pulseaudio: require pulseaudio >= 1.0

diff --git a/configure.ac b/configure.ac
index 9a2409c..eee3cb7 100644 (file)
--- a/configure.ac
+++ b/configure.ac
@@ -806,15 +806,7 @@ AG_GST_CHECK_FEATURE(PULSE, [pulseaudio plug-in], pulseaudio, [
   dnl used in ext/pulse/pulseutil.c
   AC_CHECK_HEADERS([process.h])
 
-  AG_GST_PKG_CHECK_MODULES(PULSE, libpulse >= 0.9.16)
-  AG_GST_PKG_CHECK_MODULES(PULSE_0_9_20, libpulse >= 0.9.20)
-  if test x$HAVE_PULSE_0_9_20 = xyes; then
-    AC_DEFINE(HAVE_PULSE_0_9_20, 1, [defined if pulseaudio >= 0.9.20 is available])
-  fi
-  AG_GST_PKG_CHECK_MODULES(PULSE_1_0, libpulse >= 0.98)
-  if test x$HAVE_PULSE_1_0 = xyes; then
-    AC_DEFINE(HAVE_PULSE_1_0, 1, [defined if pulseaudio >= 1.0 is available])
-  fi
+  AG_GST_PKG_CHECK_MODULES(PULSE, libpulse >= 1.0)
 ])
 
 dnl *** dv1394 ***

diff --git a/ext/pulse/plugin.c b/ext/pulse/plugin.c
index 6b2e6b4..25fc95a 100644 (file)
--- a/ext/pulse/plugin.c
+++ b/ext/pulse/plugin.c
@@ -49,11 +49,11 @@ plugin_init (GstPlugin * plugin)
           GST_TYPE_PULSESRC))
     return FALSE;
 
-#ifdef HAVE_PULSE_1_0
+  /* FIXME 0.11: this helper bin sink should just go away, reconfiguration
+   * should be handled using reconfigure events */
   if (!gst_element_register (plugin, "pulseaudiosink", GST_RANK_PRIMARY + 11,
           GST_TYPE_PULSE_AUDIO_SINK))
     return FALSE;
-#endif
 
   if (!gst_element_register (plugin, "pulsemixer", GST_RANK_NONE,
           GST_TYPE_PULSEMIXER))

diff --git a/ext/pulse/pulseaudiosink.c b/ext/pulse/pulseaudiosink.c
index e5e9ee4..adaa95e 100644 (file)
--- a/ext/pulse/pulseaudiosink.c
+++ b/ext/pulse/pulseaudiosink.c
@@ -49,7 +49,7 @@
 #include "config.h"
 #endif
 
-#ifdef HAVE_PULSE_1_0
+/* FIXME 0.11: pulseaudiosink helper bin must die */
 
 #include <gst/pbutils/pbutils.h>
 #include <gst/gst-i18n-plugin.h>
@@ -958,5 +958,3 @@ gst_pulse_audio_sink_change_state (GstElement * element,
 out:
   return ret;
 }
-
-#endif /* HAVE_PULSE_1_0 */

diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c
index 61da442..2a58b2f 100644 (file)
--- a/ext/pulse/pulsesink.c
+++ b/ext/pulse/pulsesink.c
@@ -139,13 +139,9 @@ struct _GstPulseRingBuffer
   pa_context *context;
   pa_stream *stream;
 
-#ifdef HAVE_PULSE_1_0
   pa_format_info *format;
   guint channels;
   gboolean is_pcm;
-#else
-  pa_sample_spec sample_spec;
-#endif
 
   void *m_data;
   size_t m_towrite;
@@ -230,13 +226,9 @@ gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf)
   pbuf->context = NULL;
   pbuf->stream = NULL;
 
-#ifdef HAVE_PULSE_1_0
   pbuf->format = NULL;
   pbuf->channels = 0;
   pbuf->is_pcm = FALSE;
-#else
-  pa_sample_spec_init (&pbuf->sample_spec);
-#endif
 
   pbuf->m_data = NULL;
   pbuf->m_towrite = 0;
@@ -265,14 +257,12 @@ gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
       pbuf->m_offset = 0;
       pbuf->m_lastoffset = 0;
     }
-#ifdef HAVE_PULSE_1_0
     if (pbuf->format) {
       pa_format_info_free (pbuf->format);
       pbuf->format = NULL;
       pbuf->channels = 0;
       pbuf->is_pcm = FALSE;
     }
-#endif
 
     pa_stream_disconnect (pbuf->stream);
 
@@ -424,7 +414,6 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
     if (idx != pa_stream_get_index (pbuf->stream))
       continue;
 
-#ifdef HAVE_PULSE_1_0
     if (psink->device && pbuf->is_pcm &&
         !g_str_equal (psink->device,
             pa_stream_get_device_name (pbuf->stream))) {
@@ -443,7 +432,6 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
       if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
         GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
     }
-#endif
 
     /* Actually this event is also triggered when other properties of
      * the stream change that are unrelated to the volume. However it is
@@ -738,7 +726,6 @@ gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
     gst_element_post_message (GST_ELEMENT_CAST (psink),
         gst_message_new_request_state (GST_OBJECT_CAST (psink),
             GST_STATE_PLAYING));
-#ifdef HAVE_PULSE_1_0
   } else if (!strcmp (name, PA_STREAM_EVENT_FORMAT_LOST)) {
     GstEvent *renego;
 
@@ -763,7 +750,6 @@ gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
       GST_ELEMENT_ERROR (psink, STREAM, FORMAT, ("Sink format changed"),
           ("Sink format changed"));
     }
-#endif
   } else {
     GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
   }
@@ -804,33 +790,24 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
   const pa_buffer_attr *actual;
   pa_channel_map channel_map;
   pa_operation *o = NULL;
-#ifdef HAVE_PULSE_0_9_20
   pa_cvolume v;
-#endif
   pa_cvolume *pv = NULL;
   pa_stream_flags_t flags;
   const gchar *name;
   GstAudioClock *clock;
-#ifdef HAVE_PULSE_1_0
   pa_format_info *formats[1];
 #ifndef GST_DISABLE_GST_DEBUG
   gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
 #endif
-#endif
 
   psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
   pbuf = GST_PULSERING_BUFFER_CAST (buf);
 
   GST_LOG_OBJECT (psink, "creating sample spec");
   /* convert the gstreamer sample spec to the pulseaudio format */
-#ifdef HAVE_PULSE_1_0
   if (!gst_pulse_fill_format_info (spec, &pbuf->format, &pbuf->channels))
     goto invalid_spec;
   pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
-#else
-  if (!gst_pulse_fill_sample_spec (spec, &pbuf->sample_spec))
-    goto invalid_spec;
-#endif
 
   pa_threaded_mainloop_lock (mainloop);
 
@@ -847,12 +824,8 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
   pa_operation_unref (o);
 
   /* initialize the channel map */
-#ifdef HAVE_PULSE_1_0
   if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
     pa_format_info_set_channel_map (pbuf->format, &channel_map);
-#else
-  gst_pulse_gst_to_channel_map (&channel_map, spec);
-#endif
 
   /* find a good name for the stream */
   if (psink->stream_name)
@@ -861,17 +834,10 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
     name = "Playback Stream";
 
   /* create a stream */
-#ifdef HAVE_PULSE_1_0
   formats[0] = pbuf->format;
   if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
               psink->proplist)))
     goto stream_failed;
-#else
-  GST_LOG_OBJECT (psink, "creating stream with name %s", name);
-  if (!(pbuf->stream = pa_stream_new_with_proplist (pbuf->context, name,
-              &pbuf->sample_spec, &channel_map, psink->proplist)))
-    goto stream_failed;
-#endif
 
   /* install essential callbacks */
   pa_stream_set_state_callback (pbuf->stream,
@@ -904,26 +870,19 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
   GST_INFO_OBJECT (psink, "prebuf:    %d", wanted.prebuf);
   GST_INFO_OBJECT (psink, "minreq:    %d", wanted.minreq);
 
-#ifdef HAVE_PULSE_0_9_20
   /* configure volume when we changed it, else we leave the default */
   if (psink->volume_set) {
     GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
     pv = &v;
-#ifdef HAVE_PULSE_1_0
     if (pbuf->is_pcm)
       gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
     else {
       GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
       pv = NULL;
     }
-#else
-    gst_pulse_cvolume_from_linear (pv, pbuf->sample_spec.channels,
-        psink->volume);
-#endif
   } else {
     pv = NULL;
   }
-#endif
 
   /* construct the flags */
   flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
@@ -949,7 +908,6 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
   if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
     goto connect_failed;
 
-#ifdef HAVE_PULSE_1_0
   g_free (psink->device);
   psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
 
@@ -958,7 +916,6 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
       pa_stream_get_format_info (pbuf->stream));
   GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
 #endif
-#endif
 
   /* After we passed the volume off of to PA we never want to set it
      again, since it is PA's job to save/restore volumes.  */
@@ -1032,7 +989,6 @@ gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
   gst_pulsering_destroy_stream (pbuf);
   pa_threaded_mainloop_unlock (mainloop);
 
-#ifdef HAVE_PULSE_1_0
   {
     GstPulseSink *psink;
 
@@ -1040,7 +996,6 @@ gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
     g_atomic_int_set (&psink->format_lost, FALSE);
     psink->format_lost_time = GST_CLOCK_TIME_NONE;
   }
-#endif
 
   return TRUE;
 }
@@ -1063,12 +1018,10 @@ gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
 
   psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
 
-#ifdef HAVE_PULSE_1_0
   if (g_atomic_int_get (&psink->format_lost)) {
     /* Sink format changed, stream's gone so fake being paused */
     return TRUE;
   }
-#endif
 
   GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
   if (pbuf->corked != corked) {
@@ -1251,13 +1204,11 @@ gst_pulseringbuffer_stop (GstAudioRingBuffer * buf)
     GST_DEBUG_OBJECT (psink, "signal commit thread");
     pa_threaded_mainloop_signal (mainloop, 0);
   }
-#ifdef HAVE_PULSE_1_0
   if (g_atomic_int_get (&psink->format_lost)) {
     /* Don't try to flush, the stream's probably gone by now */
     res = TRUE;
     goto cleanup;
   }
-#endif
 
   /* then try to flush, it's not fatal when this fails */
   GST_DEBUG_OBJECT (psink, "flushing");
@@ -1428,12 +1379,10 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
    * needed to properly handle reverse playback: it points to the last sample. */
   data_end = data + (bpf * inr);
 
-#ifdef HAVE_PULSE_1_0
   if (g_atomic_int_get (&psink->format_lost)) {
     /* Sink format changed, drop the data and hope upstream renegotiates */
     goto fake_done;
   }
-#endif
 
   if (pbuf->paused)
     goto was_paused;
@@ -1482,12 +1431,10 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
       for (;;) {
         pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
 
-#ifdef HAVE_PULSE_1_0
         if (g_atomic_int_get (&psink->format_lost)) {
           /* Sink format changed, give up and hope upstream renegotiates */
           goto fake_done;
         }
-#endif
 
         if (pbuf->m_writable == (size_t) - 1)
           goto writable_size_failed;
@@ -1538,13 +1485,11 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
     GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
         (guint) avail, offset);
 
-#ifdef HAVE_PULSE_1_0
     /* No trick modes for passthrough streams */
     if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
       GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
       goto unlock_and_fail;
     }
-#endif
 
     if (G_LIKELY (inr == outr && !reverse)) {
       /* no rate conversion, simply write out the samples */
@@ -1626,9 +1571,7 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
     }
   }
 
-#ifdef HAVE_PULSE_1_0
 fake_done:
-#endif
   /* we consumed all samples here */
   data = data_end + bpf;
 
@@ -1931,13 +1874,11 @@ gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
   pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
   psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
 
-#ifdef HAVE_PULSE_1_0
   if (g_atomic_int_get (&psink->format_lost)) {
     /* Stream was lost in a format change, it'll get set up again once
      * upstream renegotiates */
     return psink->format_lost_time;
   }
-#endif
 
   pa_threaded_mainloop_lock (mainloop);
   if (gst_pulsering_is_dead (psink, pbuf, TRUE))
@@ -1973,10 +1914,8 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
 {
   GstPulseRingBuffer *pbuf;
   GstPulseSink *psink;
-#ifdef HAVE_PULSE_1_0
   GList *l;
   guint8 j;
-#endif
 
   pbuf = GST_PULSERING_BUFFER_CAST (userdata);
   psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
@@ -1987,7 +1926,6 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
   g_free (psink->device_description);
   psink->device_description = g_strdup (i->description);
 
-#ifdef HAVE_PULSE_1_0
   g_mutex_lock (psink->sink_formats_lock);
 
   for (l = g_list_first (psink->sink_formats); l; l = g_list_next (l))
@@ -2001,13 +1939,11 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
         pa_format_info_copy (i->formats[j]));
 
   g_mutex_unlock (psink->sink_formats_lock);
-#endif
 
 done:
   pa_threaded_mainloop_signal (mainloop, 0);
 }
 
-#ifdef HAVE_PULSE_1_0
 /* NOTE: If you're making changes here, see if pulseaudiosink acceptcaps also
  * needs to be changed accordingly. */
 static gboolean
@@ -2139,7 +2075,6 @@ info_failed:
     goto out;
   }
 }
-#endif
 
 static void
 gst_pulsesink_init (GstPulseSink * pulsesink)
@@ -2149,10 +2084,8 @@ gst_pulsesink_init (GstPulseSink * pulsesink)
   pulsesink->device_description = NULL;
   pulsesink->client_name = gst_pulse_client_name ();
 
-#ifdef HAVE_PULSE_1_0
   pulsesink->sink_formats_lock = g_mutex_new ();
   pulsesink->sink_formats = NULL;
-#endif
 
   pulsesink->volume = DEFAULT_VOLUME;
   pulsesink->volume_set = FALSE;
@@ -2162,10 +2095,8 @@ gst_pulsesink_init (GstPulseSink * pulsesink)
 
   pulsesink->notify = 0;
 
-#ifdef HAVE_PULSE_1_0
   g_atomic_int_set (&pulsesink->format_lost, FALSE);
   pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
-#endif
 
   pulsesink->properties = NULL;
   pulsesink->proplist = NULL;
@@ -2188,22 +2119,18 @@ static void
 gst_pulsesink_finalize (GObject * object)
 {
   GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
-#ifdef HAVE_PULSE_1_0
   GList *i;
-#endif
 
   g_free (pulsesink->server);
   g_free (pulsesink->device);
   g_free (pulsesink->device_description);
   g_free (pulsesink->client_name);
 
-#ifdef HAVE_PULSE_1_0
   for (i = g_list_first (pulsesink->sink_formats); i; i = g_list_next (i))
     pa_format_info_free ((pa_format_info *) i->data);
 
   g_list_free (pulsesink->sink_formats);
   g_mutex_free (pulsesink->sink_formats_lock);
-#endif
 
   if (pulsesink->properties)
     gst_structure_free (pulsesink->properties);
@@ -2240,16 +2167,12 @@ gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
   if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
     goto no_index;
 
-#ifdef HAVE_PULSE_1_0
   if (pbuf->is_pcm)
     gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
   else
     /* FIXME: this will eventually be superceded by checks to see if the volume
      * is readable/writable */
     goto unlock;
-#else
-  gst_pulse_cvolume_from_linear (&v, pbuf->sample_spec.channels, volume);
-#endif
 
   if (!(o = pa_context_set_sink_input_volume (pbuf->context, idx,
               &v, NULL, NULL)))
@@ -2874,7 +2797,6 @@ gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
 static gboolean
 gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
 {
-#ifdef HAVE_PULSE_1_0
   GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
   gboolean ret;
 
@@ -2894,9 +2816,6 @@ gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
       break;
   }
   return ret;
-#else
-  return GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
-#endif
 }
 
 static void

diff --git a/ext/pulse/pulsesink.h b/ext/pulse/pulsesink.h
index f6f4357..9a0c58e 100644 (file)
--- a/ext/pulse/pulsesink.h
+++ b/ext/pulse/pulsesink.h
@@ -77,12 +77,10 @@ struct _GstPulseSink
   GstStructure *properties;
   pa_proplist *proplist;
 
-#ifdef HAVE_PULSE_1_0
   GMutex *sink_formats_lock;
   GList *sink_formats;
   volatile gint format_lost;
   GstClockTime format_lost_time;
-#endif
 };
 
 struct _GstPulseSinkClass
@@ -111,7 +109,6 @@ GType gst_pulsesink_get_type (void);
     "audio/x-mulaw, " \
       "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ];"
 
-#ifdef HAVE_PULSE_1_0
 #define _PULSE_SINK_CAPS_1_0 \
     "audio/x-ac3, framed = (boolean) true;" \
     "audio/x-eac3, framed = (boolean) true; " \
@@ -119,15 +116,12 @@ GType gst_pulsesink_get_type (void);
       "block-size = (int) { 512, 1024, 2048 }; " \
     "audio/mpeg, mpegversion = (int) 1, " \
       "mpegaudioversion = (int) [ 1, 2 ], parsed = (boolean) true;"
-#else
-#define _PULSE_SINK_CAPS_1_0 ""
-#endif
 
 #define PULSE_SINK_TEMPLATE_CAPS \
   _PULSE_SINK_CAPS_COMMON \
   _PULSE_SINK_CAPS_1_0
 
-#ifdef HAVE_PULSE_1_0
+/* FIXME 0.11: pulseaudiosink helper bin must die */
 
 #define GST_TYPE_PULSE_AUDIO_SINK \
   (gst_pulse_audio_sink_get_type())
@@ -144,8 +138,6 @@ GType gst_pulsesink_get_type (void);
 
 GType gst_pulse_audio_sink_get_type (void);
 
-#endif /* HAVE_PULSE_1_0 */
-
 G_END_DECLS
 
 #endif /* __GST_PULSESINK_H__ */

diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c
index 7dddf9f..8281a7e 100644 (file)
--- a/ext/pulse/pulsesrc.c
+++ b/ext/pulse/pulsesrc.c
@@ -43,9 +43,7 @@
 
 #include <gst/base/gstbasesrc.h>
 #include <gst/gsttaglist.h>
-#ifdef HAVE_PULSE_1_0
 #include <gst/interfaces/streamvolume.h>
-#endif
 
 #include "pulsesrc.h"
 #include "pulseutil.h"
@@ -58,11 +56,9 @@ GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
 #define DEFAULT_DEVICE            NULL
 #define DEFAULT_DEVICE_NAME       NULL
 
-#ifdef HAVE_PULSE_1_0
 #define DEFAULT_VOLUME          1.0
 #define DEFAULT_MUTE            FALSE
 #define MAX_VOLUME              10.0
-#endif
 
 enum
 {
@@ -73,10 +69,8 @@ enum
   PROP_CLIENT,
   PROP_STREAM_PROPERTIES,
   PROP_SOURCE_OUTPUT_INDEX,
-#ifdef HAVE_PULSE_1_0
   PROP_VOLUME,
   PROP_MUTE,
-#endif
   PROP_LAST
 };
 
@@ -245,14 +239,10 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
   gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&pad_template));
 
-#ifdef HAVE_PULSE_1_0
   /**
    * GstPulseSrc:volume
    *
-   * The volume of the record stream. Only works when using PulseAudio 1.0 or
-   * later.
-   *
-   * Since: 0.10.36
+   * The volume of the record stream.
    */
   g_object_class_install_property (gobject_class,
       PROP_VOLUME, g_param_spec_double ("volume", "Volume",
@@ -263,16 +253,12 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
   /**
    * GstPulseSrc:mute
    *
-   * Whether the stream is muted or not. Only works when using PulseAudio 1.0
-   * or later.
-   *
-   * Since: 0.10.36
+   * Whether the stream is muted or not.
    */
   g_object_class_install_property (gobject_class,
       PROP_MUTE, g_param_spec_boolean ("mute", "Mute",
           "Mute state of this stream",
           DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-#endif
 }
 
 static void
@@ -297,7 +283,6 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
   pulsesrc->paused = TRUE;
   pulsesrc->in_read = FALSE;
 
-#ifdef HAVE_PULSE_1_0
   pulsesrc->volume = DEFAULT_VOLUME;
   pulsesrc->volume_set = FALSE;
 
@@ -305,7 +290,6 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
   pulsesrc->mute_set = FALSE;
 
   pulsesrc->notify = 0;
-#endif
 
   pulsesrc->mixer = NULL;
 
@@ -346,9 +330,7 @@ gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
 
     /* Make sure we don't get any further callbacks */
     pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
-#ifdef HAVE_PULSE_1_0
     pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
-#endif
 
     pa_context_unref (pulsesrc->context);
 
@@ -469,7 +451,6 @@ no_mainloop:
   }
 }
 
-#ifdef HAVE_PULSE_1_0
 static void
 gst_pulsesrc_source_output_info_cb (pa_context * c,
     const pa_source_output_info * i, int eol, void *userdata)
@@ -721,7 +702,6 @@ mute_failed:
     goto unlock;
   }
 }
-#endif
 
 static void
 gst_pulsesrc_set_property (GObject * object,
@@ -759,14 +739,12 @@ gst_pulsesrc_set_property (GObject * object,
         pa_proplist_free (pulsesrc->proplist);
       pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
       break;
-#ifdef HAVE_PULSE_1_0
     case PROP_VOLUME:
       gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
       break;
     case PROP_MUTE:
       gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
       break;
-#endif
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -799,14 +777,12 @@ gst_pulsesrc_get_property (GObject * object,
     case PROP_SOURCE_OUTPUT_INDEX:
       g_value_set_uint (value, pulsesrc->source_output_idx);
       break;
-#ifdef HAVE_PULSE_1_0
     case PROP_VOLUME:
       g_value_set_double (value, gst_pulsesrc_get_stream_volume (pulsesrc));
       break;
     case PROP_MUTE:
       g_value_set_boolean (value, gst_pulsesrc_get_stream_mute (pulsesrc));
       break;
-#endif
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -900,7 +876,6 @@ gst_pulsesrc_stream_overflow_cb (pa_stream * s, void *userdata)
   GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got overflow");
 }
 
-#ifdef HAVE_PULSE_1_0
 static void
 gst_pulsesrc_context_subscribe_cb (pa_context * c,
     pa_subscription_event_type_t t, uint32_t idx, void *userdata)
@@ -922,7 +897,6 @@ gst_pulsesrc_context_subscribe_cb (pa_context * c,
   /* inform streaming thread to notify */
   g_atomic_int_compare_and_exchange (&psrc->notify, 0, 1);
 }
-#endif
 
 static gboolean
 gst_pulsesrc_open (GstAudioSrc * asrc)
@@ -946,10 +920,8 @@ gst_pulsesrc_open (GstAudioSrc * asrc)
 
   pa_context_set_state_callback (pulsesrc->context,
       gst_pulsesrc_context_state_cb, pulsesrc);
-#ifdef HAVE_PULSE_1_0
   pa_context_set_subscribe_callback (pulsesrc->context,
       gst_pulsesrc_context_subscribe_cb, pulsesrc);
-#endif
 
   GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
       GST_STR_NULL (pulsesrc->server));
@@ -1031,12 +1003,10 @@ gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
   pa_threaded_mainloop_lock (pulsesrc->mainloop);
   pulsesrc->in_read = TRUE;
 
-#ifdef HAVE_PULSE_1_0
   if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
     g_object_notify (G_OBJECT (pulsesrc), "volume");
     g_object_notify (G_OBJECT (pulsesrc), "mute");
   }
-#endif
 
   if (pulsesrc->paused)
     goto was_paused;
@@ -1352,13 +1322,10 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   const pa_buffer_attr *actual;
   GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
   pa_stream_flags_t flags;
-#ifdef HAVE_PULSE_1_0
   pa_operation *o;
-#endif
 
   pa_threaded_mainloop_lock (pulsesrc->mainloop);
 
-#ifdef HAVE_PULSE_1_0
   /* enable event notifications */
   GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
   if (!(o = pa_context_subscribe (pulsesrc->context,
@@ -1370,7 +1337,6 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   }
 
   pa_operation_unref (o);
-#endif
 
   wanted.maxlength = -1;
   wanted.tlength = -1;
@@ -1388,10 +1354,8 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
       PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
       PA_STREAM_START_CORKED;
 
-#ifdef HAVE_PULSE_1_0
   if (pulsesrc->mute_set && pulsesrc->mute)
     flags |= PA_STREAM_START_MUTED;
-#endif
 
   if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
           flags) < 0) {
@@ -1420,12 +1384,10 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
   g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
 
-#ifdef HAVE_PULSE_1_0
   if (pulsesrc->volume_set) {
     gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
     pulsesrc->volume_set = FALSE;
   }
-#endif
 
   /* get the actual buffering properties now */
   actual = pa_stream_get_buffer_attr (pulsesrc->stream);

diff --git a/ext/pulse/pulsesrc.h b/ext/pulse/pulsesrc.h
index a9acc87..d4d20f8 100644 (file)
--- a/ext/pulse/pulsesrc.h
+++ b/ext/pulse/pulsesrc.h
@@ -72,14 +72,12 @@ struct _GstPulseSrc
   GstPulseMixerCtrl *mixer;
   GstPulseProbe *probe;
 
-#ifdef HAVE_PULSE_1_0
   gdouble volume;
   gboolean volume_set:1;
   gboolean mute:1;
   gboolean mute_set:1;
 
   gint notify; /* atomic */
-#endif
 
   gboolean corked:1;
   gboolean stream_connected:1;

diff --git a/ext/pulse/pulseutil.c b/ext/pulse/pulseutil.c
index 0ab1d76..2c7dd56 100644 (file)
--- a/ext/pulse/pulseutil.c
+++ b/ext/pulse/pulseutil.c
@@ -156,7 +156,6 @@ gst_pulse_fill_sample_spec (GstAudioRingBufferSpec * spec, pa_sample_spec * ss)
   return TRUE;
 }
 
-#ifdef HAVE_PULSE_1_0
 gboolean
 gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec, pa_format_info ** f,
     guint * channels)
@@ -210,7 +209,6 @@ fail:
     pa_format_info_free (format);
   return FALSE;
 }
-#endif
 
 /* PATH_MAX is not defined everywhere, e.g. on GNU Hurd */
 #ifndef PATH_MAX

diff --git a/ext/pulse/pulseutil.h b/ext/pulse/pulseutil.h
index 0ac070e..8b50fa5 100644 (file)
--- a/ext/pulse/pulseutil.h
+++ b/ext/pulse/pulseutil.h
@@ -33,10 +33,8 @@
 
 gboolean gst_pulse_fill_sample_spec (GstAudioRingBufferSpec * spec,
     pa_sample_spec * ss);
-#ifdef HAVE_PULSE_1_0
 gboolean gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec,
     pa_format_info ** f, guint * channels);
-#endif
 
 gchar *gst_pulse_client_name (void);