Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets"

changes/93/493/2
Matt Jordan 10 years ago committed by Gerrit Code Review
commit 8e083830e2

@ -63,6 +63,7 @@ enum ast_audiohook_flags {
AST_AUDIOHOOK_SMALL_QUEUE = (1 << 4),
AST_AUDIOHOOK_MUTE_READ = (1 << 5), /*!< audiohook should be mute frames read */
AST_AUDIOHOOK_MUTE_WRITE = (1 << 6), /*!< audiohook should be mute frames written */
AST_AUDIOHOOK_COMPATIBLE = (1 << 7), /*!< is the audiohook native slin compatible */
};
enum ast_audiohook_init_flags {

@ -46,6 +46,8 @@ ASTERISK_REGISTER_FILE()
#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. */
#define DEFAULT_INTERNAL_SAMPLE_RATE 8000
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
struct ast_format *format;
@ -117,7 +119,7 @@ int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type
audiohook->init_flags = init_flags;
/* initialize internal rate at 8khz, this will adjust if necessary */
audiohook_set_internal_rate(audiohook, 8000, 0);
audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 0);
/* Since we are just starting out... this audiohook is new */
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
@ -361,7 +363,19 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio
struct ast_frame *read_frame = NULL, *final_frame = NULL;
struct ast_format *slin;
audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
/*
* Update the rate if compatibility mode is turned off or if it is
* turned on and the format rate is higher than the current rate.
*
* This makes it so any unnecessary rate switching/resetting does
* not take place and also any associated audiohook_list's internal
* sample rate maintains the highest sample rate between hooks.
*/
if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) ||
(ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) &&
ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) {
audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
}
if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
@ -425,6 +439,22 @@ struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook,
static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
{
struct ast_audiohook *ah = NULL;
/*
* Anytime the samplerate compatibility is set (attach/remove an audiohook) the
* list's internal sample rate needs to be reset so that the next time processing
* through write_list, if needed, it will get updated to the correct rate.
*
* A list's internal rate always chooses the higher between its own rate and a
* given rate. If the current rate is being driven by an audiohook that wanted a
* higher rate then when this audiohook is removed the list's rate would remain
* at that level when it should be lower, and with no way to lower it since any
* rate compared against it would be lower.
*
* By setting it back to the lowest rate it can recalulate the new highest rate.
*/
audiohook_list->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
audiohook_list->native_slin_compatible = 1;
AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
@ -455,7 +485,7 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list);
AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list);
/* This sample rate will adjust as necessary when writing to the list. */
ast_channel_audiohooks(chan)->list_internal_samp_rate = 8000;
ast_channel_audiohooks(chan)->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
}
/* Drop into respective list */
@ -467,8 +497,11 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
}
audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
/*
* Initialize the audiohook's rate to the default. If it needs to be,
* it will get updated later.
*/
audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 1);
audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
/* Change status over to running since it is now attached */
@ -766,14 +799,14 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
struct ast_frame *new_frame = frame;
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
* updating the list's rate, all the audiohooks in the list will be updated as well
* as the are written and read from. */
if (audiohook_list->native_slin_compatible) {
audiohook_list->list_internal_samp_rate =
MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
}
/*
* If we are capable of sample rates other that 8khz, update the internal
* audiohook_list's rate and higher sample rate audio arrives. If native
* slin compatibility is turned on all audiohooks in the list will be
* updated as well during read/write processing.
*/
audiohook_list->list_internal_samp_rate =
MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
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) {
@ -821,6 +854,36 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook
return outframe;
}
/*!
*\brief Set the audiohook's internal sample rate to the audiohook_list's rate,
* but only when native slin compatibility is turned on.
*
* \param audiohook_list audiohook_list data object
* \param audiohook the audiohook to update
* \param rate the current max internal sample rate
*/
static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list,
struct ast_audiohook *audiohook, int *rate)
{
/* The rate should always be the max between itself and the hook */
if (audiohook->hook_internal_samp_rate > *rate) {
*rate = audiohook->hook_internal_samp_rate;
}
/*
* If native slin compatibility is turned on then update the audiohook
* with the audiohook_list's current rate. Note, the audiohook's rate is
* set to the audiohook_list's rate and not the given rate. If there is
* a change in rate the hook's rate is changed on its next check.
*/
if (audiohook_list->native_slin_compatible) {
ast_set_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
} else {
ast_clear_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
}
}
/*!
* \brief Pass an AUDIO frame off to be handled by the audiohook core
*
@ -851,6 +914,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
int samples;
int middle_frame_manipulated = 0;
int removed = 0;
int internal_sample_rate;
/* ---Part_1. translate start_frame to SLINEAR if necessary. */
if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
@ -858,6 +922,19 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
samples = middle_frame->samples;
/*
* While processing each audiohook check to see if the internal sample rate needs
* to be adjusted (it should be the highest rate specified between formats and
* hooks). The given audiohook_list's internal sample rate is then set to the
* updated value before returning.
*
* If slin compatibility mode is turned on then an audiohook's internal sample
* rate is set to its audiohook_list's rate. If an audiohook_list's rate is
* adjusted during this pass then the change is picked up by the audiohooks
* on the next pass.
*/
internal_sample_rate = audiohook_list->list_internal_samp_rate;
/* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
/* Queue up signed linear frame to each spy */
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
@ -872,7 +949,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
continue;
}
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
ast_audiohook_write_frame(audiohook, direction, middle_frame);
ast_audiohook_unlock(audiohook);
}
@ -896,7 +973,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
continue;
}
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
/* Take audio from this whisper source and combine it into our main buffer */
for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
@ -929,14 +1006,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
continue;
}
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
/* Feed in frame to manipulation. */
if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
/* If the manipulation fails then the frame will be returned in its original state.
* Since there are potentially more manipulator callbacks in the list, no action should
* be taken here to exit early. */
middle_frame_manipulated = 1;
}
audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
/*
* Feed in frame to manipulation.
*
* XXX FAILURES ARE IGNORED XXX
* If the manipulation fails then the frame will be returned in its original state.
* Since there are potentially more manipulator callbacks in the list, no action should
* be taken here to exit early.
*/
audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
@ -960,6 +1039,12 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
/* Before returning, if an audiohook got removed, reset samplerate compatibility */
if (removed) {
audiohook_list_set_samplerate_compatibility(audiohook_list);
} else {
/*
* Set the audiohook_list's rate to the updated rate. Note that if a hook
* was removed then the list's internal rate is reset to the default.
*/
audiohook_list->list_internal_samp_rate = internal_sample_rate;
}
return end_frame;

Loading…
Cancel
Save