[svn-commits] mjordan: trunk r417360 - in /trunk: CHANGES apps/app_jack.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jun 26 07:43:11 CDT 2014
Author: mjordan
Date: Thu Jun 26 07:43:05 2014
New Revision: 417360
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=417360
Log:
app_jack: Support audio with a sampling rate higher than 8kHz
This patch enables the jack-audiohook to cope with dynamic sampling rates from
and to Asterisk. Information from the channel is taken to derive the channel's
sampling rate, suiting SLINxx format and frame->datalen.
There are stil a few limitations after this patch:
* Required information is taken from the channel during initialization as
the audiohook does not provide this information.
Audiohook.internal_sampl_rate(...) is set later, but no callback is available
to inform app_jack.
* Frame.datalen is computed using "rate / 50" assuming a ptime of 20ms.
There is no internal API available to determine datalen for a SLINxx.
* Ringbuffer size is now dynamic depending on the value of frame.datalen
(see above) and the number of frames, which are in RINGBUFFER_FRAME_CAPACITY,
that need to fit.
Review: https://reviewboard.asterisk.org/r/3618
Note that the patch being committed here is based on the patch posted on
ASTERISK-23836. However, Matthis Schmieder also provided a patch to enable
this functionality, and that patch is noted below.
ASTERISK-20696 #close
Reported by: Matthis Schmieder
patches:
app_jack.patch uploaded by Matthis Schmieder (License 6445)
ASTERISK-23836 #close
Reported by: Dennis Guse
patches:
patch-app_jack.c uploaded by Dennis Guse (License 6513)
Modified:
trunk/CHANGES
trunk/apps/app_jack.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=417360&r1=417359&r2=417360
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Thu Jun 26 07:43:05 2014
@@ -34,6 +34,11 @@
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
+
+JACK_HOOK
+------------------
+ * The JACK_HOOK function now supports audio with a sample rate higher than
+ 8kHz.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
Modified: trunk/apps/app_jack.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_jack.c?view=diff&rev=417360&r1=417359&r2=417360
==============================================================================
--- trunk/apps/app_jack.c (original)
+++ trunk/apps/app_jack.c Thu Jun 26 07:43:05 2014
@@ -61,7 +61,8 @@
#define RESAMPLE_QUALITY 1
-#define RINGBUFFER_SIZE 16384
+/* 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 \
@@ -128,6 +129,9 @@
jack_port_t *output_port;
jack_ringbuffer_t *input_rb;
jack_ringbuffer_t *output_rb;
+ enum ast_format_id audiohook_format_id;
+ unsigned int audiohook_rate;
+ unsigned int frame_datalen;
void *output_resampler;
double output_resample_factor;
void *input_resampler;
@@ -201,10 +205,8 @@
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;
+ 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;
@@ -289,7 +291,7 @@
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",
+ ast_log(LOG_WARNING, "Tried to write %d bytes to the ringbuffer, but only wrote %d\n",
(int) sizeof(s_buf), (int) res);
}
}
@@ -391,6 +393,28 @@
const char *client_name;
jack_status_t status = 0;
jack_options_t jack_options = JackNullOption;
+
+ struct ast_format format_slin;
+ 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_rate(ast_channel_readformat(chan));
+ jack_data->audiohook_format_id = ast_format_slin_by_rate(channel_rate);
+
+ ast_format_set(&format_slin, jack_data->audiohook_format_id, 0);
+ jack_data->audiohook_rate = ast_format_rate(&format_slin);
+
+ /* 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:%d, rate:%d, frame-len:%d, ringbuffer_size: %d\n",
+ jack_data->audiohook_format_id, jack_data->audiohook_rate, jack_data->frame_datalen, ringbuffer_size);
if (!ast_strlen_zero(jack_data->client_name)) {
client_name = jack_data->client_name;
@@ -400,10 +424,10 @@
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)))
+ 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)
@@ -573,10 +597,9 @@
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",
+ 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;
}
@@ -602,7 +625,7 @@
static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data,
struct ast_frame *out_frame)
{
- short buf[160];
+ short buf[jack_data->frame_datalen];
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
.src = "JACK",
@@ -610,7 +633,7 @@
.datalen = sizeof(buf),
.samples = ARRAY_LEN(buf),
};
- ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
+ ast_format_set(&f.subclass.format, jack_data->audiohook_format_id, 0);
for (;;) {
size_t res, read_len;
@@ -755,12 +778,12 @@
return -1;
}
- if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+ if (ast_set_read_format_by_id(chan, jack_data->audiohook_format_id)) {
destroy_jack_data(jack_data);
return -1;
}
- if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+ if (ast_set_write_format_by_id(chan, jack_data->audiohook_format_id)) {
destroy_jack_data(jack_data);
return -1;
}
@@ -826,12 +849,6 @@
if (frame->frametype != AST_FRAME_VOICE)
return 0;
- if (frame->subclass.format.id != AST_FORMAT_SLINEAR) {
- ast_log(LOG_WARNING, "Expected frame in SLINEAR for the audiohook, but got format %s\n",
- ast_getformatname(&frame->subclass.format));
- return 0;
- }
-
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &jack_hook_ds_info, NULL))) {
@@ -841,6 +858,13 @@
}
jack_data = datastore->data;
+
+ if (frame->subclass.format.id != jack_data->audiohook_format_id) {
+ ast_log(LOG_WARNING, "Expected frame in SLINEAR with id %d for the audiohook, but got format %s\n",
+ jack_data->audiohook_format_id, ast_getformatname(&frame->subclass.format));
+ ast_channel_unlock(chan);
+ return 0;
+ }
queue_voice_frame(jack_data, frame);
@@ -888,7 +912,7 @@
goto return_error;
jack_data->has_audiohook = 1;
- ast_audiohook_init(&jack_data->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "JACK_HOOK", 0);
+ 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;
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