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app_jack: Support audio with a sampling rate higher than 8kHz

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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)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2014-06-26 12:43:05 +00:00
parent f27074eeb7
commit 22e62ac6f6
2 changed files with 50 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

66
apps/app_jack.c
View File

@@ -61,7 +61,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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 @@ struct jack_data {
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 @@ static int alloc_resampler(struct jack_data *jack_data, int input)
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 @@ static void handle_input(void *buf, jack_nframes_t nframes,
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);
}
}
@@ -392,6 +394,28 @@ static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
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;
} else {
@@ -400,10 +424,10 @@ static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data)
ast_channel_unlock(chan);
}
if (!(jack_data->output_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)))
if (!(jack_data->input_rb = jack_ringbuffer_create(ringbuffer_size)))
return -1;
if (jack_data->no_start_server)
@@ -573,10 +597,9 @@ static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
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 int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
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 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d
.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 @@ static int jack_exec(struct ast_channel *chan, const char *data)
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 @@ static int jack_hook_callback(struct ast_audiohook *audiohook, struct ast_channe
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))) {
@@ -842,6 +859,13 @@ static int jack_hook_callback(struct ast_audiohook *audiohook, struct ast_channe
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);
handle_jack_audio(chan, jack_data, frame);
@@ -888,7 +912,7 @@ static int enable_jack_hook(struct ast_channel *chan, char *data)
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;