diff options
Diffstat (limited to 'main/audiohook.c')
| -rw-r--r-- | main/audiohook.c | 67 |
1 files changed, 37 insertions, 30 deletions
diff --git a/main/audiohook.c b/main/audiohook.c index 549ad31eb..33dad38f9 100644 --- a/main/audiohook.c +++ b/main/audiohook.c @@ -41,13 +41,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/slinfactory.h" #include "asterisk/frame.h" #include "asterisk/translate.h" +#include "asterisk/format_cache.h" #define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */ #define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */ struct ast_audiohook_translate { struct ast_trans_pvt *trans_pvt; - struct ast_format format; + struct ast_format *format; }; struct ast_audiohook_list { @@ -67,7 +68,7 @@ struct ast_audiohook_list { static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset) { - struct ast_format slin; + struct ast_format *slin; if (audiohook->hook_internal_samp_rate == rate) { return 0; @@ -75,7 +76,8 @@ static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate audiohook->hook_internal_samp_rate = rate; - ast_format_set(&slin, ast_format_slin_by_rate(rate), 0); + slin = ast_format_cache_get_slin_by_rate(rate); + /* Setup the factories that are needed for this audiohook type */ switch (audiohook->type) { case AST_AUDIOHOOK_TYPE_SPY: @@ -84,12 +86,13 @@ static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate ast_slinfactory_destroy(&audiohook->read_factory); ast_slinfactory_destroy(&audiohook->write_factory); } - ast_slinfactory_init_with_format(&audiohook->read_factory, &slin); - ast_slinfactory_init_with_format(&audiohook->write_factory, &slin); + ast_slinfactory_init_with_format(&audiohook->read_factory, slin); + ast_slinfactory_init_with_format(&audiohook->write_factory, slin); break; default: break; } + return 0; } @@ -143,6 +146,8 @@ int ast_audiohook_destroy(struct ast_audiohook *audiohook) if (audiohook->trans_pvt) ast_translator_free_path(audiohook->trans_pvt); + ao2_cleanup(audiohook->format); + /* Lock and trigger be gone! */ ast_cond_destroy(&audiohook->trigger); ast_mutex_destroy(&audiohook->lock); @@ -220,11 +225,11 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio short buf[samples]; struct ast_frame frame = { .frametype = AST_FRAME_VOICE, + .subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate), .data.ptr = buf, .datalen = sizeof(buf), .samples = samples, }; - ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0); /* Ensure the factory is able to give us the samples we want */ if (samples > ast_slinfactory_available(factory)) { @@ -254,7 +259,6 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho .datalen = sizeof(buf1), .samples = samples, }; - ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0); /* Make sure both factories have the required samples */ usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0); @@ -346,6 +350,8 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho /* Make the final buffer part of the frame, so it gets duplicated fine */ frame.data.ptr = final_buf; + frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate); + /* Yahoo, a combined copy of the audio! */ return ast_frdup(&frame); } @@ -353,17 +359,17 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference) { struct ast_frame *read_frame = NULL, *final_frame = NULL; - struct ast_format tmp_fmt; + struct ast_format *slin; int samples_converted; /* the number of samples requested is based on the format they are requesting. Inorder * to process this correctly samples must be converted to our internal sample rate */ - if (audiohook->hook_internal_samp_rate == ast_format_rate(format)) { + if (audiohook->hook_internal_samp_rate == ast_format_get_sample_rate(format)) { samples_converted = samples; - } else if (audiohook->hook_internal_samp_rate > ast_format_rate(format)) { - samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_rate(format)); + } else if (audiohook->hook_internal_samp_rate > ast_format_get_sample_rate(format)) { + samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_get_sample_rate(format)); } else { - samples_converted = samples * (ast_format_rate(format) / (float) audiohook->hook_internal_samp_rate); + samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate); } if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? @@ -372,21 +378,23 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio return NULL; } + slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate); + /* If they don't want signed linear back out, we'll have to send it through the translation path */ - if (format->id != ast_format_slin_by_rate(audiohook->hook_internal_samp_rate)) { + if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) { /* Rebuild translation path if different format then previously */ - if (ast_format_cmp(format, &audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) { + if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) { if (audiohook->trans_pvt) { ast_translator_free_path(audiohook->trans_pvt); audiohook->trans_pvt = NULL; } /* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */ - if (!(audiohook->trans_pvt = ast_translator_build_path(format, ast_format_set(&tmp_fmt, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0)))) { + if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) { ast_frfree(read_frame); return NULL; } - ast_format_copy(&audiohook->format, format); + ao2_replace(audiohook->format, format); } /* Convert to requested format, and allow the read in frame to be freed */ final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1); @@ -752,8 +760,7 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]); struct ast_frame *new_frame = frame; - struct ast_format tmp_fmt; - enum ast_format_id slin_id; + struct ast_format *slin; /* If we are capable of maintaining doing samplerates other that 8khz, update * the internal audiohook_list's rate and higher samplerate audio arrives. By @@ -761,24 +768,24 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l * as the are written and read from. */ if (audiohook_list->native_slin_compatible) { audiohook_list->list_internal_samp_rate = - MAX(ast_format_rate(&frame->subclass.format), audiohook_list->list_internal_samp_rate); + MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate); } - slin_id = ast_format_slin_by_rate(audiohook_list->list_internal_samp_rate); - - if (frame->subclass.format.id == slin_id) { + slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate); + if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) { return new_frame; } - if (ast_format_cmp(&frame->subclass.format, &in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) { + if (ast_format_cmp(frame->subclass.format, in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) { if (in_translate->trans_pvt) { ast_translator_free_path(in_translate->trans_pvt); } - if (!(in_translate->trans_pvt = ast_translator_build_path(ast_format_set(&tmp_fmt, slin_id, 0), &frame->subclass.format))) { + if (!(in_translate->trans_pvt = ast_translator_build_path(slin, frame->subclass.format))) { return NULL; } - ast_format_copy(&in_translate->format, &frame->subclass.format); + ao2_replace(in_translate->format, frame->subclass.format); } + if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) { return NULL; } @@ -791,16 +798,16 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook { struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]); struct ast_frame *outframe = NULL; - if (ast_format_cmp(&slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { + if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { /* rebuild translators if necessary */ - if (ast_format_cmp(&out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { + if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { if (out_translate->trans_pvt) { ast_translator_free_path(out_translate->trans_pvt); } - if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, &slin_frame->subclass.format))) { + if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) { return NULL; } - ast_format_copy(&out_translate->format, outformat); + ao2_replace(out_translate->format, outformat); } /* translate back to the format the frame came in as. */ if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) { @@ -924,7 +931,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */ if (middle_frame_manipulated) { - if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, &start_frame->subclass.format))) { + if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) { /* translation failed, so just pass back the input frame */ end_frame = start_frame; } |