Stream audio over one channel only

The port number for audio streaming is passed in command-line arguments.
The sound device can also be specified on command line and default
device now is set to plughw:0,0 due to some problems with time stamps
("default" device did not produce system time stamps).

src/audio.cpp

@@ -52,13 +52,14 @@
 	#define D2(s_port, a)
 #endif
 
-#define SAMPLE_TIME	              20	                        //< restrict ALSA to have this period, in milliseconds
-#define BUFFER_TIME	              1000                          //< approximate buffer duration for ALSA, in milliseconds
+#define SAMPLE_TIME               20                            //< restrict ALSA to have this period, in milliseconds
+#define BUFFER_TIME               1000                          //< approximate buffer duration for ALSA, in milliseconds
 #define LEN                       1200                          //< the size of data buffer for RTP packet, in bytes
+#define DEFAULT_SND_DEVICE        "plughw:0,0"                  //< default sound card
 
 using namespace std;
 
-Audio::Audio(int port, bool enable, Parameters *pars, int sample_rate, int channels) {
+Audio::Audio(int port, bool enable, Parameters *pars, string &dev_name, int sample_rate, int channels) {
 	snd_pcm_hw_params_t *hw_params;
 	snd_pcm_sw_params_t *sw_params;
 	_present = false;
@@ -101,8 +102,17 @@ Audio::Audio(int port, bool enable, Parameters *pars, int sample_rate, int chann
 		sbuffer = new short[sbuffer_len * 2 * _channels];
 		packet_buffer = new unsigned char[LEN + 20];
 		bool init_ok = false;
+
+		if (dev_name == "")
+			dev_name = DEFAULT_SND_DEVICE;
 		while (true) {
-			if ((err = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0)) < 0)
+			/* Previously, "default" device was used in snd_pcm_open(), but this resulted in incorrect time stamps
+			 * reported by snd_pcm_status_get_tstamp() (probably a bug in ALSA plugins). The time stamps did not correspond
+			 * to system time, but rather started from 0 after reboot and then increased. Opening "plughw:0,0" device
+			 * produces correct time stamps, thus this is the default device name now and it can be set from command line.
+			 */
+			D2(sensor_port, cerr << "Trying to open " << dev_name << endl);
+			if ((err = snd_pcm_open(&capture_handle, dev_name.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0)
 				break;
 			snd_pcm_hw_params_alloca(&hw_params);
 			if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0)
@@ -133,11 +143,7 @@ Audio::Audio(int port, bool enable, Parameters *pars, int sample_rate, int chann
 				break;
 			if ((err = snd_pcm_sw_params_set_tstamp_mode(capture_handle, sw_params, SND_PCM_TSTAMP_ENABLE)) < 0)
 				break;
-			/* SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY for some reason does not produce time stamps equal to system time,
-			 * so stick with monotonic time stamps and apply the difference between FPGA time and ALSA time stamps to
-			 * the current samples during transmission
-			 */
-			if ((err = snd_pcm_sw_params_set_tstamp_type(capture_handle, sw_params, SND_PCM_TSTAMP_TYPE_MONOTONIC)) < 0)
+			if ((err = snd_pcm_sw_params_set_tstamp_type(capture_handle, sw_params, SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY)) < 0)
 				break;
 			if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)
 				break;
@@ -153,6 +159,7 @@ Audio::Audio(int port, bool enable, Parameters *pars, int sample_rate, int chann
 			init_pthread((void *) this);
 		} else {
 			D(sensor_port, cerr << "Audio: init FAIL!" << endl);
+			D(sensor_port, cerr << "ALSA error message: " << snd_strerror(err) << endl);
 			_present = false;
 		}
 	}
@@ -224,8 +231,8 @@ void Audio::set_capture_volume(int nvolume) {
  * @brief Start audio stream if it was enabled
  * The time delta between FPGA time and time stamps reported by ALSA is calculated before stream is started.
  * This delta is applied to time stamps received from ALSA. Previously, this delta was calculated as time difference
- * between FPGA time and system time reported by \e getsystimeofday(), but as for now ALSA reports monotonic
- * time stamps regardless of the time stamp type set by \e snd_pcm_sw_params_set_tstamp_type(). See git commit
+ * between FPGA time and system time reported by \e getsystimeofday(), but this was changed because of some problems
+ * with getting system time stamps from sound device opened as "default" ("plughw" is used now). See git commit
  * 58b954f239b695b7deda7a33657841d2c64476ae (or earlier) for previous implementation.
  * @param   ip   ip/port pair for socket
  * @param   port ip/port pair for socket

src/audio.h

@@ -40,7 +40,7 @@ using namespace std;
 
 class Audio : public RTP_Stream {
 public:
-	Audio(int port, bool enable, Parameters *pars, int sample_rate = SAMPLE_RATE, int channels = SAMPLE_CHANNELS);
+	Audio(int port, bool enable, Parameters *pars, string &dev_name, int sample_rate = SAMPLE_RATE, int channels = SAMPLE_CHANNELS);
 	virtual ~Audio(void);
 	long sample_rate(void) { return _sample_rate; };
 	long channels(void) { return _channels; };
@@ -51,7 +51,6 @@ public:
 	void Start(string ip, long port, int ttl = -1);
 	void Stop(void);
 protected:
-//	int fd;
 	snd_pcm_t *capture_handle;                                  //< PCM handle, returned from snd_pcm_open
 	short *sbuffer;                                             //< buffer for sound data
 	long sbuffer_len;
@@ -67,8 +66,6 @@ protected:
 
 	uint64_t timestamp_rtcp;
 	long delta_fpga_sys;	                                    //< A/V clocks delta for RTCP
-//	bool is_first;
-//	bool is_first2;
 	long long delta_fpga_alsa;                                  //< time delta between FPGA time and time reported by ALSA, in microseconds
 	unsigned char *packet_buffer;                               //< buffer for RTP packet data (header plus PCM samples)
 private:

src/main.cpp

@@ -47,13 +47,33 @@ void clean_up(pthread_t *threads, size_t sz) {
 	}
 }
 
+/**
+ * Print help message on stdout.
+ * @param   argv   a list of command-line arguments, used to get the name of application
+ * @return  None
+ */
+void print_help(char *argv[])
+{
+	const char *msg = "Simple RTSP streamer implementation for Elphel393 series cameras.\n"
+			"Usage: %s [-s <port>][-D <sound_card>][-f <fps>][-h], where\n\n"
+			"\t-h\t\tprint this help message;\n"
+			"\t-s <port>\tstream sound from USB microphone over sensor port <port> channel. By default, audio streaming is not\n"
+			"\t\t\tenabled if this option is not specified;\n"
+			"\t-D <sound_card>\tuse <sound_card> device for sound stream input. The default device is plughw:0,0;\n"
+			"\t-f <fps>\tlimit frames per second for video streaming, works for free running mode only.\n"; // this one is processed in streamer class
+	printf(msg, argv[0]);
+}
+
 int main(int argc, char *argv[]) {
 	int ret_val;
+	int audio_port = -1;
 	string opt;
 	map<string, string> args;
+	map<string, string>::iterator args_it;
 	pthread_t threads[SENSOR_PORTS];
 	Streamer *streamers[SENSOR_PORTS] = {NULL};
 
+	// copy command-line arguments to a map container for further processing in streamer class
 	for (int i = 1; i < argc; i++) {
 		if (argv[i][0] == '-' && argv[i][1] != '\0') {
 			if (opt != "")
@@ -72,12 +92,26 @@ int main(int argc, char *argv[]) {
 	for (map<string, string>::iterator it = args.begin(); it != args.end(); it++) {
 		cerr << "|" << (*it).first << "| == |" << (*it).second << "|" << endl;
 	}
+	if ((args_it = args.find("h")) != args.end()) {
+		print_help(argv);
+		exit(EXIT_SUCCESS);
+	} else if ((args_it = args.find("s")) != args.end()) {
+		audio_port = strtol(args_it->second.c_str(), NULL, 10);
+		// sanity check, invalid conversion produces 0 which is fine
+		if (audio_port < 0 || audio_port >= SENSOR_PORTS)
+			audio_port = -1;
+	}
 
-	for (int i = 0; i < 1; i++) {
-//	for (int i = 0; i < SENSOR_PORTS; i++) {
+	for (int i = 0; i < SENSOR_PORTS; i++) {
+		bool audio_en;
 		pthread_attr_t attr;
+
+		if (i == audio_port)
+			audio_en = true;
+		else
+			audio_en = false;
 		cout << "Start thread " << i << endl;
-		streamers[i] = new Streamer(args, i);
+		streamers[i] = new Streamer(args, i, audio_en);
 
 		pthread_attr_init(&attr);
 		ret_val = pthread_create(&threads[i], &attr, Streamer::pthread_f, (void *) streamers[i]);

src/streamer.cpp

@@ -56,16 +56,14 @@ using namespace std;
 	#define D2(s_port, a)
 #endif
 
-//Streamer *Streamer::_streamer = NULL;
-
-Streamer::Streamer(const map<string, string> &_args, int port_num) {
+Streamer::Streamer(const map<string, string> &_args, int port_num, bool audio_en) {
 	sensor_port = port_num;
 	_streamer = this;
 	session = new Session();
 	params = new Parameters(sensor_port);
 	args = _args;
 	audio = NULL;
-	session->process_audio = true;
+	session->process_audio = audio_en;
 	session->audio.sample_rate = 0;
 	session->audio.channels = 0;
 	session->rtp_out.ip_custom = false;
@@ -98,17 +96,23 @@ void Streamer::audio_init(void) {
 		delete audio;
 	}
 	D(sensor_port, cout << "audio_enabled == " << session->process_audio << endl);
-	audio = new Audio(sensor_port, session->process_audio, params, session->audio.sample_rate, session->audio.channels);
-	if (audio->present() && session->process_audio) {
-		session->process_audio = true;
-		session->audio.type = audio->ptype();
-		session->audio.sample_rate = audio->sample_rate();
-		session->audio.channels = audio->channels();
-	} else {
-		session->process_audio = false;
-		session->audio.type = -1;
-		session->audio.sample_rate = 0;
-		session->audio.channels = 0;
+	if (session->process_audio) {
+		string dev_name = "";
+		map<string, string>::iterator args_it;
+		if ((args_it = args.find("D")) != args.end())
+			dev_name = args_it->second;
+		audio = new Audio(sensor_port, session->process_audio, params, dev_name, session->audio.sample_rate, session->audio.channels);
+		if (audio->present() && session->process_audio) {
+			session->process_audio = true;
+			session->audio.type = audio->ptype();
+			session->audio.sample_rate = audio->sample_rate();
+			session->audio.channels = audio->channels();
+		} else {
+			session->process_audio = false;
+			session->audio.type = -1;
+			session->audio.sample_rate = 0;
+			session->audio.channels = 0;
+		}
 	}
 }
 
@@ -278,8 +282,6 @@ int Streamer::update_settings(bool apply) {
 		f_audio_channels = true;
 	if ((audio_proc || session->process_audio) && (f_audio_rate || f_audio_channels))
 		audio_was_changed = true;
-	D(sensor_port, cerr << "audio_proc = " << audio_proc << ", process_audio = " << session->process_audio << ", f_audio_rate = " << f_audio_rate
-			<< ", f_audio_channels = " << f_audio_channels << endl);
 	if (apply) {
 		bool audio_restarted = false;
 		if (audio_was_changed) {

src/streamer.h

@@ -33,7 +33,7 @@ using namespace std;
 
 class Streamer {
 public:
-	Streamer(const map<string, string> &args, int port_num);
+	Streamer(const map<string, string> &args, int port_num, bool audio_en = false);
 	~Streamer();
 	void Main(void);
 	bool opt_present(string name) {