diff options
author | Matthew Jordan <mjordan@digium.com> | 2014-06-26 12:43:05 +0000 |
---|---|---|
committer | Matthew Jordan <mjordan@digium.com> | 2014-06-26 12:43:05 +0000 |
commit | 22e62ac6f640f6442968e5ab9146d3a7fe496f95 (patch) | |
tree | c0a567f7ef54b3a2129401f997f2d06b346602de | |
parent | f27074eeb7319e5c12771fe76d81951c730ded46 (diff) |
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r-- | CHANGES | 5 | ||||
-rw-r--r-- | apps/app_jack.c | 66 |
2 files changed, 50 insertions, 21 deletions
@@ -35,6 +35,11 @@ chan_dahdi * 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 ------------ ------------------------------------------------------------------------------ diff --git a/apps/app_jack.c b/apps/app_jack.c index f32c59ff0..9c59ceaf4 100644 --- a/apps/app_jack.c +++ b/apps/app_jack.c @@ -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; |