gnunet-helper-audio-record-gst.c File Reference

program to record audio data from the microphone (GStreamer version) More...

#include "platform.h"
#include "gnunet_protocols.h"
#include "conversation.h"
#include <gst/gst.h>
#include <gst/app/gstappsink.h>
#include <gst/audio/gstaudiobasesrc.h>
#include <glib.h>

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Go to the source code of this file.

Macros
#define	DEBUG_RECORD_PURE_OGG   1
#define	OPUS_CHANNELS   1
	Number of channels. More...
#define	MAX_PAYLOAD_SIZE   (1024 / OPUS_CHANNELS)
	Maximal size of a single opus packet. More...
#define	OPUS_FRAME_SIZE   40
	Size of a single frame fed to the encoder, in ms. More...
#define	PACKET_LOSS_PERCENTAGE   1
	Expected packet loss to prepare for, in percents. More...
#define	INBAND_FEC_MODE   1
	Set to 1 to enable forward error correction. More...
#define	BUFFER_TIME   1000 /* 1ms */
	Max number of microseconds to buffer in audiosource. More...
#define	LATENCY_TIME   1000 /* 1ms */
	Min number of microseconds to buffer in audiosource. More...
#define	OGG_MAX_DELAY   0
	Maximum delay in multiplexing streams, in ns. More...
#define	OGG_MAX_PAGE_DELAY   0
	Maximum delay for sending out a page, in ns. More...

Functions
static void	quit ()
static gboolean	bus_call (GstBus *bus, GstMessage *msg, gpointer data)
static void	source_child_added (GstChildProxy *child_proxy, GObject *object, gchar *name, gpointer user_data)
static void	signalhandler (int s)
int	main (int argc, char **argv)

Variables
static GstElement *	pipeline
	Main pipeline. More...
static int	dump_pure_ogg

Detailed Description

program to record audio data from the microphone (GStreamer version)

Author

LRN

Definition in file gnunet-helper-audio-record-gst.c.

Macro Definition Documentation

DEBUG_RECORD_PURE_OGG

#define DEBUG_RECORD_PURE_OGG   1

Definition at line 34 of file gnunet-helper-audio-record-gst.c.

OPUS_CHANNELS

#define OPUS_CHANNELS   1

Number of channels.

Must be one of the following (from libopusenc documentation): 1, 2

Definition at line 41 of file gnunet-helper-audio-record-gst.c.

MAX_PAYLOAD_SIZE

#define MAX_PAYLOAD_SIZE   (1024 / OPUS_CHANNELS)

Maximal size of a single opus packet.

Definition at line 46 of file gnunet-helper-audio-record-gst.c.

OPUS_FRAME_SIZE

#define OPUS_FRAME_SIZE   40

Size of a single frame fed to the encoder, in ms.

Must be one of the following (from libopus documentation): 2.5, 5, 10, 20, 40 or 60

Definition at line 53 of file gnunet-helper-audio-record-gst.c.

PACKET_LOSS_PERCENTAGE

#define PACKET_LOSS_PERCENTAGE   1

Expected packet loss to prepare for, in percents.

Definition at line 58 of file gnunet-helper-audio-record-gst.c.

INBAND_FEC_MODE

#define INBAND_FEC_MODE   1

Set to 1 to enable forward error correction.

Set to 0 to disable.

Definition at line 64 of file gnunet-helper-audio-record-gst.c.

BUFFER_TIME

#define BUFFER_TIME   1000 /* 1ms */

Max number of microseconds to buffer in audiosource.

Default is 200000

Definition at line 70 of file gnunet-helper-audio-record-gst.c.

LATENCY_TIME

#define LATENCY_TIME   1000 /* 1ms */

Min number of microseconds to buffer in audiosource.

Default is 10000

Definition at line 76 of file gnunet-helper-audio-record-gst.c.

OGG_MAX_DELAY

#define OGG_MAX_DELAY   0

Maximum delay in multiplexing streams, in ns.

Setting this to 0 forces page flushing, which decreases delay, but increases overhead.

Definition at line 83 of file gnunet-helper-audio-record-gst.c.

OGG_MAX_PAGE_DELAY

#define OGG_MAX_PAGE_DELAY   0

Maximum delay for sending out a page, in ns.

Setting this to 0 forces page flushing, which decreases delay, but increases overhead.

Definition at line 90 of file gnunet-helper-audio-record-gst.c.

Function Documentation

quit()

static void quit ( )

static

Definition at line 102 of file gnunet-helper-audio-record-gst.c.

{
  if (NULL != pipeline)
    gst_element_set_state (pipeline, GST_STATE_NULL);
}

References pipeline.

Referenced by bus_call(), main(), and signalhandler().

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bus_call()

static gboolean bus_call ( GstBus *  bus, GstMessage *  msg, gpointer  data  )

static

Definition at line 110 of file gnunet-helper-audio-record-gst.c.

{
  GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Bus message\n");
  switch (GST_MESSAGE_TYPE (msg))
  {
  case GST_MESSAGE_EOS:
    GNUNET_log (GNUNET_ERROR_TYPE_INFO, "End of stream\n");
    quit ();
    break;
 
  case GST_MESSAGE_ERROR:
    {
      gchar  *debug;
      GError *error;
 
      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);
 
      GNUNET_log (GNUNET_ERROR_TYPE_ERROR, "Error: %s\n", error->message);
      g_error_free (error);
 
      quit ();
      break;
    }
 
  default:
    break;
  }
 
  return TRUE;
}

References find_typedefs::debug, GNUNET_ERROR_TYPE_DEBUG, GNUNET_ERROR_TYPE_ERROR, GNUNET_ERROR_TYPE_INFO, GNUNET_log, msg, and quit().

Referenced by main().

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source_child_added()

static void source_child_added ( GstChildProxy *  child_proxy, GObject *  object, gchar *  name, gpointer  user_data  )

static

Definition at line 144 of file gnunet-helper-audio-record-gst.c.

{
  if (GST_IS_AUDIO_BASE_SRC (object))
    g_object_set (object, "buffer-time", (gint64) BUFFER_TIME, "latency-time",
                  (gint64) LATENCY_TIME, NULL);
}

References BUFFER_TIME, and LATENCY_TIME.

Referenced by main().

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signalhandler()

static void signalhandler ( int  s )

static

Definition at line 154 of file gnunet-helper-audio-record-gst.c.

{
  quit ();
}

References quit().

Referenced by main().

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main()

int main	(	int	argc,
		char **	argv
	)

Definition at line 161 of file gnunet-helper-audio-record-gst.c.

{
  GstElement *source, *filter, *encoder, *conv, *resampler, *sink, *oggmux;
  GstCaps *caps;
  GstBus *bus;
  guint bus_watch_id;
  struct AudioMessage audio_message;
  int abort_send = 0;
 
  typedef void (*SignalHandlerPointer) (int);
 
  SignalHandlerPointer inthandler, termhandler;
  inthandler = signal (SIGINT, signalhandler);
  termhandler = signal (SIGTERM, signalhandler);
 
#ifdef DEBUG_RECORD_PURE_OGG
  dump_pure_ogg = getenv ("GNUNET_RECORD_PURE_OGG") ? 1 : 0;
#endif
 
  /* Initialisation */
  gst_init (&argc, &argv);
 
  GNUNET_assert (GNUNET_OK ==
                 GNUNET_log_setup ("gnunet-helper-audio-record",
                                   "WARNING",
                                   NULL));
 
  GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
              "Audio source starts\n");
 
  audio_message.header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
 
  /* Create gstreamer elements */
  pipeline = gst_pipeline_new ("audio-recorder");
  source = gst_element_factory_make ("autoaudiosrc", "audiosource");
  filter = gst_element_factory_make ("capsfilter", "filter");
  conv = gst_element_factory_make ("audioconvert", "converter");
  resampler = gst_element_factory_make ("audioresample", "resampler");
  encoder = gst_element_factory_make ("opusenc", "opus-encoder");
  oggmux = gst_element_factory_make ("oggmux", "ogg-muxer");
  sink = gst_element_factory_make ("appsink", "audio-output");
 
  if (! pipeline || ! filter || ! source || ! conv || ! resampler ||
      ! encoder || ! oggmux || ! sink)
  {
    GNUNET_log (GNUNET_ERROR_TYPE_ERROR,
                "One element could not be created. Exiting.\n");
    return -1;
  }
 
  g_signal_connect (source, "child-added", G_CALLBACK (source_child_added),
                    NULL);
 
  /* Set up the pipeline */
 
  caps = gst_caps_new_simple ("audio/x-raw",
                              "format", G_TYPE_STRING, "S16LE",
/*    "rate", G_TYPE_INT, SAMPLING_RATE,*/
                              "channels", G_TYPE_INT, OPUS_CHANNELS,
/*    "layout", G_TYPE_STRING, "interleaved",*/
                              NULL);
  g_object_set (G_OBJECT (filter),
                "caps", caps,
                NULL);
  gst_caps_unref (caps);
 
  g_object_set (G_OBJECT (encoder),
/*      "bitrate", 64000, */
/*      "bandwidth", OPUS_BANDWIDTH_FULLBAND, */
                "inband-fec", INBAND_FEC_MODE,
                "packet-loss-percentage", PACKET_LOSS_PERCENTAGE,
                "max-payload-size", MAX_PAYLOAD_SIZE,
                "audio", FALSE, /* VoIP, not audio */
                "frame-size", OPUS_FRAME_SIZE,
                NULL);
 
  g_object_set (G_OBJECT (oggmux),
                "max-delay", OGG_MAX_DELAY,
                "max-page-delay", OGG_MAX_PAGE_DELAY,
                NULL);
 
  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  bus_watch_id = gst_bus_add_watch (bus, bus_call, pipeline);
  gst_object_unref (bus);
 
  /* we add all elements into the pipeline */
  /* audiosource | converter | resampler | opus-encoder | audio-output */
  gst_bin_add_many (GST_BIN (pipeline), source, filter, conv, resampler,
                    encoder,
                    oggmux, sink, NULL);
 
  /* we link the elements together */
  gst_element_link_many (source, filter, conv, resampler, encoder, oggmux, sink,
                         NULL);
 
  /* Set the pipeline to "playing" state*/
  GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Now playing\n");
  gst_element_set_state (pipeline, GST_STATE_PLAYING);
 
 
  GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Running...\n");
  /* Iterate */
  while (! abort_send)
  {
    GstSample *s;
    GstBuffer *b;
    GstMapInfo m;
    size_t len, msg_size;
    const char *ptr;
    int phase;
 
    GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "pulling...\n");
    s = gst_app_sink_pull_sample (GST_APP_SINK (sink));
    if (NULL == s)
    {
      GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "pulled NULL\n");
      break;
    }
    GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "...pulled!\n");
    {
      const GstStructure *si;
      char *si_str;
      GstCaps *s_caps;
      char *caps_str;
      si = gst_sample_get_info (s);
      if (si)
      {
        si_str = gst_structure_to_string (si);
        if (si_str)
        {
          GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Got sample %s\n", si_str);
          g_free (si_str);
        }
      }
      else
        GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Got sample with no info\n");
      s_caps = gst_sample_get_caps (s);
      if (s_caps)
      {
        caps_str = gst_caps_to_string (s_caps);
        if (caps_str)
        {
          GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Got sample with caps %s\n",
                      caps_str);
          g_free (caps_str);
        }
      }
      else
        GNUNET_log (GNUNET_ERROR_TYPE_DEBUG, "Got sample with no caps\n");
    }
    b = gst_sample_get_buffer (s);
    if ((NULL == b) || ! gst_buffer_map (b, &m, GST_MAP_READ))
    {
      GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
                  "got NULL buffer %p or failed to map the buffer\n", b);
      gst_sample_unref (s);
      continue;
    }
 
    len = m.size;
    if (len > UINT16_MAX - sizeof(struct AudioMessage))
    {
      GNUNET_break (0);
      len = UINT16_MAX - sizeof(struct AudioMessage);
    }
    msg_size = sizeof(struct AudioMessage) + len;
    audio_message.header.size = htons ((uint16_t) msg_size);
 
    GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
                "Sending %u bytes of audio data\n", (unsigned int) msg_size);
    for (phase = 0; phase < 2; phase++)
    {
      size_t offset;
      size_t to_send;
      ssize_t ret;
      if (0 == phase)
      {
#ifdef DEBUG_RECORD_PURE_OGG
        if (dump_pure_ogg)
          continue;
#endif
        ptr = (const char *) &audio_message;
        to_send = sizeof(audio_message);
      }
      else
      {
        ptr = (const char *) m.data;
        to_send = len;
      }
      GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
                  "Sending %u bytes on phase %d\n", (unsigned int) to_send,
                  phase);
      for (offset = 0; offset < to_send; offset += ret)
      {
        ret = write (1, &ptr[offset], to_send - offset);
        if (0 >= ret)
        {
          if (-1 == ret)
            GNUNET_log (GNUNET_ERROR_TYPE_DEBUG,
                        "Failed to write %u bytes at offset %u (total %u) in phase %d: %s\n",
                        (unsigned int) (to_send - offset),
                        (unsigned int) offset,
                        (unsigned int) (to_send + offset),
                        phase,
                        strerror (errno));
          abort_send = 1;
          break;
        }
      }
      if (abort_send)
        break;
    }
    gst_buffer_unmap (b, &m);
    gst_sample_unref (s);
  }
 
  signal (SIGINT, inthandler);
  signal (SIGINT, termhandler);
 
  GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Returned, stopping playback\n");
  quit ();
 
  GNUNET_log (GNUNET_ERROR_TYPE_INFO, "Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));
  pipeline = NULL;
  g_source_remove (bus_watch_id);
 
  return 0;
}

References audio_message, bus_call(), conv, dump_pure_ogg, filter, getenv(), GNUNET_assert, GNUNET_break, GNUNET_ERROR_TYPE_DEBUG, GNUNET_ERROR_TYPE_ERROR, GNUNET_ERROR_TYPE_INFO, GNUNET_log, GNUNET_log_setup(), GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO, GNUNET_OK, gst_element_factory_make, AudioMessage::header, INBAND_FEC_MODE, m, MAX_PAYLOAD_SIZE, OGG_MAX_DELAY, OGG_MAX_PAGE_DELAY, OPUS_CHANNELS, OPUS_FRAME_SIZE, PACKET_LOSS_PERCENTAGE, phase, pipeline, quit(), resampler, ret, signalhandler(), sink, GNUNET_MessageHeader::size, source, source_child_added(), and GNUNET_MessageHeader::type.

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Variable Documentation

pipeline

GstElement* pipeline

static

Main pipeline.

Definition at line 95 of file gnunet-helper-audio-record-gst.c.

Referenced by main(), and quit().

dump_pure_ogg

int dump_pure_ogg

static

Definition at line 98 of file gnunet-helper-audio-record-gst.c.

Referenced by main().