fix sample rate logic again, to be robust to filters not implementing MS_FILTER_GET_SAMPLE_RATE

include/mediastreamer2/msfilter.h

@@ -435,6 +435,15 @@ MS2_PUBLIC int ms_filter_call_method_noarg(MSFilter *f, unsigned int id);
  */
 MS2_PUBLIC bool_t ms_filter_has_method(MSFilter *f, unsigned int id);
 
+/**
+ * Returns whether a filter implements a given interface.
+ * @param f a MSFilter object
+ * @param id an interface id.
+ * 
+ * Returns TRUE if interface is implemented, FALSE, otherwise.
+**/
+bool_t ms_filter_implements_interface(MSFilter *f, MSFilterInterfaceId id);
+
 /**
  * Set a callback on filter's to be informed of private filter's event.
  * This callback is called from the filter's MSTicker, unless a global event queue

src/base/msfilter.c

@@ -211,6 +211,10 @@ bool_t ms_filter_desc_implements_interface(MSFilterDesc *desc, MSFilterInterface
 	return FALSE;
 }
 
+bool_t ms_filter_implements_interface(MSFilter *f, MSFilterInterfaceId id){
+	return ms_filter_desc_implements_interface(f->desc,id);
+}
+
 MSList *ms_filter_lookup_by_interface(MSFilterInterfaceId id){
 	MSList *ret=NULL;
 	MSList *elem;

src/voip/audioconference.c

@@ -105,7 +105,9 @@ static void cut_audio_stream_graph(MSAudioEndpoint *ep, bool_t is_remote){
 
 	if (ms_filter_has_method(st->ms.encoder,MS_FILTER_GET_SAMPLE_RATE)){
 		ms_filter_call_method(st->ms.encoder,MS_FILTER_GET_SAMPLE_RATE,&ep->samplerate);
-	}else ms_warning("MSAudioConference: filter %s does not implement MS_FILTER_GET_SAMPLE_RATE, assuming 8khz",st->ms.encoder->desc->name);
+	}else{
+		ms_filter_call_method(st->ms.rtpsend,MS_FILTER_GET_SAMPLE_RATE,&ep->samplerate);
+	}
 
 	if (is_remote){
 		ep->mixer_in.filter=ep->in_cut_point_prev.filter;

src/voip/audiostream.c

@@ -86,7 +86,11 @@ static void on_dtmf_received(RtpSession *s, int dtmf, void * user_data)
 	}
 }
 
-static void audio_stream_configure_resampler(MSFilter *resampler,MSFilter *from,MSFilter *to) {
+/*
+ * note: since not all filters implement MS_FILTER_GET_SAMPLE_RATE, fallback_from_rate and fallback_to_rate are expected to provide sample rates
+ * obtained by another context, such as the RTP clock rate for example.
+ */
+static void audio_stream_configure_resampler(MSFilter *resampler,MSFilter *from, MSFilter *to, int fallback_from_rate, int fallback_to_rate) {
 	int from_rate=0, to_rate=0;
 	int from_channels = 0, to_channels = 0;
 	ms_filter_call_method(from,MS_FILTER_GET_SAMPLE_RATE,&from_rate);
@@ -102,12 +106,12 @@ static void audio_stream_configure_resampler(MSFilter *resampler,MSFilter *from,
 		ms_error("Filter %s does not implement the MS_FILTER_GET_NCHANNELS method", to->desc->name);
 	}
 	if (from_rate == 0){
-		ms_error("Filter %s does not implement the MS_FILTER_GET_SAMPLE_RATE method, assuming 8000hz", from->desc->name);
-		from_rate=8000;
+		ms_error("Filter %s does not implement the MS_FILTER_GET_SAMPLE_RATE method", from->desc->name);
+		from_rate=fallback_from_rate;
 	}
 	if (to_rate == 0){
-		ms_error("Filter %s does not implement the MS_FILTER_GET_SAMPLE_RATE method, assuming 8000hz", to->desc->name);
-		to_rate=8000;
+		ms_error("Filter %s does not implement the MS_FILTER_GET_SAMPLE_RATE method", to->desc->name);
+		to_rate=fallback_to_rate;
 	}
 	ms_filter_call_method(resampler,MS_FILTER_SET_SAMPLE_RATE,&from_rate);
 	ms_filter_call_method(resampler,MS_FILTER_SET_OUTPUT_SAMPLE_RATE,&to_rate);
@@ -360,14 +364,12 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	}else {
 		stream->soundread=ms_filter_new(MS_FILE_PLAYER_ID);
 		stream->read_resampler=ms_filter_new(MS_RESAMPLE_ID);
-		if (infile!=NULL) audio_stream_play(stream,infile);
 	}
 	if (playcard!=NULL) {
 		if (stream->soundwrite==NULL)
 			stream->soundwrite=ms_snd_card_create_writer(playcard);
 	}else {
 		stream->soundwrite=ms_filter_new(MS_FILE_REC_ID);
-		if (outfile!=NULL) audio_stream_record(stream,outfile);
 	}
 
 	/* creates the couple of encoder/decoder */
@@ -444,6 +446,13 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		stream->volrecv=NULL;
 	audio_stream_enable_echo_limiter(stream,stream->el_type);
 	audio_stream_enable_noise_gate(stream,stream->use_ng);
+	
+	if (ms_filter_implements_interface(stream->soundread,MSFilterPlayerInterface) && infile){
+		audio_stream_play(stream,infile);
+	}
+	if (ms_filter_implements_interface(stream->soundwrite,MSFilterPlayerInterface) && outfile){
+		audio_stream_record(stream,outfile);
+	}
 
 	if (stream->use_agc){
 		int tmp=1;
@@ -542,11 +551,11 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 	
 	/*configure resamplers if needed*/
 	if (stream->read_resampler){
-		audio_stream_configure_resampler(stream->read_resampler,stream->soundread,stream->ms.encoder);
+		audio_stream_configure_resampler(stream->read_resampler,stream->soundread,stream->ms.encoder,8000,pt->clock_rate);
 	}
 
 	if (stream->write_resampler){
-		audio_stream_configure_resampler(stream->write_resampler,stream->ms.decoder,stream->soundwrite);
+		audio_stream_configure_resampler(stream->write_resampler,stream->ms.decoder,stream->soundwrite,pt->clock_rate,8000);
 	}
 	
 	if (stream->ms.use_rc){
@@ -698,7 +707,9 @@ void audio_stream_play(AudioStream *st, const char *name){
 		if (name != NULL) {
 			ms_filter_call_method(st->soundread,MS_FILE_PLAYER_OPEN,(void*)name);
 			if (st->read_resampler){
-				audio_stream_configure_resampler(st->read_resampler,st->soundread,st->ms.rtpsend);
+				int fallback_to_rate=8000;
+				ms_filter_call_method(st->ms.rtpsend,MS_FILTER_GET_SAMPLE_RATE,&fallback_to_rate);
+				audio_stream_configure_resampler(st->read_resampler,st->soundread,st->ms.encoder, 8000, fallback_to_rate);
 			}
 			ms_filter_call_method_noarg(st->soundread,MS_FILE_PLAYER_START);
 		}