Bring in the code from team/russell/jack/.

Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function.  Both
interfaces create an input and output JACK port.  The application makes
these ports the endpoint of the call.  The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
audiohook on the channel.  This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio.  This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.

In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/).  I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98628 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 973 insertions, 2 deletions

diff --git a/CHANGES b/CHANGES
index a34585d1b..b8f6897d8 100644
--- a/CHANGES
+++ b/CHANGES
@@ -284,7 +284,7 @@ Queue changes
      rules in queuerules.conf. See configs/queuerules.conf.sample for details
   * Added a new parameter for member definition, called state_interface. This may be
     used so that a member may be called via one interface but have a different interface's
-	device state reported.
+    device state reported.
 
 MeetMe Changes
 --------------
@@ -460,4 +460,16 @@ Miscellaneous
      is currently only intended to provision a single user account per phone.
      An example profile and set of templates for Polycom phones is provided.
      NOTE: Polycom firmware is not included, but should be placed in
-     AST_DATA_DIR/phoneprov/configs to match up with the included templates. 
+     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
+  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
+     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
+     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
+     interfaces create an input and output JACK port.  The application makes
+     these ports the endpoint of the call.  The audio coming from the channel
+     goes out the output port and whatever comes back in on the input port is
+     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
+     audiohook on the channel.  This lets you run the audio coming from a
+     channel through JACK, and whatever comes back in is what gets forwarded
+     on as the channel's audio.  This is very useful for building custom
+     vocoders or doing recording or analysis of the channel's audio in another
+     application.
diff --git a/apps/app_jack.c b/apps/app_jack.c
new file mode 100644
index 000000000..46d00f6d5
--- /dev/null
+++ b/apps/app_jack.c
@@ -0,0 +1,959 @@
+/*
+ * 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.
+ *
+ * \arg http://www.jackaudio.org/
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<depend>jack</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <limits.h>
+
+#include <jack/jack.h>
+#include <jack/ringbuffer.h>
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/strings.h"
+#include "asterisk/lock.h"
+#include "asterisk/libresample.h"
+#include "asterisk/app.h"
+#include "asterisk/pbx.h"
+#include "asterisk/audiohook.h"
+
+#define RESAMPLE_QUALITY 0
+
+#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"
+
+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(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;
+	/*! 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] = 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_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 *chan_name;
+	jack_status_t status = 0;
+
+	ast_channel_lock(chan);
+	chan_name = ast_strdupa(chan->name);
+	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 (!ast_strlen_zero(jack_data->server_name)) {
+		jack_data->client = jack_client_open(chan_name, JackServerName, &status,
+			jack_data->server_name);
+	} else {
+		jack_data->client = jack_client_open(chan_name, JackNullOption, &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;
+	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 = AST_FORMAT_SLINEAR,
+		.src = "JACK",
+		.data = 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 : 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, 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),
+};
+
+enum {
+	OPT_ARG_SERVER_NAME,
+	OPT_ARG_INPUT_PORT,
+	OPT_ARG_OUTPUT_PORT,
+	/* 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),
+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_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;
+		}
+	}
+
+	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 (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;
+	}
+
+	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_channel_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_channel_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");