[Asterisk-code-review] func_volume: Accept decimal number as argument (asterisk[17])

[Friendly Automation]

Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/14174 )

Change subject: func_volume: Accept decimal number as argument
......................................................................

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
---
A doc/CHANGES-staging/func_volume.txt
M funcs/func_volume.c
M include/asterisk/frame.h
M include/asterisk/utils.h
M main/frame.c
5 files changed, 68 insertions(+), 6 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/doc/CHANGES-staging/func_volume.txt b/doc/CHANGES-staging/func_volume.txt
new file mode 100644
index 0000000..e73295b
--- /dev/null
+++ b/doc/CHANGES-staging/func_volume.txt
@@ -0,0 +1,3 @@
+Subject: func_volume
+
+Accept decimal number as argument.
diff --git a/funcs/func_volume.c b/funcs/func_volume.c
index 533eb55..8587479 100644
--- a/funcs/func_volume.c
+++ b/funcs/func_volume.c
@@ -70,8 +70,8 @@
 
 struct volume_information {
 	struct ast_audiohook audiohook;
-	int tx_gain;
-	int rx_gain;
+	float tx_gain;
+	float rx_gain;
 	unsigned int flags;
 };
 
@@ -107,7 +107,7 @@
 {
 	struct ast_datastore *datastore = NULL;
 	struct volume_information *vi = NULL;
-	int *gain = NULL;
+	float *gain = NULL;
 
 	/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
 	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
@@ -141,7 +141,7 @@
 		if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
 			return 0;
 		/* Apply gain to frame... easy as pi */
-		ast_frame_adjust_volume(frame, *gain);
+		ast_frame_adjust_volume_float(frame, *gain);
 	}
 
 	return 0;
@@ -193,9 +193,9 @@
 	}
 
 	if (!strcasecmp(args.direction, "tx")) {
-		vi->tx_gain = atoi(value);
+		vi->tx_gain = atof(value);
 	} else if (!strcasecmp(args.direction, "rx")) {
-		vi->rx_gain = atoi(value);
+		vi->rx_gain = atof(value);
 	} else {
 		ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
 	}
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index b618d19..5e3a432 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -595,6 +595,14 @@
 int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
 
 /*!
+  \brief Adjusts the volume of the audio samples contained in a frame.
+  \param f The frame containing the samples (must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR)
+  \param adjustment The number of dB to adjust up or down.
+  \return 0 for success, non-zero for an error
+ */
+int ast_frame_adjust_volume_float(struct ast_frame *f, float adjustment);
+
+/*!
   \brief Sums two frames of audio samples.
   \param f1 The first frame (which will contain the result)
   \param f2 The second frame
diff --git a/include/asterisk/utils.h b/include/asterisk/utils.h
index 2b332a5..10dfe83 100644
--- a/include/asterisk/utils.h
+++ b/include/asterisk/utils.h
@@ -375,11 +375,35 @@
 		*input = (short) res;
 }
 
+static force_inline void ast_slinear_saturated_multiply_float(short *input, float *value)
+{
+	float res;
+
+	res = (float) *input * *value;
+	if (res > 32767)
+		*input = 32767;
+	else if (res < -32768)
+		*input = -32768;
+	else
+		*input = (short) res;
+}
+
 static force_inline void ast_slinear_saturated_divide(short *input, short *value)
 {
 	*input /= *value;
 }
 
+static force_inline void ast_slinear_saturated_divide_float(short *input, float *value)
+{
+	float res = (float) *input / *value;
+	if (res > 32767)
+		*input = 32767;
+	else if (res < -32768)
+		*input = -32768;
+	else
+		*input = (short) res;
+}
+
 #ifdef localtime_r
 #undef localtime_r
 #endif
diff --git a/main/frame.c b/main/frame.c
index 208c82d..b2c5040 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -43,6 +43,8 @@
 #include "asterisk/dsp.h"
 #include "asterisk/file.h"
 
+#include <math.h>
+
 #if !defined(LOW_MEMORY)
 static void frame_cache_cleanup(void *data);
 
@@ -703,6 +705,31 @@
 	return 0;
 }
 
+int ast_frame_adjust_volume_float(struct ast_frame *f, float adjustment)
+{
+	int count;
+	short *fdata = f->data.ptr;
+	float adjust_value = fabs(adjustment);
+
+	if ((f->frametype != AST_FRAME_VOICE) || !(ast_format_cache_is_slinear(f->subclass.format))) {
+		return -1;
+	}
+
+	if (!adjustment) {
+		return 0;
+	}
+
+	for (count = 0; count < f->samples; count++) {
+		if (adjustment > 0) {
+			ast_slinear_saturated_multiply_float(&fdata[count], &adjust_value);
+		} else if (adjustment < 0) {
+			ast_slinear_saturated_divide_float(&fdata[count], &adjust_value);
+		}
+	}
+
+	return 0;
+}
+
 int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2)
 {
 	int count;

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Gerrit-Project: asterisk
Gerrit-Branch: 17
Gerrit-Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c
Gerrit-Change-Number: 14174
Gerrit-PatchSet: 3
Gerrit-Owner: Jean Aunis - Prescom <jean.aunis at prescom.fr>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: merged
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