Audiostream stats

coreapi/linphonecall.c

@@ -2230,6 +2230,27 @@ static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
 	}
 }
 
+// this method should be put in mediastreamer but has been kept here, private,
+// to save time.
+void linphone_call_stream_stats_hack(MediaStream* stream, LinphoneCallStats *stats, OrtpEvent *ev) {
+	OrtpEventType evt=ortp_event_get_type(ev);
+	OrtpEventData *evd=ortp_event_get_data(ev);
+	if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
+		stats->round_trip_delay = rtp_session_get_round_trip_propagation(stream->session);
+		if(stats->received_rtcp != NULL) freemsg(stats->received_rtcp);
+		stats->received_rtcp = evd->packet;
+		evd->packet = NULL;
+		update_local_stats(stats,stream);
+
+	} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
+		memcpy(&stats->jitter_stats, rtp_session_get_jitter_stats(stream->session), sizeof(jitter_stats_t));
+		if(stats->sent_rtcp != NULL) freemsg(stats->sent_rtcp);
+		stats->sent_rtcp = evd->packet;
+		evd->packet = NULL;
+		update_local_stats(stats,stream);
+	}
+}
+
 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
 	LinphoneCore* lc = call->core;
 	int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
@@ -2259,6 +2280,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 #ifdef VIDEO_ENABLED
 	if (call->videostream!=NULL) {
 		OrtpEvent *ev;
+		LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_VIDEO];
 
 		/* Ensure there is no dangling ICE check list. */
 		if (call->ice_session == NULL) call->videostream->ms.ice_check_list = NULL;
@@ -2272,24 +2294,10 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 			OrtpEventData *evd=ortp_event_get_data(ev);
 			if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
 				linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
-			} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
-				call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->ms.session);
-				if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
-					freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
-				call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
-				evd->packet = NULL;
-				update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
+			} else if  (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED || evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
+				linphone_call_stream_stats_hack(&call->videostream->ms, stats, ev);
 				if (lc->vtable.call_stats_updated)
-					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
-			} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
-				memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->ms.session), sizeof(jitter_stats_t));
-				if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
-					freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
-				call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
-				evd->packet = NULL;
-				update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
-				if (lc->vtable.call_stats_updated)
-					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
+					lc->vtable.call_stats_updated(lc, call, stats);
 			} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
 				|| (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
 				handle_ice_events(call, ev);
@@ -2300,6 +2308,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 #endif
 	if (call->audiostream!=NULL) {
 		OrtpEvent *ev;
+		LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_AUDIO];
 
 		/* Ensure there is no dangling ICE check list. */
 		if (call->ice_session == NULL) call->audiostream->ms.ice_check_list = NULL;
@@ -2315,24 +2324,10 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 				linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
 			} else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
 				linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
-			} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
-				call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->ms.session);
-				if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
-					freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
-				call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
-				evd->packet = NULL;
-				update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
+			} else if  (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED || evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
+				linphone_call_stream_stats_hack(&call->audiostream->ms, stats, ev);
 				if (lc->vtable.call_stats_updated)
-					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
-			} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
-				memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->ms.session), sizeof(jitter_stats_t));
-				if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
-					freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
-				call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
-				evd->packet = NULL;
-				update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
-				if (lc->vtable.call_stats_updated)
-					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
+					lc->vtable.call_stats_updated(lc, call, stats);
 			} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
 				|| (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
 				handle_ice_events(call, ev);

coreapi/private.h

@@ -730,6 +730,7 @@ typedef enum _LinphoneToneID{
 void linphone_core_play_named_tone(LinphoneCore *lc, LinphoneToneID id);
 bool_t linphone_core_tone_indications_enabled(LinphoneCore*lc);
 
+void linphone_call_stream_stats_hack(MediaStream* stream, LinphoneCallStats *stats, OrtpEvent *ev);
 #ifdef __cplusplus
 }
 #endif

daemon/Makefile.am

@@ -11,6 +11,7 @@ linphone_daemon_SOURCES=daemon.cc \
 			commands/audio-codec-toggle.cc \
 			commands/audio-stream-start.cc \
 			commands/audio-stream-stop.cc \
+			commands/audio-stream-stats.cc \
 			commands/call.cc \
 			commands/call-stats.cc \
 			commands/call-status.cc \
@@ -41,6 +42,7 @@ linphone_daemon_SOURCES=daemon.cc \
 			commands/audio-codec-toggle.h \
 			commands/audio-stream-start.h \
 			commands/audio-stream-stop.h \
+			commands/audio-stream-stats.h \
 			commands/call.h \
 			commands/call-stats.h \
 			commands/call-status.h \

daemon/commands/audio-stream-stats.cc

@@ -0,0 +1,46 @@
+#include "audio-stream-stats.h"
+
+using namespace std;
+
+AudioStreamStatsCommand::AudioStreamStatsCommand() :
+		DaemonCommand("audio-stream-stats", "audio-stream-stats <stream id>", "Return stats of a given audio stream.") {
+	addExample(new DaemonCommandExample("audio-stream-stats 1",
+						"Status: Ok\n\n"
+						"Audio-ICE state: Not activated\n"
+						"Audio-RoundTripDelay: 0.0859833\n"
+						"Audio-Jitter: 296\n"
+						"Audio-JitterBufferSizeMs: 47.7778\n"
+						"Audio-Received-InterarrivalJitter: 154\n"
+						"Audio-Received-FractionLost: 0\n"
+						"Audio-Sent-InterarrivalJitter: 296\n"
+						"Audio-Sent-FractionLost: 0\n"
+						"Audio-Payload-type-number: 111\n"
+						"Audio-Clock-rate: 16000\n"
+						"Audio-Bitrate: 44000\n"
+						"Audio-Mime: speex\n"
+						"Audio-Channels: 1\n"
+						"Audio-Recv-fmtp: vbr=on\n"
+						"Audio-Send-fmtp: vbr=on"));
+	addExample(new DaemonCommandExample("audio-stream-stats 2",
+						"Status: Error\n"
+						"Reason: No audio stream with such id."));
+}
+
+void AudioStreamStatsCommand::exec(Daemon *app, const char *args) {
+	int sid;
+	AudioStreamAndOther *stream = NULL;
+	if (sscanf(args, "%i", &sid) == 1) {
+		stream = app->findAudioStreamAndOther(sid);
+		if (!stream) {
+			app->sendResponse(Response("No audio stream with such id."));
+			return;
+		}
+	} else {
+		app->sendResponse(Response("No stream specified."));
+		return;
+	}
+
+	ostringstream ostr;
+	ostr << AudioStreamStatsResponse(app, stream->stream, &stream->stats, false).getBody();
+	app->sendResponse(Response(ostr.str().c_str(), Response::Ok));
+}

daemon/commands/audio-stream-stats.h

@@ -0,0 +1,12 @@
+#ifndef COMMAND_AUDIO_STREAM_STATS_H_
+#define COMMAND_AUDIO_STREAM_STATS_H_
+
+#include "../daemon.h"
+
+class AudioStreamStatsCommand: public DaemonCommand {
+public:
+	AudioStreamStatsCommand();
+	virtual void exec(Daemon *app, const char *args);
+};
+
+#endif //COMMAND_AUDIO_STREAM_STATS_H_

daemon/daemon.cc

@@ -22,6 +22,7 @@
 #include "commands/audio-codec-toggle.h"
 #include "commands/audio-stream-start.h"
 #include "commands/audio-stream-stop.h"
+#include "commands/audio-stream-stats.h"
 #include "commands/call.h"
 #include "commands/call-stats.h"
 #include "commands/call-status.h"
@@ -43,6 +44,7 @@
 #include "commands/quit.h"
 #include "commands/version.h"
 
+#include "private.h"
 using namespace std;
 
 #ifndef WIN32
@@ -99,25 +101,7 @@ DtmfResponse::DtmfResponse(Daemon *daemon, LinphoneCall *call, int dtmf) {
 	ms_free(remote);
 }
 
-CallStatsResponse::CallStatsResponse(Daemon *daemon, LinphoneCall *call, const LinphoneCallStats *stats, bool event) {
-	const LinphoneCallParams *callParams = linphone_call_get_current_params(call);
-	const char *prefix = "";
-
-	ostringstream ostr;
-	if (event) {
-		ostr << "Event-type: call-stats\n";
-		ostr << "Id: " << daemon->updateCallId(call) << "\n";
-		ostr << "Type: ";
-		if (stats->type == LINPHONE_CALL_STATS_AUDIO) {
-			ostr << "Audio";
-		} else {
-			ostr << "Video";
-		}
-		ostr << "\n";
-	} else {
-		prefix = ((stats->type == LINPHONE_CALL_STATS_AUDIO) ? "Audio-" : "Video-");
-	}
-
+static ostream &printCallStatsHelper(ostream &ostr, const LinphoneCallStats *stats, const string &prefix) {
 	ostr << prefix << "ICE state: " << ice_state_str[stats->ice_state] << "\n";
 	ostr << prefix << "RoundTripDelay: " << stats->round_trip_delay << "\n";
 	ostr << prefix << "Jitter: " << stats->jitter_stats.jitter << "\n";
@@ -163,6 +147,29 @@ CallStatsResponse::CallStatsResponse(Daemon *daemon, LinphoneCall *call, const L
 		ostr << prefix << "Sent-InterarrivalJitter: " << ij << "\n";
 		ostr << prefix << "Sent-FractionLost: " << flost << "\n";
 	}
+	return ostr;
+}
+
+CallStatsResponse::CallStatsResponse(Daemon *daemon, LinphoneCall *call, const LinphoneCallStats *stats, bool event) {
+	const LinphoneCallParams *callParams = linphone_call_get_current_params(call);
+	const char *prefix = "";
+
+	ostringstream ostr;
+	if (event) {
+		ostr << "Event-type: call-stats\n";
+		ostr << "Id: " << daemon->updateCallId(call) << "\n";
+		ostr << "Type: ";
+		if (stats->type == LINPHONE_CALL_STATS_AUDIO) {
+			ostr << "Audio";
+		} else {
+			ostr << "Video";
+		}
+		ostr << "\n";
+	} else {
+		prefix = ((stats->type == LINPHONE_CALL_STATS_AUDIO) ? "Audio-" : "Video-");
+	}
+
+	printCallStatsHelper(ostr, stats, prefix);
 
 	if (stats->type == LINPHONE_CALL_STATS_AUDIO) {
 		const PayloadType *audioCodec = linphone_call_params_get_used_audio_codec(callParams);
@@ -175,6 +182,31 @@ CallStatsResponse::CallStatsResponse(Daemon *daemon, LinphoneCall *call, const L
 	setBody(ostr.str().c_str());
 }
 
+
+AudioStreamStatsResponse::AudioStreamStatsResponse(Daemon* daemon, AudioStream* stream,
+		const LinphoneCallStats *stats, bool event) {
+	const char *prefix = "";
+	
+	ostringstream ostr;
+	if (event) {
+		ostr << "Event-type: audio-stream-stats\n";
+		ostr << "Id: " << daemon->updateAudioStreamId(stream) << "\n";
+		ostr << "Type: ";
+		if (stats->type == LINPHONE_CALL_STATS_AUDIO) {
+			ostr << "Audio";
+		} else {
+			ostr << "Video";
+		}
+		ostr << "\n";
+	} else {
+		prefix = ((stats->type == LINPHONE_CALL_STATS_AUDIO) ? "Audio-" : "Video-");
+	}
+	
+	printCallStatsHelper(ostr, stats, prefix);
+	
+	setBody(ostr.str().c_str());
+}
+
 PayloadTypeResponse::PayloadTypeResponse(LinphoneCore *core, const PayloadType *payloadType, int index, const string &prefix, bool enabled_status) {
 	ostringstream ostr;
 	if (payloadType != NULL) {
@@ -336,27 +368,36 @@ LinphoneProxyConfig *Daemon::findProxy(int id) {
 }
 
 int Daemon::updateAudioStreamId(AudioStream *audio_stream) {
-	for (std::map<int, AudioStream*>::iterator it = mAudioStreams.begin(); it != mAudioStreams.end(); ++it) {
-		if (it->second == audio_stream)
+	for (std::map<int, AudioStreamAndOther*>::iterator it = mAudioStreams.begin(); it != mAudioStreams.end(); ++it) {
+		if (it->second->stream == audio_stream)
 			return it->first;
 	}
 
 	++mProxyIds;
-	mAudioStreams.insert(std::pair<int, AudioStream*>(mProxyIds, audio_stream));
+	mAudioStreams.insert(make_pair(mProxyIds, new AudioStreamAndOther(audio_stream)));
 	return mProxyIds;
 }
 
-AudioStream *Daemon::findAudioStream(int id) {
-	std::map<int, AudioStream*>::iterator it = mAudioStreams.find(id);
+AudioStreamAndOther *Daemon::findAudioStreamAndOther(int id) {
+	std::map<int, AudioStreamAndOther*>::iterator it = mAudioStreams.find(id);
 	if (it != mAudioStreams.end())
 		return it->second;
 	return NULL;
 }
 
-void Daemon::removeAudioStream(int id) {
-	std::map<int, AudioStream*>::iterator it = mAudioStreams.find(id);
+AudioStream *Daemon::findAudioStream(int id) {
+	std::map<int, AudioStreamAndOther*>::iterator it = mAudioStreams.find(id);
 	if (it != mAudioStreams.end())
+		return it->second->stream;
+	return NULL;
+}
+
+void Daemon::removeAudioStream(int id) {
+	std::map<int, AudioStreamAndOther*>::iterator it = mAudioStreams.find(id);
+	if (it != mAudioStreams.end()) {
 		mAudioStreams.erase(it);
+		delete(it->second);
+	}
 }
 
 void Daemon::initCommands() {
@@ -379,6 +420,7 @@ void Daemon::initCommands() {
 	mCommands.push_back(new AudioCodecSetCommand());
 	mCommands.push_back(new AudioStreamStartCommand());
 	mCommands.push_back(new AudioStreamStopCommand());
+	mCommands.push_back(new AudioStreamStatsCommand());
 	mCommands.push_back(new MSFilterAddFmtpCommand());
 	mCommands.push_back(new PtimeCommand());
 	mCommands.push_back(new IPv6Command());
@@ -452,8 +494,24 @@ void Daemon::dtmfReceived(LinphoneCore *lc, LinphoneCall *call, int dtmf) {
 	app->dtmfReceived(call, dtmf);
 }
 
+void Daemon::iterateStreamStats() {
+	for (std::map<int, AudioStreamAndOther*>::iterator it = mAudioStreams.begin(); it != mAudioStreams.end(); ++it) {
+		OrtpEvent *ev;
+		while (it->second->queue && (NULL != (ev=ortp_ev_queue_get(it->second->queue)))){
+			OrtpEventType evt=ortp_event_get_type(ev);
+			if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED || evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
+				linphone_call_stream_stats_hack(&it->second->stream->ms, &it->second->stats, ev);
+				if (mUseStatsEvents) mEventQueue.push(new AudioStreamStatsResponse(this,
+					it->second->stream, &it->second->stats, true));
+			}
+			ortp_event_destroy(ev);
+		}
+	}
+}
+
 void Daemon::iterate() {
 	linphone_core_iterate(mLc);
+	iterateStreamStats();
 	if (mChildFd == -1) {
 		if (!mEventQueue.empty()) {
 			Response *r = mEventQueue.front();
@@ -707,8 +765,8 @@ void Daemon::enableLSD(bool enabled) {
 Daemon::~Daemon() {
 	uninitCommands();
 
-	for (std::map<int, AudioStream *>::iterator it = mAudioStreams.begin(); it != mAudioStreams.end(); ++it) {
-		audio_stream_stop(it->second);
+	for (std::map<int, AudioStreamAndOther *>::iterator it = mAudioStreams.begin(); it != mAudioStreams.end(); ++it) {
+		audio_stream_stop(it->second->stream);
 	}
 
 	enableLSD(false);

daemon/daemon.h

@@ -126,25 +126,27 @@ private:
 class EventResponse: public Response {
 public:
 	EventResponse(Daemon *daemon, LinphoneCall *call, LinphoneCallState state);
-private:
 };
 
 class CallStatsResponse: public Response {
 public:
 	CallStatsResponse(Daemon *daemon, LinphoneCall *call, const LinphoneCallStats *stats, bool unique);
-private:
+};
+
+class AudioStreamStatsResponse: public Response {
+public:
+	AudioStreamStatsResponse(Daemon *daemon, AudioStream *stream,
+		const LinphoneCallStats *stats, bool event);
 };
 
 class DtmfResponse: public Response {
 public:
 	DtmfResponse(Daemon *daemon, LinphoneCall *call, int dtmf);
-private:
 };
 
 class PayloadTypeResponse: public Response {
 public:
 	PayloadTypeResponse(LinphoneCore *core, const PayloadType *payloadType, int index = -1, const std::string &prefix = std::string(), bool enabled_status = true);
-private:
 };
 
 class PayloadTypeParser {
@@ -159,6 +161,21 @@ private:
 	int mPayloadTypeNumber;
 };
 
+struct AudioStreamAndOther {
+	AudioStream *stream;
+	OrtpEvQueue *queue;
+	LinphoneCallStats stats;
+	AudioStreamAndOther(AudioStream *as) : stream(as) {
+		queue = ortp_ev_queue_new();
+		rtp_session_register_event_queue(as->ms.session, queue);
+		memset(&stats, 0, sizeof(stats));
+	}
+	~AudioStreamAndOther() {
+		rtp_session_unregister_event_queue(stream->ms.session, queue);
+		ortp_ev_queue_destroy(queue);
+	}
+};
+
 class Daemon {
 	friend class DaemonCommand;
 public:
@@ -174,6 +191,7 @@ public:
 	LinphoneCall *findCall(int id);
 	LinphoneProxyConfig *findProxy(int id);
 	AudioStream *findAudioStream(int id);
+	AudioStreamAndOther *findAudioStreamAndOther(int id);
 	void removeAudioStream(int id);
 	bool pullEvent();
 	int updateCallId(LinphoneCall *call);
@@ -196,10 +214,12 @@ private:
 	char *readLine(const char *);
 	char *readPipe(char *buffer, int buflen);
 	void iterate();
+	void iterateStreamStats();
 	void startThread();
 	void stopThread();
 	void initCommands();
 	void uninitCommands();
+	void iterateStreamStats(LinphoneCore *lc);
 	LinphoneCore *mLc;
 	LinphoneSoundDaemon *mLSD;
 	std::list<DaemonCommand*> mCommands;
@@ -216,7 +236,7 @@ private:
 	ms_thread_t mThread;
 	ms_mutex_t mMutex;
 	static const int sLineSize = 512;
-	std::map<int, AudioStream*> mAudioStreams;
+	std::map<int, AudioStreamAndOther*> mAudioStreams;
 };
 
 #endif //DAEMON_H_

mediastreamer2

@@ -1 +1 @@
-Subproject commit aebfe443e91aa1e1c1e32b2410cc1d9a5db629c1
+Subproject commit 43b68b22cf8d0d1b0b5a064516d35899e5527358

oRTP

@@ -1 +1 @@
-Subproject commit 49b16793b9ef8251a4c42434b57387c6e3c6d251
+Subproject commit 7c2805c2e2c1edadf9e9efe313d98f06c4c3614a