obs-outputs: Fix writing MP4/MOV audio with sample rates > 65535

plugins/obs-outputs/mp4-mux-internal.h

@@ -91,7 +91,7 @@ struct mp4_track {
 	/* Time Base (1/FPS for video, 1/sample rate for audio) */
 	uint32_t timebase_num;
 	uint32_t timebase_den;
-	/* Output timescale calculated from time base (Video only) */
+	/* Output timescale calculated from time base */
 	uint32_t timescale;
 
 	/* First PTS this track has seen (in track timescale) */

plugins/obs-outputs/mp4-mux.c

@@ -906,6 +906,15 @@ static inline void mp4_write_audio_sample_entry(struct mp4_mux *mux, struct mp4_
 		s_wb16(s, is_mov && !is_pcm ? -2 : 0); // pre_defined (compression ID in MOV)
 		s_wb16(s, 0);                          // reserved
 
+		/* The sample rate field is limited to 16-bits. Technically version 1 supports a "srat" box wich
+		 * provides 32 bits, but this is not supported by most software (including FFmpeg and Chromium).
+		 * For encoded codecs (AAC etc.), the sample rate can be read from the encoded data itself.
+		 * For PCM FFmpeg will try to use the timescale as sample rate. */
+		if (sample_rate > UINT16_MAX) {
+			warn("Sample rate too high for MP4, file may not play back correctly.");
+			sample_rate = 0;
+		}
+
 		s_wb32(s, sample_rate << 16); // samplerate
 
 		/* MOV-only data: https://developer.apple.com/documentation/quicktime-file-format/sound_sample_description_version_1 */
@@ -1226,8 +1235,10 @@ static size_t mp4_write_mov_audio_tag(struct mp4_mux *mux, struct mp4_track *tra
 
 	const char *tag = NULL;
 	audio_t *audio = obs_encoder_audio(track->encoder);
+	uint32_t sample_rate = audio_output_get_sample_rate(audio);
+	size_t channels = audio_output_get_channels(audio);
 	// More than 2 channels requires v2
-	uint8_t version = audio_output_get_channels(audio) > 2 ? 2 : 1;
+	uint8_t version = (channels > 2 || sample_rate > UINT16_MAX) ? 2 : 1;
 
 	if (track->codec == CODEC_PCM_F32 || track->codec == CODEC_PCM_I16 || track->codec == CODEC_PCM_I24) {
 		tag = "lpcm";