obs-ffmpeg: Fix potential integer overflows

This particularly affected audio encoding, audio encoding previously
would count samples and use it to create an encoding timestamp, but
because I was using a standard integer (which is 32bit by default on
x86), it would max out at about 0x7FFFFFFF samples, which is about 12
hours of samples at 48000 sample rate.  After that, it would start going
into negative territory (overflowing).  By changing it to int64_t, it
will make it so that audio at 48000 samples per second would only be
able to overflow after about..  6.09 million years.  In other words,
this should fix the issue for good.

plugins/obs-ffmpeg/obs-ffmpeg-aac.c

@@ -33,7 +33,7 @@ struct aac_encoder {
 
 	uint8_t          *samples[MAX_AV_PLANES];
 	AVFrame          *aframe;
-	int              total_samples;
+	int64_t          total_samples;
 
 	DARRAY(uint8_t)  packet_buffer;
 

plugins/obs-ffmpeg/obs-ffmpeg-output.c

@@ -57,13 +57,14 @@ struct ffmpeg_data {
 	AVFormatContext    *output;
 	struct SwsContext  *swscale;
 
+	int64_t            total_frames;
 	AVPicture          dst_picture;
 	AVFrame            *vframe;
 	int                frame_size;
-	int                total_frames;
 
 	uint64_t           start_timestamp;
 
+	int64_t            total_samples;
 	uint32_t           audio_samplerate;
 	enum audio_format  audio_format;
 	size_t             audio_planes;
@@ -71,7 +72,6 @@ struct ffmpeg_data {
 	struct circlebuf   excess_frames[MAX_AV_PLANES];
 	uint8_t            *samples[MAX_AV_PLANES];
 	AVFrame            *aframe;
-	int                total_samples;
 
 	struct ffmpeg_cfg  config;