diff options
Diffstat (limited to 'apps/app_jack.c')
| -rw-r--r-- | apps/app_jack.c | 29 |
1 files changed, 14 insertions, 15 deletions
diff --git a/apps/app_jack.c b/apps/app_jack.c index 9c59ceaf4..8adfbc4cb 100644 --- a/apps/app_jack.c +++ b/apps/app_jack.c @@ -58,6 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/app.h" #include "asterisk/pbx.h" #include "asterisk/audiohook.h" +#include "asterisk/format_cache.h" #define RESAMPLE_QUALITY 1 @@ -129,7 +130,7 @@ 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; + struct ast_format *audiohook_format; unsigned int audiohook_rate; unsigned int frame_datalen; void *output_resampler; @@ -394,7 +395,6 @@ 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; @@ -402,19 +402,17 @@ static int init_jack_data(struct ast_channel *chan, struct jack_data *jack_data) /* 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); + 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:%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); + 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; @@ -628,12 +626,12 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d 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), }; - ast_format_set(&f.subclass.format, jack_data->audiohook_format_id, 0); for (;;) { size_t res, read_len; @@ -778,12 +776,12 @@ static int jack_exec(struct ast_channel *chan, const char *data) return -1; } - if (ast_set_read_format_by_id(chan, jack_data->audiohook_format_id)) { + if (ast_set_read_format(chan, jack_data->audiohook_format)) { destroy_jack_data(jack_data); return -1; } - if (ast_set_write_format_by_id(chan, jack_data->audiohook_format_id)) { + if (ast_set_write_format(chan, jack_data->audiohook_format)) { destroy_jack_data(jack_data); return -1; } @@ -859,9 +857,10 @@ 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)); + 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; } |