codecs/gtk: fix int to size_t

Change-Id: I8f467f09375c8227c4b70aef47ff3a590a0c00d7
Reviewed-on: https://code.wireshark.org/review/10413
Petri-Dish: Anders Broman <a.broman58@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Guy Harris <guy@alum.mit.edu>

1 files changed, 18 insertions, 14 deletions

diff --git a/ui/gtk/rtp_player.c b/ui/gtk/rtp_player.c
index 9dc9767846..50bbf70fbe 100644
--- a/ui/gtk/rtp_player.c
+++ b/ui/gtk/rtp_player.c
@@ -55,6 +55,7 @@
 #include <math.h>
 #include <string.h>
 #include <portaudio.h>
+#include <assert.h>
 
 #include <gtk/gtk.h>
 
@@ -119,11 +120,11 @@ static int new_jitter_buff;
 /* a hash table with the RTP streams to play per audio channel */
 static GHashTable *rtp_channels_hash = NULL;
 
-static int sample_rate = 8000;
-static int channels = 1;
+static unsigned sample_rate = 8000;
+static unsigned channels = 1;
 
 /* Port Audio stuff */
-static int output_channels = 2;
+static unsigned output_channels = 2;
 
 #define PA_SAMPLE_TYPE  paInt16
 typedef gint16 SAMPLE;
@@ -456,15 +457,15 @@ mark_all_rtp_stream_to_decode(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer
 /* Decode a RTP packet
  * Return the number of decoded bytes
  */
-static int
+static size_t
 decode_rtp_packet(rtp_packet_t *rp, SAMPLE **out_buff, GHashTable *decoders_hash)
 {
 	unsigned int  payload_type;
 	const gchar *p;
 	rtp_decoder_t *decoder;
 	SAMPLE *tmp_buff = NULL;
-	int tmp_buff_len;
-	int decoded_bytes = 0;
+	size_t tmp_buff_len;
+	size_t decoded_bytes = 0;
 
 	if ((rp->payload_data == NULL) || (rp->info->info_payload_len == 0) ) {
 		return 0;
@@ -533,7 +534,8 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 	GList*  rtp_packet_list;
 	rtp_packet_t *rp;
 
-	int i;
+	size_t i;
+	gint32 j;
 	double rtp_time;
 	double rtp_time_prev;
 	double arrive_time;
@@ -551,8 +553,8 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 #endif
 	char *src_addr, *dst_addr;
 
-	int decoded_bytes;
-	int decoded_bytes_prev;
+	size_t decoded_bytes;
+	size_t decoded_bytes_prev;
 	int jitter_buff;
 	SAMPLE *out_buff = NULL;
 	sample_t silence;
@@ -611,7 +613,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 	} else {
 		/* Add silence between the two streams if needed */
 		silence_frames = (gint32)((nstime_to_msec(&rsi->start_fd->abs_ts) - nstime_to_msec(&rci->stop_time_abs)) * sample_rate);
-		for (i = 0; i< silence_frames; i++) {
+		for (j = 0; j < silence_frames; j++) {
 			g_array_append_val(rci->samples, silence);
 		}
 		rci->num_packets += rsi->packet_count;
@@ -720,7 +722,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 				if (silence_frames > MAX_SILENCE_FRAMES)
 					silence_frames = MAX_SILENCE_FRAMES;
 
-				for (i = 0; i< silence_frames; i++) {
+				for (j = 0; j < silence_frames; j++) {
 					silence.status = status;
 					g_array_append_val(rci->samples, silence);
 
@@ -756,7 +758,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 			if (silence_frames > MAX_SILENCE_FRAMES)
 				silence_frames = MAX_SILENCE_FRAMES;
 
-			for (i = 0; i< silence_frames; i++) {
+			for (j = 0; j < silence_frames; j++) {
 				silence.status = status;
 				g_array_append_val(rci->samples, silence);
 
@@ -1888,6 +1890,7 @@ play_channels(void)
 	/* if not PAUSE, then start to PLAY */
 	} else {
 #if PORTAUDIO_API_1
+		assert(output_channels <= INT_MAX);
 		err = Pa_OpenStream(
 			  &pa_stream,
 			  paNoDevice,     /* default input device */
@@ -1895,7 +1898,7 @@ play_channels(void)
 			  PA_SAMPLE_TYPE,
 			  NULL,
 			  Pa_GetDefaultOutputDeviceID(),
-			  output_channels,
+			  (int)output_channels,
 			  PA_SAMPLE_TYPE,
 			  NULL,
 			  sample_rate,
@@ -1935,10 +1938,11 @@ play_channels(void)
 		}
 #else /* PORTAUDIO_API_1 */
 		if (Pa_GetDefaultOutputDevice() != paNoDevice) {
+		        assert(output_channels <= INT_MAX);
 			err = Pa_OpenDefaultStream(
 				&pa_stream,
 				0,
-				output_channels,
+				(int)output_channels,
 				PA_SAMPLE_TYPE,
 				sample_rate,
 				FRAMES_PER_BUFFER,