Stream audio over one channel only (b8393f4e) · Commits · Elphel / elphel-apps-astreamer

src/audio.cpp

+18 −11

Original line number	Original line	Diff line number	Diff line
	@@ -55,10 +55,11 @@
	#define SAMPLE_TIME 20 //< restrict ALSA to have this period, 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 BUFFER_TIME 1000 //< approximate buffer duration for ALSA, in milliseconds
	#define LEN 1200 //< the size of data buffer for RTP packet, in bytes		#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;		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_hw_params_t *hw_params;
	snd_pcm_sw_params_t *sw_params;		snd_pcm_sw_params_t *sw_params;
	_present = false;		_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];		sbuffer = new short[sbuffer_len * 2 * _channels];
	packet_buffer = new unsigned char[LEN + 20];		packet_buffer = new unsigned char[LEN + 20];
	bool init_ok = false;		bool init_ok = false;

			if (dev_name == "")
			dev_name = DEFAULT_SND_DEVICE;
	while (true) {		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;		break;
	snd_pcm_hw_params_alloca(&hw_params);		snd_pcm_hw_params_alloca(&hw_params);
	if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0)		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;		break;
	if ((err = snd_pcm_sw_params_set_tstamp_mode(capture_handle, sw_params, SND_PCM_TSTAMP_ENABLE)) < 0)		if ((err = snd_pcm_sw_params_set_tstamp_mode(capture_handle, sw_params, SND_PCM_TSTAMP_ENABLE)) < 0)
	break;		break;
	/* SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY for some reason does not produce time stamps equal to system time,		if ((err = snd_pcm_sw_params_set_tstamp_type(capture_handle, sw_params, SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY)) < 0)
	* 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)
	break;		break;
	if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)		if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)
	break;		break;
	@@ -153,6 +159,7 @@ Audio::Audio(int port, bool enable, Parameters *pars, int sample_rate, int chann
	init_pthread((void *) this);		init_pthread((void *) this);
	} else {		} else {
	D(sensor_port, cerr << "Audio: init FAIL!" << endl);		D(sensor_port, cerr << "Audio: init FAIL!" << endl);
			D(sensor_port, cerr << "ALSA error message: " << snd_strerror(err) << endl);
	_present = false;		_present = false;
	}		}
	}		}
	@@ -224,8 +231,8 @@ void Audio::set_capture_volume(int nvolume) {
	* @brief Start audio stream if it was enabled		* @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.		* 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		* 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		* between FPGA time and system time reported by \e getsystimeofday(), but this was changed because of some problems
	* time stamps regardless of the time stamp type set by \e snd_pcm_sw_params_set_tstamp_type(). See git commit		* with getting system time stamps from sound device opened as "default" ("plughw" is used now). See git commit
	* 58b954f239b695b7deda7a33657841d2c64476ae (or earlier) for previous implementation.		* 58b954f239b695b7deda7a33657841d2c64476ae (or earlier) for previous implementation.
	* @param ip ip/port pair for socket		* @param ip ip/port pair for socket
	* @param port ip/port pair for socket		* @param port ip/port pair for socket

src/audio.h

+1 −4

Original line number	Original line	Diff line number	Diff line
	@@ -40,7 +40,7 @@ using namespace std;

	class Audio : public RTP_Stream {		class Audio : public RTP_Stream {
	public:		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);		virtual ~Audio(void);
	long sample_rate(void) { return _sample_rate; };		long sample_rate(void) { return _sample_rate; };
	long channels(void) { return _channels; };		long channels(void) { return _channels; };
	@@ -51,7 +51,6 @@ public:
	void Start(string ip, long port, int ttl = -1);		void Start(string ip, long port, int ttl = -1);
	void Stop(void);		void Stop(void);
	protected:		protected:
	// int fd;
	snd_pcm_t *capture_handle; //< PCM handle, returned from snd_pcm_open		snd_pcm_t *capture_handle; //< PCM handle, returned from snd_pcm_open
	short *sbuffer; //< buffer for sound data		short *sbuffer; //< buffer for sound data
	long sbuffer_len;		long sbuffer_len;
	@@ -67,8 +66,6 @@ protected:

	uint64_t timestamp_rtcp;		uint64_t timestamp_rtcp;
	long delta_fpga_sys; //< A/V clocks delta for 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		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)		unsigned char *packet_buffer; //< buffer for RTP packet data (header plus PCM samples)
	private:		private:

src/main.cpp

+37 −3

Original line number	Original line	Diff line number	Diff line
	@@ -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 main(int argc, char *argv[]) {
	int ret_val;		int ret_val;
			int audio_port = -1;
	string opt;		string opt;
	map<string, string> args;		map<string, string> args;
			map<string, string>::iterator args_it;
	pthread_t threads[SENSOR_PORTS];		pthread_t threads[SENSOR_PORTS];
	Streamer *streamers[SENSOR_PORTS] = {NULL};		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++) {		for (int i = 1; i < argc; i++) {
	if (argv[i][0] == '-' && argv[i][1] != '\0') {		if (argv[i][0] == '-' && argv[i][1] != '\0') {
	if (opt != "")		if (opt != "")
	@@ -72,12 +92,26 @@ int main(int argc, char *argv[]) {
	for (map<string, string>::iterator it = args.begin(); it != args.end(); it++) {		for (map<string, string>::iterator it = args.begin(); it != args.end(); it++) {
	cerr << "\|" << (it).first << "\| == \|" << (it).second << "\|" << endl;		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;		pthread_attr_t attr;

			if (i == audio_port)
			audio_en = true;
			else
			audio_en = false;
	cout << "Start thread " << i << endl;		cout << "Start thread " << i << endl;
	streamers[i] = new Streamer(args, i);		streamers[i] = new Streamer(args, i, audio_en);

	pthread_attr_init(&attr);		pthread_attr_init(&attr);
	ret_val = pthread_create(&threads[i], &attr, Streamer::pthread_f, (void *) streamers[i]);		ret_val = pthread_create(&threads[i], &attr, Streamer::pthread_f, (void *) streamers[i]);

src/streamer.cpp

+19 −17

Original line number	Original line	Diff line number	Diff line
	@@ -56,16 +56,14 @@ using namespace std;
	#define D2(s_port, a)		#define D2(s_port, a)
	#endif		#endif

	//Streamer *Streamer::_streamer = NULL;		Streamer::Streamer(const map<string, string> &_args, int port_num, bool audio_en) {

	Streamer::Streamer(const map<string, string> &_args, int port_num) {
	sensor_port = port_num;		sensor_port = port_num;
	_streamer = this;		_streamer = this;
	session = new Session();		session = new Session();
	params = new Parameters(sensor_port);		params = new Parameters(sensor_port);
	args = _args;		args = _args;
	audio = NULL;		audio = NULL;
	session->process_audio = true;		session->process_audio = audio_en;
	session->audio.sample_rate = 0;		session->audio.sample_rate = 0;
	session->audio.channels = 0;		session->audio.channels = 0;
	session->rtp_out.ip_custom = false;		session->rtp_out.ip_custom = false;
	@@ -98,7 +96,12 @@ void Streamer::audio_init(void) {
	delete audio;		delete audio;
	}		}
	D(sensor_port, cout << "audio_enabled == " << session->process_audio << endl);		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 (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) {		if (audio->present() && session->process_audio) {
	session->process_audio = true;		session->process_audio = true;
	session->audio.type = audio->ptype();		session->audio.type = audio->ptype();
	@@ -111,6 +114,7 @@ void Streamer::audio_init(void) {
	session->audio.channels = 0;		session->audio.channels = 0;
	}		}
	}		}
			}

	Streamer::~Streamer(void) {		Streamer::~Streamer(void) {
	delete video;		delete video;
	@@ -278,8 +282,6 @@ int Streamer::update_settings(bool apply) {
	f_audio_channels = true;		f_audio_channels = true;
	if ((audio_proc \|\| session->process_audio) && (f_audio_rate \|\| f_audio_channels))		if ((audio_proc \|\| session->process_audio) && (f_audio_rate \|\| f_audio_channels))
	audio_was_changed = true;		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) {		if (apply) {
	bool audio_restarted = false;		bool audio_restarted = false;
	if (audio_was_changed) {		if (audio_was_changed) {

src/streamer.h

+1 −1

Original line number	Original line	Diff line number	Diff line
	@@ -33,7 +33,7 @@ using namespace std;

	class Streamer {		class Streamer {
	public:		public:
	Streamer(const map<string, string> &args, int port_num);		Streamer(const map<string, string> &args, int port_num, bool audio_en = false);
	~Streamer();		~Streamer();
	void Main(void);		void Main(void);
	bool opt_present(string name) {		bool opt_present(string name) {