE)))
> return -1;
>
> if (jack_data->no_start_server)
> jack_options |= JackNoStartServer;
>
> if (!ast_strlen_zero(jack_data->server_name)) {
> jack_options |= JackServerName;
> jack_data->client = jack_client_open(client_name,
> jack_options, &status,
> jack_data->server_name);
> } else {
> jack_data->client = jack_client_open(client_name,
> jack_options, &status);
> }
>
> if (status)
> log_jack_status("Client Open Status", status);
>
> if (!jack_data->client)
> return -1;
>
> jack_data->input_port = jack_port_register(jack_data->client,
> "input",
> JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput |
> JackPortIsTerminal, 0);
> if (!jack_data->input_port) {
> ast_log(LOG_ERROR, "Failed to create input port for jack
> port\n");
> return -1;
> }
>
> jack_data->output_port = jack_port_register(jack_data->client,
> "output",
> JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput |
> JackPortIsTerminal, 0);
> if (!jack_data->output_port) {
> ast_log(LOG_ERROR, "Failed to create output port for jack
> port\n");
> return -1;
> }
>
> if (jack_set_process_callback(jack_data->client, jack_process,
> jack_data)) {
> ast_log(LOG_ERROR, "Failed to register process callback with
> jack client\n");
> return -1;
> }
>
> jack_on_shutdown(jack_data->client, jack_shutdown, jack_data);
>
> if (jack_activate(jack_data->client)) {
> ast_log(LOG_ERROR, "Unable to activate jack client\n");
> return -1;
> }
>
> while (!ast_strlen_zero(jack_data->connect_input_port)) {
> const char **ports;
> int i;
>
> ports = jack_get_ports(jack_data->client,
> jack_data->connect_input_port,
> NULL, JackPortIsInput);
>
> if (!ports) {
> ast_log(LOG_ERROR, "No input port matching '%s' was
> found\n",
> jack_data->connect_input_port);
> break;
> }
>
> for (i = 0; ports[i]; i++) {
> ast_debug(1, "Found port '%s' that matched specified
> input port '%s'\n",
> ports[i], jack_data->connect_input_port);
> }
>
> if (jack_connect(jack_data->client,
> jack_port_name(jack_data->output_port), ports[0])) {
> ast_log(LOG_ERROR, "Failed to connect '%s' to
> '%s'\n", ports[0],
> jack_port_name(jack_data->output_port));
> } else {
> ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
> jack_port_name(jack_data->output_port));
> }
>
> free((void *) ports);
>
> break;
> }
>
> while (!ast_strlen_zero(jack_data->connect_output_port)) {
> const char **ports;
> int i;
>
> ports = jack_get_ports(jack_data->client,
> jack_data->connect_output_port,
> NULL, JackPortIsOutput);
>
> if (!ports) {
> ast_log(LOG_ERROR, "No output port matching '%s' was
> found\n",
> jack_data->connect_output_port);
> break;
> }
>
> for (i = 0; ports[i]; i++) {
> ast_debug(1, "Found port '%s' that matched specified
> output port '%s'\n",
> ports[i], jack_data->connect_output_port);
> }
>
> if (jack_connect(jack_data->client, ports[0],
> jack_port_name(jack_data->input_port))) {
> ast_log(LOG_ERROR, "Failed to connect '%s' to
> '%s'\n", ports[0],
> jack_port_name(jack_data->input_port));
> } else {
> ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
> jack_port_name(jack_data->input_port));
> }
>
> free((void *) ports);
>
> break;
> }
>
> return 0;
> }
>
> static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame
> *f)
> {
> float f_buf[f->samples * 8];
> size_t f_buf_used = 0;
> int i;
> int16_t *s_buf = f->data.ptr;
> size_t res;
>
> // memset(f_buf, 0, sizeof(f_buf)); // not needed if no resample
>
> if (!jack_data->output_resample_factor)
> alloc_resampler(jack_data, 0);
>
> if (jack_data->output_resampler) {
> float in_buf[f->samples];
> int total_in_buf_used = 0;
> int total_out_buf_used = 0;
>
> memset(f_buf, 0, sizeof(f_buf));
>
> // memset(in_buf, 0, sizeof(in_buf)); // not needed (see the
> next for)
>
> for (i = 0; i < f->samples; i++)
> in_buf[i] = ((float) s_buf[i]) / ((float) SHRT_MAX);
>
> while (total_in_buf_used < ARRAY_LEN(in_buf)) {
> int in_buf_used;
> int out_buf_used;
>
> out_buf_used =
> resample_process(jack_data->output_resampler,
> jack_data->output_resample_factor,
> &in_buf[total_in_buf_used],
> ARRAY_LEN(in_buf) - total_in_buf_used,
> 0, &in_buf_used,
> &f_buf[total_out_buf_used], ARRAY_LEN(f_buf)
> - total_out_buf_used);
>
> if (out_buf_used < 0)
> break;
>
> total_out_buf_used += out_buf_used;
> total_in_buf_used += in_buf_used;
>
> if (total_out_buf_used == ARRAY_LEN(f_buf)) {
> ast_log(LOG_ERROR, "Output buffer filled ...
> need to increase its size\n");
> break;
> }
> }
>
> f_buf_used = total_out_buf_used;
> if (f_buf_used > ARRAY_LEN(f_buf))
> f_buf_used = ARRAY_LEN(f_buf);
> } else {
> /* No resampling needed */
>
> for (i = 0; i < f->samples; i++)
> f_buf[i] = ((float) s_buf[i]) / ((float) SHRT_MAX);
>
> f_buf_used = f->samples;
> }
>
> res = jack_ringbuffer_write(jack_data->output_rb, (const char *)
> f_buf, f_buf_used * sizeof(float));
> if (res != (f_buf_used * sizeof(float))) {
> ast_debug(2, "Tried to write %d bytes to the ringbuffer, but
> only wrote %d\n",
> (int) (f_buf_used * sizeof(float)), (int) res);
> }
>
> return 0;
> }
>
> /*!
> * \brief handle jack audio
> *
> * \param[in] chan The Asterisk channel to write the frames to if no
output
> frame
> * is provided.
> * \param[in] jack_data This is the jack_data struct that contains the
> input
> * ringbuffer that audio will be read from.
> * \param[out] out_frame If this argument is non-NULL, then assuming there
> is
> * enough data avilable in the ringbuffer, the audio in this
> frame
> * will get replaced with audio from the input buffer. If
there
> is
> * not enough data available to read at this time, then the
> frame
> * data gets zeroed out.
> *
> * Read data from the input ringbuffer, which is the properly resampled
> audio
> * that was read from the jack input port. Write it to the channel in 20
ms
> frames,
> * or fill up an output frame instead if one is provided.
> *
> * \return Nothing.
> */
> static void handle_jack_audio(struct ast_channel *chan, struct jack_data
> *jack_data,
> struct ast_frame *out_frame)
> {
> short buf[160];
> struct ast_frame f = {
> .frametype = AST_FRAME_VOICE,
> .subclass = AST_FORMAT_SLINEAR,
> .src = "JACK",
> .data.ptr = buf,
> .datalen = sizeof(buf),
> .samples = ARRAY_LEN(buf),
> };
>
> for (;;) {
> size_t res, read_len;
> char *read_buf;
>
> read_len = out_frame ? out_frame->datalen : sizeof(buf);
> read_buf = out_frame ? out_frame->data.ptr : buf;
>
> res = jack_ringbuffer_read_space(jack_data->input_rb);
>
> if (res < read_len) {
> /* Not enough data ready for another frame, move on
> ... */
> if (out_frame) {
> ast_debug(1, "Sending an empty frame for the
> JACK_HOOK\n");
> memset(out_frame->data.ptr, 0,
> out_frame->datalen);
> }
> break;
> }
>
> res = jack_ringbuffer_read(jack_data->input_rb, (char *)
> read_buf, read_len);
>
> if (res < read_len) {
> ast_log(LOG_ERROR, "Error reading from ringbuffer,
> even though it said there was enough data\n");
> break;
> }
>
> if (out_frame) {
> /* If an output frame was provided, then we just
> want to fill up the
> * buffer in that frame and return. */
> break;
> }
>
> ast_write(chan, &f);
> }
> }
>
> enum {
> OPT_SERVER_NAME = (1 << 0),
> OPT_INPUT_PORT = (1 << 1),
> OPT_OUTPUT_PORT = (1 << 2),
> OPT_NOSTART_SERVER = (1 << 3),
> OPT_CLIENT_NAME = (1 << 4),
> };
>
> enum {
> OPT_ARG_SERVER_NAME,
> OPT_ARG_INPUT_PORT,
> OPT_ARG_OUTPUT_PORT,
> OPT_ARG_CLIENT_NAME,
>
> /* Must be the last element */
> OPT_ARG_ARRAY_SIZE,
> };
>
> AST_APP_OPTIONS(jack_exec_options, BEGIN_OPTIONS
> AST_APP_OPTION_ARG('s', OPT_SERVER_NAME, OPT_ARG_SERVER_NAME),
> AST_APP_OPTION_ARG('i', OPT_INPUT_PORT, OPT_ARG_INPUT_PORT),
> AST_APP_OPTION_ARG('o', OPT_OUTPUT_PORT, OPT_ARG_OUTPUT_PORT),
> AST_APP_OPTION('n', OPT_NOSTART_SERVER),
> AST_APP_OPTION_ARG('c', OPT_CLIENT_NAME, OPT_ARG_CLIENT_NAME),
> END_OPTIONS );
>
> static struct jack_data *jack_data_alloc(void)
> {
> struct jack_data *jack_data;
>
> if (!(jack_data = ast_calloc(1, sizeof(*jack_data))))
> return NULL;
>
> if (ast_string_field_init(jack_data, 32)) {
> ast_free(jack_data);
> return NULL;
> }
>
> return jack_data;
> }
>
> /*!
> * \note This must be done before calling init_jack_data().
> */
> static int handle_options(struct jack_data *jack_data, const char
> *__options_str)
> {
> struct ast_flags options = { 0, };
> char *option_args[OPT_ARG_ARRAY_SIZE];
> char *options_str;
>
> options_str = ast_strdupa(__options_str);
>
> ast_app_parse_options(jack_exec_options, &options, option_args,
> options_str);
>
> if (ast_test_flag(&options, OPT_SERVER_NAME)) {
> if (!ast_strlen_zero(option_args[OPT_ARG_SERVER_NAME]))
> ast_string_field_set(jack_data, server_name,
> option_args[OPT_ARG_SERVER_NAME]);
> else {
> ast_log(LOG_ERROR, "A server name must be provided
> with the s() option\n");
> return -1;
> }
> }
>
> if (ast_test_flag(&options, OPT_CLIENT_NAME)) {
> if (!ast_strlen_zero(option_args[OPT_ARG_CLIENT_NAME]))
> ast_string_field_set(jack_data, client_name,
> option_args[OPT_ARG_CLIENT_NAME]);
> else {
> ast_log(LOG_ERROR, "A client name must be provided
> with the c() option\n");
> return -1;
> }
> }
>
> if (ast_test_flag(&options, OPT_INPUT_PORT)) {
> if (!ast_strlen_zero(option_args[OPT_ARG_INPUT_PORT]))
> ast_string_field_set(jack_data, connect_input_port,
> option_args[OPT_ARG_INPUT_PORT]);
> else {
> ast_log(LOG_ERROR, "A name must be provided with the
> i() option\n");
> return -1;
> }
> }
>
> if (ast_test_flag(&options, OPT_OUTPUT_PORT)) {
> if (!ast_strlen_zero(option_args[OPT_ARG_OUTPUT_PORT]))
> ast_string_field_set(jack_data, connect_output_port,
> option_args[OPT_ARG_OUTPUT_PORT]);
> else {
> ast_log(LOG_ERROR, "A name must be provided with the
> o() option\n");
> return -1;
> }
> }
>
> jack_data->no_start_server = ast_test_flag(&options,
> OPT_NOSTART_SERVER) ? 1 : 0;
>
> return 0;
> }
>
> static int jack_exec(struct ast_channel *chan, void *data)
> {
> struct jack_data *jack_data;
> AST_DECLARE_APP_ARGS(args,
> AST_APP_ARG(options);
> );
>
> if (!(jack_data = jack_data_alloc()))
> return -1;
>
> args.options = data;
>
> if (!ast_strlen_zero(args.options) && handle_options(jack_data,
> args.options)) {
> destroy_jack_data(jack_data);
> return -1;
> }
>
> if (init_jack_data(chan, jack_data)) {
> destroy_jack_data(jack_data);
> return -1;
> }
>
> if (ast_set_read_format(chan, AST_FORMAT_SLINEAR)) {
> destroy_jack_data(jack_data);
> return -1;
> }
>
> if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
> destroy_jack_data(jack_data);
> return -1;
> }
>
> while (!jack_data->stop) {
> struct ast_frame *f;
>
> ast_waitfor(chan, -1);
>
> f = ast_read(chan);
> if (!f) {
> jack_data->stop = 1;
> continue;
> }
>
> switch (f->frametype) {
> case AST_FRAME_CONTROL:
> if (f->subclass == AST_CONTROL_HANGUP)
> jack_data->stop = 1;
> break;
> case AST_FRAME_VOICE:
> queue_voice_frame(jack_data, f);
> default:
> break;
> }
>
> ast_frfree(f);
>
> handle_jack_audio(chan, jack_data, NULL);
> }
>
> jack_data = destroy_jack_data(jack_data);
>
> return 0;
> }
>
> static void jack_hook_ds_destroy(void *data)
> {
> struct jack_data *jack_data = data;
>
> destroy_jack_data(jack_data);
> }
>
> static const struct ast_datastore_info jack_hook_ds_info = {
> .type = "JACK_HOOK",
> .destroy = jack_hook_ds_destroy,
> };
>
> static int jack_hook_callback(struct ast_audiohook *audiohook, struct
> ast_channel *chan,
> struct ast_frame *frame, enum ast_audiohook_direction direction)
> {
> struct ast_datastore *datastore;
> struct jack_data *jack_data;
>
> if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
> return 0;
>
> if (frame->frametype != AST_FRAME_VOICE)
> return 0;
>
> if (frame->subclass != AST_FORMAT_SLINEAR)
> {
> ast_log(LOG_WARNING, "Expected frame in SLINEAR for the
> audiohook, but got format %d\n",
> frame->subclass);
> return 0;
> }
>
> ast_channel_lock(chan);
>
> if (!(datastore = ast_channel_datastore_find(chan,
> &jack_hook_ds_info, NULL))) {
> ast_log(LOG_ERROR, "JACK_HOOK datastore not found for
> '%s'\n", chan->name);
> ast_channel_unlock(chan);
> return -1;
> }
>
> jack_data = datastore->data;
>
> if (direction != AST_AUDIOHOOK_DIRECTION_READ)
> {
> queue_voice_frame(jack_data, frame);
> }
> else
> {
> handle_jack_audio(chan, jack_data, frame);
> }
>
> ast_channel_unlock(chan);
>
> return 0;
> }
>
> static int enable_jack_hook(struct ast_channel *chan, char *data)
> {
> struct ast_datastore *datastore;
> struct jack_data *jack_data = NULL;
> AST_DECLARE_APP_ARGS(args,
> AST_APP_ARG(mode);
> AST_APP_ARG(options);
> );
>
> AST_STANDARD_APP_ARGS(args, data);
>
> ast_channel_lock(chan);
>
> if ((datastore = ast_channel_datastore_find(chan,
> &jack_hook_ds_info, NULL))) {
> ast_log(LOG_ERROR, "JACK_HOOK already enabled for '%s'\n",
> chan->name);
> goto return_error;
> }
>
> if (ast_strlen_zero(args.mode) || strcasecmp(args.mode,
> "manipulate")) {
> ast_log(LOG_ERROR, "'%s' is not a supported mode. Only
> manipulate is supported.\n",
> S_OR(args.mode, "<none>"));
> goto return_error;
> }
>
> if (!(jack_data = jack_data_alloc()))
> goto return_error;
>
> if (!ast_strlen_zero(args.options) && handle_options(jack_data,
> args.options))
> goto return_error;
>
> if (init_jack_data(chan, jack_data))
> goto return_error;
>
> if (!(datastore = ast_datastore_alloc(&jack_hook_ds_info, NULL)))
> goto return_error;
>
> jack_data->has_audiohook = 1;
> ast_audiohook_init(&jack_data->audiohook,
> AST_AUDIOHOOK_TYPE_MANIPULATE, "JACK_HOOK");
> jack_data->audiohook.manipulate_callback = jack_hook_callback;
>
> datastore->data = jack_data;
>
> if (ast_audiohook_attach(chan, &jack_data->audiohook))
> goto return_error;
>
> if (ast_channel_datastore_add(chan, datastore))
> goto return_error;
>
> ast_channel_unlock(chan);
>
> return 0;
>
> return_error:
> ast_channel_unlock(chan);
>
> if (jack_data)
> destroy_jack_data(jack_data);
>
> return -1;
> }
>
> static int disable_jack_hook(struct ast_channel *chan)
> {
> struct ast_datastore *datastore;
> struct jack_data *jack_data;
>
> ast_channel_lock(chan);
>
> if (!(datastore = ast_channel_datastore_find(chan,
> &jack_hook_ds_info, NULL))) {
> ast_channel_unlock(chan);
> ast_log(LOG_WARNING, "No JACK_HOOK found to disable\n");
> return -1;
> }
>
> ast_channel_datastore_remove(chan, datastore);
>
> jack_data = datastore->data;
> ast_audiohook_detach(&jack_data->audiohook);
>
> /* Keep the channel locked while we destroy the datastore, so that
> we can
> * ensure that all of the jack stuff is stopped just in case another
> frame
> * tries to come through the audiohook callback. */
> ast_datastore_free(datastore);
>
> ast_channel_unlock(chan);
>
> return 0;
> }
>
> static int jack_hook_write(struct ast_channel *chan, const char *cmd, char
> *data,
> const char *value)
> {
> int res;
>
> if (!strcasecmp(value, "on"))
> res = enable_jack_hook(chan, data);
> else if (!strcasecmp(value, "off"))
> res = disable_jack_hook(chan);
> else {
> ast_log(LOG_ERROR, "'%s' is not a valid value for
> JACK_HOOK()\n", value);
> res = -1;
> }
>
> return res;
> }
>
> static struct ast_custom_function jack_hook_function = {
> .name = "JACK_HOOK",
> .synopsis = "Enable a jack hook on a channel",
> .syntax = "JACK_HOOK(<mode>,[options])",
> .desc =
> " The JACK_HOOK allows turning on or off jack connectivity to this
> channel.\n"
> "When the JACK_HOOK is turned on, jack ports will get created that
> allow\n"
> "access to the audio stream for this channel. The mode specifies
> which mode\n"
> "this hook should run in. A mode must be specified when turning the
> JACK_HOOK.\n"
> "on. However, all arguments are optional when turning it off.\n"
> "\n"
> " Valid modes are:\n"
> #if 0
> /* XXX TODO */
> " spy - Create a read-only audio hook. Only an output
> jack port will\n"
> " get created.\n"
> " whisper - Create a write-only audio hook. Only an input
> jack port will\n"
> " get created.\n"
> #endif
> " manipulate - Create a read/write audio hook. Both an input and
> an output\n"
> " jack port will get created. Audio from the
> channel will be\n"
> " sent out the output port and will be replaced by
> the audio\n"
> " coming in on the input port as it gets passed
> on.\n"
> "\n"
> " Valid options are:\n"
> COMMON_OPTIONS
> "\n"
> " Examples:\n"
> " To turn on the JACK_HOOK,\n"
> "
>
Set(JACK_HOOK(manipulate,i(pure_data_0:input0)o(pure_data_0:output0))=on)\n"
> " To turn off the JACK_HOOK,\n"
> " Set(JACK_HOOK()=off)\n"
> "",
> .write = jack_hook_write,
> };
>
> static int unload_module(void)
> {
> int res;
>
> res = ast_unregister_application(jack_app);
> res |= ast_custom_function_unregister(&jack_hook_function);
>
> return res;
> }
>
> static int load_module(void)
> {
> if (ast_register_application(jack_app, jack_exec, jack_synopsis,
> jack_desc)) {
> return AST_MODULE_LOAD_DECLINE;
> }
>
> if (ast_custom_function_register(&jack_hook_function)) {
> ast_unregister_application(jack_app);
> return AST_MODULE_LOAD_DECLINE;
> }
>
> return AST_MODULE_LOAD_SUCCESS;
> }
>
> AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "JACK Interface");
>
>
> -----Message d'origine-----
> De : asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] De la part de Fabien COMTE
> Envoyé : lundi 5 octobre 2009 16:18
> À : asterisk-users at lists.digium.com; asterisk-dev at lists.digium.com
> Objet : [asterisk-dev] Questions about app_jack.c
>
> Hello,
>
> My configuration is :
> Card 0 - kernel dummy sound card
> Card 1 - my soundcard
>
> I have a jackd running in background. My jackd launch command is :
> jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
> --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
> --outchannels 2 --dither triangular &
>
> 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to the
> kernel dummy sound card (allow me dial command). I do a call with a
> JACK_HOOK from app_jack.so, sound is sent but no one is received.
>
> My extensions.conf :
> exten => _0.,1,Answer
> exten =>
>
_0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu
> t)))=on)
> exten => _0.,n,Dial(SIP/freephonie-out/${EXTEN:1})
>
> Asterisk command :
> console dial 0xxxxxxxx
>
> 2) Jackd works well with anothers applications when I force them to use
jack
> as input/output. -> probably not a jack configuration problem.
>
> 3) If I kill jackd and I use chan_alsa.so with the real soundcard, it
works.
> -> probably not a network or sip configuration problem.
>
>
> 4) If I replace "f_buf[i] = s_buf[i] * (1.0 / SHRT_MAX);" with "f_buf[i] =
> 0.5 * sin(0.3454 * ((float) i));" in app_jack.c and I retry the test 2, I
> get test sound.
> It looks like no sound was read in channel...
>
> Do you have any idea ?
>
> Fabien
>
>
>
>
>
> _______________________________________________
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>
>
>
> _______________________________________________
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>
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