pulsesrc.c 23.3 KB
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/*
 *  GStreamer pulseaudio plugin
 *
 *  Copyright (c) 2004-2008 Lennart Poettering
 *
 *  gst-pulse is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU Lesser General Public License as
 *  published by the Free Software Foundation; either version 2.1 of the
 *  License, or (at your option) any later version.
 *
 *  gst-pulse is distributed in the hope that it will be useful, but
 *  WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
 *  Lesser General Public License for more details.
 *
 *  You should have received a copy of the GNU Lesser General Public
 *  License along with gst-pulse; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
 *  USA.
 */

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/**
 * SECTION:element-pulsesrc
 * @short_description: Capture audio from a PulseAudio sound server
 * @see_also: pulsesink, pulsemixer
 *
 * <refsect2>
 * <para>
 * This element captures audio from a PulseAudio sound server.
 * </para>
 * <title>Example pipelines</title>
 * <para>
 * <programlisting>
 * gst-launch -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
 * </programlisting>
 * Record from a sound card using ALSA and encode to Ogg/Vorbis.
 * </para>
 * </refsect2>
 */

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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <string.h>
#include <stdio.h>

#include <gst/base/gstbasesrc.h>
#include <gst/gsttaglist.h>

#include "pulsesrc.h"
#include "pulseutil.h"
#include "pulsemixerctrl.h"

GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug

enum
{
  PROP_SERVER = 1,
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  PROP_DEVICE,
  PROP_DEVICE_NAME
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};

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static void gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc);
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static void gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc);
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static void gst_pulsesrc_set_property (GObject * object, guint prop_id,
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    const GValue * value, GParamSpec * pspec);
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static void gst_pulsesrc_get_property (GObject * object, guint prop_id,
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    GValue * value, GParamSpec * pspec);
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static void gst_pulsesrc_finalize (GObject * object);
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static void gst_pulsesrc_dispose (GObject * object);
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static gboolean gst_pulsesrc_open (GstAudioSrc * asrc);
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static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
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static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
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    GstRingBufferSpec * spec);
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static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
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static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
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    guint length);
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static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
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static gboolean gst_pulsesrc_negotiate (GstBaseSrc * basesrc);

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static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
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    element, GstStateChange transition);

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static void gst_pulsesrc_init_interfaces (GType type);

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#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ENDIANNESS   "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ENDIANNESS   "BIG_ENDIAN, LITTLE_ENDIAN"
#endif

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GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS (GstPulseSrc, gst_pulsesrc);
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GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSrc, gst_pulsesrc);
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GST_BOILERPLATE_FULL (GstPulseSrc, gst_pulsesrc, GstAudioSrc,
    GST_TYPE_AUDIO_SRC, gst_pulsesrc_init_interfaces);
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static gboolean
gst_pulsesrc_interface_supported (GstImplementsInterface *
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    iface, GType interface_type)
{
  GstPulseSrc *this = GST_PULSESRC (iface);

  if (interface_type == GST_TYPE_MIXER && this->mixer)
    return TRUE;

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  if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
    return TRUE;

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  return FALSE;
}

static void
gst_pulsesrc_implements_interface_init (GstImplementsInterfaceClass * klass)
{
  klass->supported = gst_pulsesrc_interface_supported;
}

static void
gst_pulsesrc_init_interfaces (GType type)
{
  static const GInterfaceInfo implements_iface_info = {
    (GInterfaceInitFunc) gst_pulsesrc_implements_interface_init,
    NULL,
    NULL,
  };
  static const GInterfaceInfo mixer_iface_info = {
    (GInterfaceInitFunc) gst_pulsesrc_mixer_interface_init,
    NULL,
    NULL,
  };
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  static const GInterfaceInfo probe_iface_info = {
    (GInterfaceInitFunc) gst_pulsesrc_property_probe_interface_init,
    NULL,
    NULL,
  };
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  g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
      &implements_iface_info);
  g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
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  g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
      &probe_iface_info);
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}

static void
gst_pulsesrc_base_init (gpointer g_class)
{

  static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
      GST_PAD_SRC,
      GST_PAD_ALWAYS,
      GST_STATIC_CAPS ("audio/x-raw-int, "
          "endianness = (int) { " ENDIANNESS " }, "
          "signed = (boolean) TRUE, "
          "width = (int) 16, "
          "depth = (int) 16, "
          "rate = (int) [ 1, MAX ], "
          "channels = (int) [ 1, 16 ];"
          "audio/x-raw-int, "
          "endianness = (int) { " ENDIANNESS " }, "
          "signed = (boolean) TRUE, "
          "width = (int) 32, "
          "depth = (int) 32, "
          "rate = (int) [ 1, MAX ], "
          "channels = (int) [ 1, 16 ];"
          "audio/x-raw-float, "
          "endianness = (int) { " ENDIANNESS " }, "
          "width = (int) 32, "
          "rate = (int) [ 1, MAX ], "
          "channels = (int) [ 1, 16 ];"
          "audio/x-raw-int, "
          "signed = (boolean) FALSE, "
          "width = (int) 8, "
          "depth = (int) 8, "
          "rate = (int) [ 1, MAX ], "
          "channels = (int) [ 1, 16 ];"
          "audio/x-alaw, "
          "rate = (int) [ 1, MAX], "
          "channels = (int) [ 1, 16 ];"
          "audio/x-mulaw, "
          "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 16 ]")
      );

  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

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  gst_element_class_set_details_simple (element_class,
      "PulseAudio Audio Source",
      "Source/Audio",
      "Captures audio from a PulseAudio server", "Lennart Poettering");
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  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&pad_template));
}

static void
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gst_pulsesrc_class_init (GstPulseSrcClass * klass)
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{
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  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
  GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
  GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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  gstelement_class->change_state =
      GST_DEBUG_FUNCPTR (gst_pulsesrc_change_state);

  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pulsesrc_dispose);
  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesrc_finalize);
  gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_set_property);
  gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_get_property);

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  gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_pulsesrc_negotiate);

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  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_pulsesrc_open);
  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_pulsesrc_close);
  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_prepare);
  gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_unprepare);
  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_pulsesrc_read);
  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesrc_delay);

  /* Overwrite GObject fields */
  g_object_class_install_property (gobject_class,
      PROP_SERVER,
      g_param_spec_string ("server", "Server",
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          "The PulseAudio server to connect to", NULL,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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  g_object_class_install_property (gobject_class, PROP_DEVICE,
      g_param_spec_string ("device", "Source",
          "The PulseAudio source device to connect to", NULL,
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          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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  g_object_class_install_property (gobject_class,
      PROP_DEVICE_NAME,
      g_param_spec_string ("device-name", "Device name",
          "Human-readable name of the sound device", NULL,
          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}

static void
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gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass)
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{
  int e;

  pulsesrc->server = pulsesrc->device = NULL;

  pulsesrc->context = NULL;
  pulsesrc->stream = NULL;

  pulsesrc->read_buffer = NULL;
  pulsesrc->read_buffer_length = 0;

  pulsesrc->mainloop = pa_threaded_mainloop_new ();
  g_assert (pulsesrc->mainloop);

  e = pa_threaded_mainloop_start (pulsesrc->mainloop);
  g_assert (e == 0);

  pulsesrc->mixer = NULL;
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  pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->device, FALSE, TRUE);        /* FALSE for sinks, TRUE for sources */
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}

static void
gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc)
{
  if (pulsesrc->stream) {
    pa_stream_disconnect (pulsesrc->stream);
    pa_stream_unref (pulsesrc->stream);
    pulsesrc->stream = NULL;
  }
}

static void
gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
{

  gst_pulsesrc_destroy_stream (pulsesrc);

  if (pulsesrc->context) {
    pa_context_disconnect (pulsesrc->context);
    pa_context_unref (pulsesrc->context);
    pulsesrc->context = NULL;
  }
}

static void
gst_pulsesrc_finalize (GObject * object)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (object);

  pa_threaded_mainloop_stop (pulsesrc->mainloop);

  gst_pulsesrc_destroy_context (pulsesrc);

  g_free (pulsesrc->server);
  g_free (pulsesrc->device);

  pa_threaded_mainloop_free (pulsesrc->mainloop);

  if (pulsesrc->mixer)
    gst_pulsemixer_ctrl_free (pulsesrc->mixer);

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  if (pulsesrc->probe) {
    gst_pulseprobe_free (pulsesrc->probe);
    pulsesrc->probe = NULL;
  }

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  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static void
gst_pulsesrc_dispose (GObject * object)
{
  G_OBJECT_CLASS (parent_class)->dispose (object);
}

static void
gst_pulsesrc_set_property (GObject * object,
    guint prop_id, const GValue * value, GParamSpec * pspec)
{

  GstPulseSrc *pulsesrc = GST_PULSESRC (object);

  switch (prop_id) {
    case PROP_SERVER:
      g_free (pulsesrc->server);
      pulsesrc->server = g_value_dup_string (value);
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      if (pulsesrc->probe)
        gst_pulseprobe_set_server (pulsesrc->probe, pulsesrc->server);

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      break;

    case PROP_DEVICE:
      g_free (pulsesrc->device);
      pulsesrc->device = g_value_dup_string (value);
      break;

    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_pulsesrc_get_property (GObject * object,
    guint prop_id, GValue * value, GParamSpec * pspec)
{

  GstPulseSrc *pulsesrc = GST_PULSESRC (object);

  switch (prop_id) {
    case PROP_SERVER:
      g_value_set_string (value, pulsesrc->server);
      break;

    case PROP_DEVICE:
      g_value_set_string (value, pulsesrc->device);
      break;

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    case PROP_DEVICE_NAME:

      if (pulsesrc->mixer)
        g_value_set_string (value, pulsesrc->mixer->description);
      else
        g_value_set_string (value, NULL);

      break;

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    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_pulsesrc_context_state_cb (pa_context * c, void *userdata)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);

  switch (pa_context_get_state (c)) {
    case PA_CONTEXT_READY:
    case PA_CONTEXT_TERMINATED:
    case PA_CONTEXT_FAILED:
      pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
      break;

    case PA_CONTEXT_UNCONNECTED:
    case PA_CONTEXT_CONNECTING:
    case PA_CONTEXT_AUTHORIZING:
    case PA_CONTEXT_SETTING_NAME:
      break;
  }
}

static void
gst_pulsesrc_stream_state_cb (pa_stream * s, void *userdata)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);

  switch (pa_stream_get_state (s)) {

    case PA_STREAM_READY:
    case PA_STREAM_FAILED:
    case PA_STREAM_TERMINATED:
      pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
      break;

    case PA_STREAM_UNCONNECTED:
    case PA_STREAM_CREATING:
      break;
  }
}

static void
gst_pulsesrc_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);

  pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}

static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  gchar *name = gst_pulse_client_name ();

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  if (!(pulsesrc->context =
          pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
              name))) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
        (NULL));
    goto unlock_and_fail;
  }

  pa_context_set_state_callback (pulsesrc->context,
      gst_pulsesrc_context_state_cb, pulsesrc);

  if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  /* Wait until the context is ready */
  pa_threaded_mainloop_wait (pulsesrc->mainloop);

  if (pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  g_free (name);
  return TRUE;

unlock_and_fail:

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  g_free (name);
  return FALSE;
}

static gboolean
gst_pulsesrc_close (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  pa_threaded_mainloop_lock (pulsesrc->mainloop);
  gst_pulsesrc_destroy_context (pulsesrc);
  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;
}

static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  pa_threaded_mainloop_lock (pulsesrc->mainloop);
  gst_pulsesrc_destroy_stream (pulsesrc);

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  pulsesrc->read_buffer = NULL;
  pulsesrc->read_buffer_length = 0;

  return TRUE;
}

#define CHECK_DEAD_GOTO(pulsesrc, label) \
if (!(pulsesrc)->context || pa_context_get_state((pulsesrc)->context) != PA_CONTEXT_READY || \
    !(pulsesrc)->stream || pa_stream_get_state((pulsesrc)->stream) != PA_STREAM_READY) { \
    GST_ELEMENT_ERROR((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s", (pulsesrc)->context ? pa_strerror(pa_context_errno((pulsesrc)->context)) : NULL), (NULL)); \
    goto label; \
}

static guint
gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  size_t sum = 0;

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);

  while (length > 0) {
    size_t l;

    if (!pulsesrc->read_buffer) {

      for (;;) {
        if (pa_stream_peek (pulsesrc->stream, &pulsesrc->read_buffer,
                &pulsesrc->read_buffer_length) < 0) {
          GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
              ("pa_stream_peek() failed: %s",
                  pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
          goto unlock_and_fail;
        }

        if (pulsesrc->read_buffer)
          break;

        pa_threaded_mainloop_wait (pulsesrc->mainloop);

        CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);
      }
    }

    g_assert (pulsesrc->read_buffer && pulsesrc->read_buffer_length);

    l = pulsesrc->read_buffer_length >
        length ? length : pulsesrc->read_buffer_length;

    memcpy (data, pulsesrc->read_buffer, l);

    pulsesrc->read_buffer = (const guint8 *) pulsesrc->read_buffer + l;
    pulsesrc->read_buffer_length -= l;

    data = (guint8 *) data + l;
    length -= l;

    sum += l;

    if (pulsesrc->read_buffer_length <= 0) {

      if (pa_stream_drop (pulsesrc->stream) < 0) {
        GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
            ("pa_stream_drop() failed: %s",
                pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
        goto unlock_and_fail;
      }

      pulsesrc->read_buffer = NULL;
      pulsesrc->read_buffer_length = 0;
    }
  }

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return sum;

unlock_and_fail:
  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
  return 0;
}

static guint
gst_pulsesrc_delay (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  pa_usec_t t;

  int negative;

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);

  if (pa_stream_get_latency (pulsesrc->stream, &t, &negative) < 0) {

    if (pa_context_errno (pulsesrc->context) != PA_ERR_NODATA) {
      GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
          ("pa_stream_get_latency() failed: %s",
              pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
      goto unlock_and_fail;
    }

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    GST_WARNING_OBJECT (pulsesrc, "Not data while querying latency");
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    t = 0;
  } else if (negative)
    t = 0;

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);

unlock_and_fail:

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
  return 0;
}

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static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps * caps)
{
  pa_channel_map channel_map;
  GstStructure *s;
  gboolean need_channel_layout = FALSE;
  GstRingBufferSpec spec;

  memset (&spec, 0, sizeof (GstRingBufferSpec));
  spec.latency_time = GST_SECOND;
  if (!gst_ring_buffer_parse_caps (&spec, caps)) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Can't parse caps."), (NULL));
    goto fail;
  }
  /* Keep the refcount of the caps at 1 to make them writable */
  gst_caps_unref (spec.caps);

  if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec)) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Invalid sample specification."), (NULL));
    goto fail;
  }

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  if (!pulsesrc->context
      || pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context state: %s",
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            pulsesrc->
            context ? pa_strerror (pa_context_errno (pulsesrc->context)) :
            NULL), (NULL));
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    goto unlock_and_fail;
  }

  s = gst_caps_get_structure (caps, 0);
  if (!gst_structure_has_field (s, "channel-layout") ||
      !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
    if (spec.channels == 1)
      pa_channel_map_init_mono (&channel_map);
    else if (spec.channels == 2)
      pa_channel_map_init_stereo (&channel_map);
    else
      need_channel_layout = TRUE;
  }

  if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
              "Record Stream",
              &pulsesrc->sample_spec,
              (need_channel_layout) ? NULL : &channel_map))) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to create stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  if (need_channel_layout) {
    const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);

    gst_pulse_channel_map_to_gst (m, &spec);
    caps = spec.caps;
  }

  GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, caps);

  pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
      pulsesrc);
  pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
      pulsesrc);

  return TRUE;

unlock_and_fail:
  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

fail:
  return FALSE;
}

/* This is essentially gst_base_src_negotiate_default() but the caps
 * are guaranteed to have a channel layout for > 2 channels
 */
static gboolean
gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
{
  GstCaps *thiscaps;
  GstCaps *caps = NULL;
  GstCaps *peercaps = NULL;
  gboolean result = FALSE;

  /* first see what is possible on our source pad */
  thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc));
  GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
  /* nothing or anything is allowed, we're done */
  if (thiscaps == NULL || gst_caps_is_any (thiscaps))
    goto no_nego_needed;

  /* get the peer caps */
  peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
  GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
  if (peercaps) {
    GstCaps *icaps;

    /* get intersection */
    icaps = gst_caps_intersect (thiscaps, peercaps);
    GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, icaps);
    gst_caps_unref (thiscaps);
    gst_caps_unref (peercaps);
    if (icaps) {
      /* take first (and best, since they are sorted) possibility */
      caps = gst_caps_copy_nth (icaps, 0);
      gst_caps_unref (icaps);
    }
  } else {
    /* no peer, work with our own caps then */
    caps = thiscaps;
  }
  if (caps) {
    caps = gst_caps_make_writable (caps);
    gst_caps_truncate (caps);

    /* now fixate */
    if (!gst_caps_is_empty (caps)) {
      gst_pad_fixate_caps (GST_BASE_SRC_PAD (basesrc), caps);
      GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);

      if (gst_caps_is_any (caps)) {
        /* hmm, still anything, so element can do anything and
         * nego is not needed */
        result = TRUE;
      } else if (gst_caps_is_fixed (caps)) {
        /* yay, fixed caps, use those then */
        result = gst_pulsesrc_create_stream (GST_PULSESRC (basesrc), caps);
        if (result)
          gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
        result = TRUE;
      }
    }
    gst_caps_unref (caps);
  }
  return result;

no_nego_needed:
  {
    GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
    if (thiscaps)
      gst_caps_unref (thiscaps);
    return TRUE;
  }
}

static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
  pa_buffer_attr buf_attr;
  GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);

  memset (&buf_attr, 0, sizeof (buf_attr));
  buf_attr.maxlength = spec->segtotal * spec->segsize * 2;
  buf_attr.fragsize = spec->segsize;

  if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &buf_attr,
          PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
          PA_STREAM_NOT_MONOTONOUS) < 0) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to connect stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  /* Wait until the stream is ready */
  pa_threaded_mainloop_wait (pulsesrc->mainloop);

  if (pa_stream_get_state (pulsesrc->stream) != PA_STREAM_READY) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to connect stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;

unlock_and_fail:
  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
  return FALSE;
}

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static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
  GstPulseSrc *this = GST_PULSESRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:

      if (!this->mixer)
        this->mixer =
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            gst_pulsemixer_ctrl_new (G_OBJECT (this), this->server,
            this->device, GST_PULSEMIXER_SOURCE);
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      break;

    case GST_STATE_CHANGE_READY_TO_NULL:

      if (this->mixer) {
        gst_pulsemixer_ctrl_free (this->mixer);
        this->mixer = NULL;
      }

      break;

    default:
      ;
  }

  if (GST_ELEMENT_CLASS (parent_class)->change_state)
    return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  return GST_STATE_CHANGE_SUCCESS;
}