implement audio conference recording.

include/mediastreamer2/msconference.h

@@ -172,6 +172,35 @@ MS2_PUBLIC MSAudioEndpoint * ms_audio_endpoint_get_from_stream(AudioStream *st,
 **/
 MS2_PUBLIC void ms_audio_endpoint_release_from_stream(MSAudioEndpoint *obj);
 
+/**
+ * Creates an audio endpoint (or virtual participant) to record the conference into a wav file.
+ * @param path Path to the wav file to record.
+**/
+MS2_PUBLIC MSAudioEndpoint * ms_audio_endpoint_new_recorder(void);
+
+/**
+ * Start audio recording.
+ * The endpoint must have been created by ms_audio_endpoint_new_recorder().
+ * @param ep the endpoint
+ * @param path path for the wav file where to record samples.
+ * @return 0 if successful, -1 if the path is invalid.
+**/
+MS2_PUBLIC int ms_audio_recorder_endpoint_start(MSAudioEndpoint *ep, const char *path);
+
+/**
+ * Stop audio recording.
+ * The endpoint must have been created by ms_audio_endpoint_new_recorder().
+ * @param ep the endpoint
+ * @return 0 if successful, -1 if the record wasn't started.
+**/
+MS2_PUBLIC int ms_audio_recorder_endpoint_stop(MSAudioEndpoint *ep);
+
+/**
+ * Destroy an audio endpoint.
+ * @note Endpoints created by ms_audio_endpoint_get_from_stream() must be released by ms_audio_endpoint_release_from_stream().
+**/
+void ms_audio_endpoint_destroy(MSAudioEndpoint *ep);
+
 #ifdef __cplusplus
 }
 #endif

src/voip/audioconference.c

@@ -38,11 +38,13 @@ struct _MSAudioEndpoint{
 	MSCPoint mixer_in;
 	MSCPoint mixer_out;
 	MSAudioConference *conference;
+	MSFilter *recorder; /* in case it is a recorder endpoint*/
+	MSFilter *player; /* not used at the moment, but we need it so that there is a source connected to the mixer*/
 	int pin;
 	int samplerate;
 };
 
-void ms_audio_endpoint_destroy(MSAudioEndpoint *ep);
+
 
 MSAudioConference * ms_audio_conference_new(const MSAudioConferenceParams *params){
 	MSAudioConference *obj=ms_new0(MSAudioConference,1);
@@ -138,12 +140,25 @@ static int find_free_pin(MSFilter *mixer){
 static void plumb_to_conf(MSAudioEndpoint *ep){
 	MSAudioConference *conf=ep->conference;
 	int in_rate=ep->samplerate,out_rate=ep->samplerate;
+	
+	if (ep->samplerate!=-1){
+		out_rate=in_rate=ep->samplerate;
+	}else in_rate=out_rate=conf->params.samplerate;
+	
+	if (ep->recorder){
+		ms_filter_call_method(ep->recorder,MS_FILTER_SET_SAMPLE_RATE,&conf->params.samplerate);
+	}
+	
 	ep->pin=find_free_pin(conf->mixer);
 	
-	ms_filter_link(ep->mixer_in.filter,ep->mixer_in.pin,ep->in_resampler,0);
-	ms_filter_link(ep->in_resampler,0,conf->mixer,ep->pin);
-	ms_filter_link(conf->mixer,ep->pin,ep->out_resampler,0);
-	ms_filter_link(ep->out_resampler,0,ep->mixer_out.filter,ep->mixer_out.pin);
+	if (ep->mixer_in.filter){
+		ms_filter_link(ep->mixer_in.filter,ep->mixer_in.pin,ep->in_resampler,0);
+		ms_filter_link(ep->in_resampler,0,conf->mixer,ep->pin);
+	}
+	if (ep->mixer_out.filter){
+		ms_filter_link(conf->mixer,ep->pin,ep->out_resampler,0);
+		ms_filter_link(ep->out_resampler,0,ep->mixer_out.filter,ep->mixer_out.pin);
+	}
 
 	/*configure resamplers*/
 	ms_filter_call_method(ep->in_resampler,MS_FILTER_SET_OUTPUT_SAMPLE_RATE,&conf->params.samplerate);
@@ -165,10 +180,14 @@ void ms_audio_conference_add_member(MSAudioConference *obj, MSAudioEndpoint *ep)
 static void unplumb_from_conf(MSAudioEndpoint *ep){
 	MSAudioConference *conf=ep->conference;
 	
-	ms_filter_unlink(ep->mixer_in.filter,ep->mixer_in.pin,ep->in_resampler,0);
-	ms_filter_unlink(ep->in_resampler,0,conf->mixer,ep->pin);
-	ms_filter_unlink(conf->mixer,ep->pin,ep->out_resampler,0);
-	ms_filter_unlink(ep->out_resampler,0,ep->mixer_out.filter,ep->mixer_out.pin);
+	if (ep->mixer_in.filter){
+		ms_filter_unlink(ep->mixer_in.filter,ep->mixer_in.pin,ep->in_resampler,0);
+		ms_filter_unlink(ep->in_resampler,0,conf->mixer,ep->pin);
+	}
+	if (ep->mixer_out.filter){
+		ms_filter_unlink(conf->mixer,ep->pin,ep->out_resampler,0);
+		ms_filter_unlink(ep->out_resampler,0,ep->mixer_out.filter,ep->mixer_out.pin);
+	}
 }
 
 void ms_audio_conference_remove_member(MSAudioConference *obj, MSAudioEndpoint *ep){
@@ -186,6 +205,11 @@ void ms_audio_conference_mute_member(MSAudioConference *obj, MSAudioEndpoint *ep
 	ms_filter_call_method(ep->conference->mixer, MS_AUDIO_MIXER_SET_ACTIVE, &ctl);
 }
 
+int ms_audio_conference_get_size(MSAudioConference *obj){
+	return obj->nmembers;
+}
+
+
 void ms_audio_conference_destroy(MSAudioConference *obj){
 	ms_ticker_destroy(obj->ticker);
 	ms_filter_destroy(obj->mixer);
@@ -215,9 +239,40 @@ void ms_audio_endpoint_release_from_stream(MSAudioEndpoint *obj){
 void ms_audio_endpoint_destroy(MSAudioEndpoint *ep){
 	if (ep->in_resampler) ms_filter_destroy(ep->in_resampler);
 	if (ep->out_resampler) ms_filter_destroy(ep->out_resampler);
+	if (ep->recorder) ms_filter_destroy(ep->recorder);
+	if (ep->player) ms_filter_destroy(ep->player);
 	ms_free(ep);
 }
 
-int ms_audio_conference_get_size(MSAudioConference *obj){
-	return obj->nmembers;
+MSAudioEndpoint * ms_audio_endpoint_new_recorder(){
+	MSAudioEndpoint *ep=ms_audio_endpoint_new();
+	ep->recorder=ms_filter_new(MS_FILE_REC_ID);
+	ep->player=ms_filter_new(MS_FILE_PLAYER_ID);
+	ep->mixer_out.filter=ep->recorder;
+	ep->mixer_in.filter=ep->player;
+	ep->samplerate=-1;
+	return ep;
+}
+
+int ms_audio_recorder_endpoint_start(MSAudioEndpoint *ep, const char *path){
+	int err;
+	MSRecorderState state;
+	if (!ep->recorder){
+		ms_error("This endpoint isn't a recorder endpoint.");
+		return -1;
+	}
+	ms_filter_call_method(ep->recorder,MS_RECORDER_GET_STATE,&state);
+	if (state!=MSRecorderClosed)
+		ms_filter_call_method_noarg(ep->recorder,MS_RECORDER_CLOSE);
+	err=ms_filter_call_method(ep->recorder,MS_RECORDER_OPEN,(void*)path);
+	if (err==-1) return -1;
+	return ms_filter_call_method_noarg(ep->recorder,MS_RECORDER_START);
+}
+
+int ms_audio_recorder_endpoint_stop(MSAudioEndpoint *ep){
+	if (!ep->recorder){
+		return -1;
+	}
+	return ms_filter_call_method_noarg(ep->recorder,MS_RECORDER_CLOSE);
 }
+