func_volume: Accept decimal number as argument
Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c
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@ -0,0 +1,3 @@
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Subject: func_volume
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Accept decimal number as argument.
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@ -70,8 +70,8 @@
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struct volume_information {
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struct ast_audiohook audiohook;
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int tx_gain;
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int rx_gain;
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float tx_gain;
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float rx_gain;
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unsigned int flags;
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};
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@ -107,7 +107,7 @@ static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *
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{
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struct ast_datastore *datastore = NULL;
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struct volume_information *vi = NULL;
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int *gain = NULL;
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float *gain = NULL;
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/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
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@ -141,7 +141,7 @@ static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *
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if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
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return 0;
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/* Apply gain to frame... easy as pi */
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ast_frame_adjust_volume(frame, *gain);
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ast_frame_adjust_volume_float(frame, *gain);
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}
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return 0;
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@ -193,9 +193,9 @@ static int volume_write(struct ast_channel *chan, const char *cmd, char *data, c
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}
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if (!strcasecmp(args.direction, "tx")) {
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vi->tx_gain = atoi(value);
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vi->tx_gain = atof(value);
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} else if (!strcasecmp(args.direction, "rx")) {
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vi->rx_gain = atoi(value);
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vi->rx_gain = atof(value);
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} else {
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ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
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}
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@ -594,6 +594,14 @@ struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f,
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*/
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int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
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/*!
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\brief Adjusts the volume of the audio samples contained in a frame.
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\param f The frame containing the samples (must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR)
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\param adjustment The number of dB to adjust up or down.
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\return 0 for success, non-zero for an error
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*/
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int ast_frame_adjust_volume_float(struct ast_frame *f, float adjustment);
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/*!
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\brief Sums two frames of audio samples.
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\param f1 The first frame (which will contain the result)
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@ -375,11 +375,35 @@ static force_inline void ast_slinear_saturated_multiply(short *input, short *val
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*input = (short) res;
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}
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static force_inline void ast_slinear_saturated_multiply_float(short *input, float *value)
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{
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float res;
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res = (float) *input * *value;
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if (res > 32767)
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*input = 32767;
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else if (res < -32768)
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*input = -32768;
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else
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*input = (short) res;
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}
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static force_inline void ast_slinear_saturated_divide(short *input, short *value)
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{
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*input /= *value;
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}
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static force_inline void ast_slinear_saturated_divide_float(short *input, float *value)
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{
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float res = (float) *input / *value;
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if (res > 32767)
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*input = 32767;
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else if (res < -32768)
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*input = -32768;
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else
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*input = (short) res;
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}
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#ifdef localtime_r
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#undef localtime_r
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#endif
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27
main/frame.c
27
main/frame.c
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@ -43,6 +43,8 @@
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#include "asterisk/dsp.h"
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#include "asterisk/file.h"
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#include <math.h>
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#if !defined(LOW_MEMORY)
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static void frame_cache_cleanup(void *data);
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@ -703,6 +705,31 @@ int ast_frame_adjust_volume(struct ast_frame *f, int adjustment)
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return 0;
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}
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int ast_frame_adjust_volume_float(struct ast_frame *f, float adjustment)
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{
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int count;
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short *fdata = f->data.ptr;
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float adjust_value = fabs(adjustment);
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if ((f->frametype != AST_FRAME_VOICE) || !(ast_format_cache_is_slinear(f->subclass.format))) {
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return -1;
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}
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if (!adjustment) {
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return 0;
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}
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for (count = 0; count < f->samples; count++) {
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if (adjustment > 0) {
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ast_slinear_saturated_multiply_float(&fdata[count], &adjust_value);
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} else if (adjustment < 0) {
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ast_slinear_saturated_divide_float(&fdata[count], &adjust_value);
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}
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}
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return 0;
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}
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int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2)
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{
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int count;
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