Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1022 lines
27 KiB
C
1022 lines
27 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2007 - 2008, Russell Bryant
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*
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* Russell Bryant <russell@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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* \brief Jack Application
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*
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* \author Russell Bryant <russell@digium.com>
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*
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* This is an application to connect an Asterisk channel to an input
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* and output jack port so that the audio can be processed through
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* another application, or to play audio from another application.
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*
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* \extref http://www.jackaudio.org/
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*
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* \note To install libresample, check it out of the following repository:
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* <code>$ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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<depend>jack</depend>
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<depend>resample</depend>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <limits.h>
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#include <jack/jack.h>
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#include <jack/ringbuffer.h>
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#include <libresample.h>
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/strings.h"
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#include "asterisk/lock.h"
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#include "asterisk/app.h"
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#include "asterisk/pbx.h"
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#include "asterisk/audiohook.h"
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#define RESAMPLE_QUALITY 1
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#define RINGBUFFER_SIZE 16384
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/*! \brief Common options between the Jack() app and JACK_HOOK() function */
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#define COMMON_OPTIONS \
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" s(<name>) - Connect to the specified jack server name.\n" \
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" i(<name>) - Connect the output port that gets created to the specified\n" \
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" jack input port.\n" \
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" o(<name>) - Connect the input port that gets created to the specified\n" \
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" jack output port.\n" \
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" n - Do not automatically start the JACK server if it is not already\n" \
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" running.\n" \
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" c(<name>) - By default, Asterisk will use the channel name for the jack client\n" \
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" name. Use this option to specify a custom client name.\n"
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/*** DOCUMENTATION
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<application name="JACK" language="en_US">
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<synopsis>
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Jack Audio Connection Kit
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</synopsis>
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<syntax>
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<parameter name="options" required="false">
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<optionlist>
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<option name="s">
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<argument name="name" required="true">
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<para>Connect to the specified jack server name</para>
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</argument>
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</option>
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<option name="i">
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<argument name="name" required="true">
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<para>Connect the output port that gets created to the specified jack input port</para>
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</argument>
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</option>
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<option name="o">
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<argument name="name" required="true">
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<para>Connect the input port that gets created to the specified jack output port</para>
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</argument>
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</option>
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<option name="c">
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<argument name="name" required="true">
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<para>By default, Asterisk will use the channel name for the jack client name.</para>
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<para>Use this option to specify a custom client name.</para>
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</argument>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>When executing this application, two jack ports will be created;
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one input and one output. Other applications can be hooked up to
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these ports to access audio coming from, or being send to the channel.</para>
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</description>
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</application>
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***/
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static const char jack_app[] = "JACK";
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struct jack_data {
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AST_DECLARE_STRING_FIELDS(
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AST_STRING_FIELD(server_name);
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AST_STRING_FIELD(client_name);
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AST_STRING_FIELD(connect_input_port);
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AST_STRING_FIELD(connect_output_port);
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);
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jack_client_t *client;
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jack_port_t *input_port;
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jack_port_t *output_port;
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jack_ringbuffer_t *input_rb;
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jack_ringbuffer_t *output_rb;
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void *output_resampler;
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double output_resample_factor;
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void *input_resampler;
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double input_resample_factor;
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unsigned int stop:1;
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unsigned int has_audiohook:1;
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unsigned int no_start_server:1;
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/*! Only used with JACK_HOOK */
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struct ast_audiohook audiohook;
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};
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static const struct {
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jack_status_t status;
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const char *str;
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} jack_status_table[] = {
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{ JackFailure, "Failure" },
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{ JackInvalidOption, "Invalid Option" },
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{ JackNameNotUnique, "Name Not Unique" },
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{ JackServerStarted, "Server Started" },
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{ JackServerFailed, "Server Failed" },
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{ JackServerError, "Server Error" },
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{ JackNoSuchClient, "No Such Client" },
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{ JackLoadFailure, "Load Failure" },
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{ JackInitFailure, "Init Failure" },
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{ JackShmFailure, "Shared Memory Access Failure" },
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{ JackVersionError, "Version Mismatch" },
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};
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static const char *jack_status_to_str(jack_status_t status)
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{
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int i;
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for (i = 0; i < ARRAY_LEN(jack_status_table); i++) {
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if (jack_status_table[i].status == status)
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return jack_status_table[i].str;
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}
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return "Unknown Error";
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}
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static void log_jack_status(const char *prefix, jack_status_t status)
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{
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struct ast_str *str = ast_str_alloca(512);
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int i, first = 0;
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for (i = 0; i < (sizeof(status) * 8); i++) {
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if (!(status & (1 << i)))
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continue;
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if (!first) {
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ast_str_set(&str, 0, "%s", jack_status_to_str((1 << i)));
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first = 1;
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} else
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ast_str_append(&str, 0, ", %s", jack_status_to_str((1 << i)));
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}
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ast_log(LOG_NOTICE, "%s: %s\n", prefix, ast_str_buffer(str));
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}
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static int alloc_resampler(struct jack_data *jack_data, int input)
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{
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double from_srate, to_srate, jack_srate;
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void **resampler;
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double *resample_factor;
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if (input && jack_data->input_resampler)
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return 0;
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if (!input && jack_data->output_resampler)
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return 0;
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jack_srate = jack_get_sample_rate(jack_data->client);
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/* XXX Hard coded 8 kHz */
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to_srate = input ? 8000.0 : jack_srate;
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from_srate = input ? jack_srate : 8000.0;
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resample_factor = input ? &jack_data->input_resample_factor :
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&jack_data->output_resample_factor;
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if (from_srate == to_srate) {
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/* Awesome! The jack sample rate is the same as ours.
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* Resampling isn't needed. */
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*resample_factor = 1.0;
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return 0;
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}
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*resample_factor = to_srate / from_srate;
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resampler = input ? &jack_data->input_resampler :
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&jack_data->output_resampler;
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if (!(*resampler = resample_open(RESAMPLE_QUALITY,
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*resample_factor, *resample_factor))) {
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ast_log(LOG_ERROR, "Failed to open %s resampler\n",
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input ? "input" : "output");
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return -1;
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}
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return 0;
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}
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/*!
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* \brief Handle jack input port
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*
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* Read nframes number of samples from the input buffer, resample it
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* if necessary, and write it into the appropriate ringbuffer.
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*/
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static void handle_input(void *buf, jack_nframes_t nframes,
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struct jack_data *jack_data)
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{
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short s_buf[nframes];
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float *in_buf = buf;
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size_t res;
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int i;
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size_t write_len = sizeof(s_buf);
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if (jack_data->input_resampler) {
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int total_in_buf_used = 0;
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int total_out_buf_used = 0;
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float f_buf[nframes + 1];
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memset(f_buf, 0, sizeof(f_buf));
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while (total_in_buf_used < nframes) {
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int in_buf_used;
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int out_buf_used;
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out_buf_used = resample_process(jack_data->input_resampler,
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jack_data->input_resample_factor,
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&in_buf[total_in_buf_used], nframes - total_in_buf_used,
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0, &in_buf_used,
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&f_buf[total_out_buf_used], ARRAY_LEN(f_buf) - total_out_buf_used);
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if (out_buf_used < 0)
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break;
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total_out_buf_used += out_buf_used;
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total_in_buf_used += in_buf_used;
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if (total_out_buf_used == ARRAY_LEN(f_buf)) {
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ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size, "
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"nframes '%d', total_out_buf_used '%d'\n", nframes, total_out_buf_used);
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break;
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}
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}
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for (i = 0; i < total_out_buf_used; i++)
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s_buf[i] = f_buf[i] * (SHRT_MAX / 1.0);
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write_len = total_out_buf_used * sizeof(int16_t);
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} else {
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/* No resampling needed */
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for (i = 0; i < nframes; i++)
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s_buf[i] = in_buf[i] * (SHRT_MAX / 1.0);
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}
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res = jack_ringbuffer_write(jack_data->input_rb, (const char *) s_buf, write_len);
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if (res != write_len) {
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ast_debug(2, "Tried to write %d bytes to the ringbuffer, but only wrote %d\n",
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(int) sizeof(s_buf), (int) res);
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}
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}
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/*!
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* \brief Handle jack output port
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*
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* Read nframes number of samples from the ringbuffer and write it out to the
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* output port buffer.
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*/
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static void handle_output(void *buf, jack_nframes_t nframes,
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struct jack_data *jack_data)
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{
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size_t res, len;
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len = nframes * sizeof(float);
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res = jack_ringbuffer_read(jack_data->output_rb, buf, len);
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if (len != res) {
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ast_debug(2, "Wanted %d bytes to send to the output port, "
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"but only got %d\n", (int) len, (int) res);
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}
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}
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static int jack_process(jack_nframes_t nframes, void *arg)
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{
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struct jack_data *jack_data = arg;
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void *input_port_buf, *output_port_buf;
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if (!jack_data->input_resample_factor)
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alloc_resampler(jack_data, 1);
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input_port_buf = jack_port_get_buffer(jack_data->input_port, nframes);
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handle_input(input_port_buf, nframes, jack_data);
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output_port_buf = jack_port_get_buffer(jack_data->output_port, nframes);
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handle_output(output_port_buf, nframes, jack_data);
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return 0;
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}
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static void jack_shutdown(void *arg)
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{
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struct jack_data *jack_data = arg;
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jack_data->stop = 1;
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}
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static struct jack_data *destroy_jack_data(struct jack_data *jack_data)
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{
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if (jack_data->input_port) {
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jack_port_unregister(jack_data->client, jack_data->input_port);
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jack_data->input_port = NULL;
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}
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if (jack_data->output_port) {
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jack_port_unregister(jack_data->client, jack_data->output_port);
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jack_data->output_port = NULL;
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}
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if (jack_data->client) {
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jack_client_close(jack_data->client);
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jack_data->client = NULL;
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}
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if (jack_data->input_rb) {
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jack_ringbuffer_free(jack_data->input_rb);
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jack_data->input_rb = NULL;
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}
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if (jack_data->output_rb) {
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jack_ringbuffer_free(jack_data->output_rb);
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jack_data->output_rb = NULL;
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}
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if (jack_data->output_resampler) {
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resample_close(jack_data->output_resampler);
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jack_data->output_resampler = NULL;
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}
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if (jack_data->input_resampler) {
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resample_close(jack_data->input_resampler);
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jack_data->input_resampler = NULL;
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}
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if (jack_data->has_audiohook)
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ast_audiohook_destroy(&jack_data->audiohook);
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ast_string_field_free_memory(jack_data);
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ast_free(jack_data);
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return NULL;
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}
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static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
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{
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const char *client_name;
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jack_status_t status = 0;
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jack_options_t jack_options = JackNullOption;
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if (!ast_strlen_zero(jack_data->client_name)) {
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client_name = jack_data->client_name;
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} else {
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ast_channel_lock(chan);
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client_name = ast_strdupa(chan->name);
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ast_channel_unlock(chan);
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}
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if (!(jack_data->output_rb = jack_ringbuffer_create(RINGBUFFER_SIZE)))
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return -1;
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if (!(jack_data->input_rb = jack_ringbuffer_create(RINGBUFFER_SIZE)))
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return -1;
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if (jack_data->no_start_server)
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jack_options |= JackNoStartServer;
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if (!ast_strlen_zero(jack_data->server_name)) {
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jack_options |= JackServerName;
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jack_data->client = jack_client_open(client_name, jack_options, &status,
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jack_data->server_name);
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} else {
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jack_data->client = jack_client_open(client_name, jack_options, &status);
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}
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if (status)
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log_jack_status("Client Open Status", status);
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if (!jack_data->client)
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return -1;
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jack_data->input_port = jack_port_register(jack_data->client, "input",
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JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0);
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if (!jack_data->input_port) {
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ast_log(LOG_ERROR, "Failed to create input port for jack port\n");
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return -1;
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}
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jack_data->output_port = jack_port_register(jack_data->client, "output",
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JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, 0);
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if (!jack_data->output_port) {
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ast_log(LOG_ERROR, "Failed to create output port for jack port\n");
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return -1;
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}
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if (jack_set_process_callback(jack_data->client, jack_process, jack_data)) {
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ast_log(LOG_ERROR, "Failed to register process callback with jack client\n");
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return -1;
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}
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jack_on_shutdown(jack_data->client, jack_shutdown, jack_data);
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if (jack_activate(jack_data->client)) {
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ast_log(LOG_ERROR, "Unable to activate jack client\n");
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return -1;
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}
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while (!ast_strlen_zero(jack_data->connect_input_port)) {
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const char **ports;
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int i;
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ports = jack_get_ports(jack_data->client, jack_data->connect_input_port,
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NULL, JackPortIsInput);
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if (!ports) {
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ast_log(LOG_ERROR, "No input port matching '%s' was found\n",
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jack_data->connect_input_port);
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break;
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}
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for (i = 0; ports[i]; i++) {
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ast_debug(1, "Found port '%s' that matched specified input port '%s'\n",
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ports[i], jack_data->connect_input_port);
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}
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if (jack_connect(jack_data->client, jack_port_name(jack_data->output_port), ports[0])) {
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ast_log(LOG_ERROR, "Failed to connect '%s' to '%s'\n", ports[0],
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jack_port_name(jack_data->output_port));
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} else {
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ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
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jack_port_name(jack_data->output_port));
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}
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free((void *) ports);
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break;
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}
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while (!ast_strlen_zero(jack_data->connect_output_port)) {
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const char **ports;
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int i;
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ports = jack_get_ports(jack_data->client, jack_data->connect_output_port,
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NULL, JackPortIsOutput);
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if (!ports) {
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ast_log(LOG_ERROR, "No output port matching '%s' was found\n",
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jack_data->connect_output_port);
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break;
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}
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for (i = 0; ports[i]; i++) {
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ast_debug(1, "Found port '%s' that matched specified output port '%s'\n",
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ports[i], jack_data->connect_output_port);
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}
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if (jack_connect(jack_data->client, ports[0], jack_port_name(jack_data->input_port))) {
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ast_log(LOG_ERROR, "Failed to connect '%s' to '%s'\n", ports[0],
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jack_port_name(jack_data->input_port));
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} else {
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ast_debug(1, "Connected '%s' to '%s'\n", ports[0],
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jack_port_name(jack_data->input_port));
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}
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free((void *) ports);
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break;
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}
|
|
|
|
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));
|
|
|
|
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;
|
|
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] = s_buf[i] * (1.0 / 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.codec = 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, const char *data)
|
|
{
|
|
struct jack_data *jack_data;
|
|
|
|
if (!(jack_data = jack_data_alloc()))
|
|
return -1;
|
|
|
|
if (!ast_strlen_zero(data) && handle_options(jack_data, data)) {
|
|
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.integer == 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 (direction != AST_AUDIOHOOK_DIRECTION_READ)
|
|
return 0;
|
|
|
|
if (frame->frametype != AST_FRAME_VOICE)
|
|
return 0;
|
|
|
|
if (frame->subclass.codec != AST_FORMAT_SLINEAR) {
|
|
ast_log(LOG_WARNING, "Expected frame in SLINEAR for the audiohook, but got format %s\n",
|
|
ast_getformatname(frame->subclass.codec));
|
|
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;
|
|
|
|
queue_voice_frame(jack_data, frame);
|
|
|
|
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_xml(jack_app, jack_exec)) {
|
|
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");
|