app_jack.c

Go to the documentation of this file.

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2007 - 2008, Russell Bryant
 *
 * Russell Bryant <russell@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@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.
 *
 * \extref 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>
    <support_level>extended</support_level>
 ***/
 
#include "asterisk.h"
 
#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"
#include "asterisk/format_cache.h"
 
#define RESAMPLE_QUALITY 1
 
/* The number of frames the ringbuffers can store. The actual size is RINGBUFFER_FRAME_CAPACITY * jack_data->frame_datalen */
#define RINGBUFFER_FRAME_CAPACITY 100
 
/*! \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"
/*** DOCUMENTATION
    <application name="JACK" language="en_US">
        <since>
            <version>1.6.0</version>
        </since>
        <synopsis>
            Jack Audio Connection Kit
        </synopsis>
        <syntax>
            <parameter name="options" required="false">
                <optionlist>
                    <option name="s">
                        <argument name="name" required="true">
                            <para>Connect to the specified jack server name</para>
                        </argument>
                    </option>
                    <option name="i">
                        <argument name="name" required="true">
                            <para>Connect the output port that gets created to the specified jack input port</para>
                        </argument>
                    </option>
                    <option name="o">
                        <argument name="name" required="true">
                            <para>Connect the input port that gets created to the specified jack output port</para>
                        </argument>
                    </option>
                    <option name="c">
                        <argument name="name" required="true">
                            <para>By default, Asterisk will use the channel name for the jack client name.</para>
                            <para>Use this option to specify a custom client name.</para>
                        </argument>
                    </option>
                </optionlist>
            </parameter>
        </syntax>
        <description>
            <para>When executing this application, two jack ports will be created;
            one input and one output. Other applications can be hooked up to
            these ports to access audio coming from, or being send to the channel.</para>
        </description>
    </application>
 ***/
 
static const char jack_app[] = "JACK";
 
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;
    struct ast_format *audiohook_format;
    unsigned int audiohook_rate;
    unsigned int frame_datalen;
    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, ast_str_buffer(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);
 
    to_srate = input ? jack_data->audiohook_rate : jack_srate;
    from_srate = input ? jack_srate : jack_data->audiohook_rate;
 
    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] = 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);
    if (res != write_len) {
        ast_log(LOG_WARNING, "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;
 
    unsigned int channel_rate;
 
    unsigned int ringbuffer_size;
 
    /* Deducing audiohook sample rate from channel format
       ATTENTION: Might be problematic, if channel has different sampling than used by audiohook!
    */
    channel_rate = ast_format_get_sample_rate(ast_channel_readformat(chan));
    jack_data->audiohook_format = ast_format_cache_get_slin_by_rate(channel_rate);
    jack_data->audiohook_rate = ast_format_get_sample_rate(jack_data->audiohook_format);
 
    /* Guessing frame->datalen assuming a ptime of 20ms */
    jack_data->frame_datalen = jack_data->audiohook_rate / 50;
 
    ringbuffer_size = jack_data->frame_datalen * RINGBUFFER_FRAME_CAPACITY;
 
    ast_debug(1, "Audiohook parameters: slin-format:%s, rate:%d, frame-len:%d, ringbuffer_size: %d\n",
        ast_format_get_name(jack_data->audiohook_format), jack_data->audiohook_rate, jack_data->frame_datalen, ringbuffer_size);
 
    if (!ast_strlen_zero(jack_data->client_name)) {
        client_name = jack_data->client_name;
    } else {
        ast_channel_lock(chan);
        client_name = ast_strdupa(ast_channel_name(chan));
        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)))
        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));
        }
 
        jack_free(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));
        }
 
        jack_free(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));
 
    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_log(LOG_WARNING, "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 available 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.
 */
static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data,
    struct ast_frame *out_frame)
{
    short buf[jack_data->frame_datalen];
    struct ast_frame f = {
        .frametype = AST_FRAME_VOICE,
        .subclass.format = jack_data->audiohook_format,
        .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_with_stringfields(1, struct jack_data, 32))) {
        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, jack_data->audiohook_format)) {
        destroy_jack_data(jack_data);
        return -1;
    }
 
    if (ast_set_write_format(chan, jack_data->audiohook_format)) {
        destroy_jack_data(jack_data);
        return -1;
    }
 
    while (!jack_data->stop) {
        struct ast_frame *f;
 
        if (ast_waitfor(chan, -1) < 0) {
            break;
        }
 
        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;
 
    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", ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    jack_data = datastore->data;
 
    if (ast_format_cmp(frame->subclass.format, jack_data->audiohook_format) == AST_FORMAT_CMP_NOT_EQUAL) {
        ast_log(LOG_WARNING, "Expected frame in %s for the audiohook, but got format %s\n",
            ast_format_get_name(jack_data->audiohook_format),
            ast_format_get_name(frame->subclass.format));
        ast_channel_unlock(chan);
        return 0;
    }
 
    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", ast_channel_name(chan));
        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", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
    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);
    }
 
    if (datastore) {
        datastore->data = NULL;
        ast_datastore_free(datastore);
    }
 
    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 (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    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_EXTENDED(ASTERISK_GPL_KEY, "JACK Interface");