mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-26 19:51:11 +00:00
pulseaudio: require pulseaudio >= 1.0
This commit is contained in:
Tim-Philipp Müller
parent
be0d6baac5
commit
eeaa9e0bbc
9 changed files with 7 additions and 150 deletions
10
configure.ac
10
configure.ac
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@ -806,15 +806,7 @@ AG_GST_CHECK_FEATURE(PULSE, [pulseaudio plug-in], pulseaudio, [
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dnl used in ext/pulse/pulseutil.c
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AC_CHECK_HEADERS([process.h])
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AG_GST_PKG_CHECK_MODULES(PULSE, libpulse >= 0.9.16)
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AG_GST_PKG_CHECK_MODULES(PULSE_0_9_20, libpulse >= 0.9.20)
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if test x$HAVE_PULSE_0_9_20 = xyes; then
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AC_DEFINE(HAVE_PULSE_0_9_20, 1, [defined if pulseaudio >= 0.9.20 is available])
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fi
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AG_GST_PKG_CHECK_MODULES(PULSE_1_0, libpulse >= 0.98)
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if test x$HAVE_PULSE_1_0 = xyes; then
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AC_DEFINE(HAVE_PULSE_1_0, 1, [defined if pulseaudio >= 1.0 is available])
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fi
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AG_GST_PKG_CHECK_MODULES(PULSE, libpulse >= 1.0)
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])
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dnl *** dv1394 ***
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@ -49,11 +49,11 @@ plugin_init (GstPlugin * plugin)
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GST_TYPE_PULSESRC))
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return FALSE;
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#ifdef HAVE_PULSE_1_0
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/* FIXME 0.11: this helper bin sink should just go away, reconfiguration
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* should be handled using reconfigure events */
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if (!gst_element_register (plugin, "pulseaudiosink", GST_RANK_PRIMARY + 11,
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GST_TYPE_PULSE_AUDIO_SINK))
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return FALSE;
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#endif
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if (!gst_element_register (plugin, "pulsemixer", GST_RANK_NONE,
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GST_TYPE_PULSEMIXER))
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@ -49,7 +49,7 @@
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#include "config.h"
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#endif
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#ifdef HAVE_PULSE_1_0
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/* FIXME 0.11: pulseaudiosink helper bin must die */
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#include <gst/pbutils/pbutils.h>
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#include <gst/gst-i18n-plugin.h>
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@ -958,5 +958,3 @@ gst_pulse_audio_sink_change_state (GstElement * element,
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out:
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return ret;
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}
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#endif /* HAVE_PULSE_1_0 */
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@ -139,13 +139,9 @@ struct _GstPulseRingBuffer
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pa_context *context;
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pa_stream *stream;
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#ifdef HAVE_PULSE_1_0
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pa_format_info *format;
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guint channels;
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gboolean is_pcm;
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#else
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pa_sample_spec sample_spec;
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#endif
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void *m_data;
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size_t m_towrite;
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@ -230,13 +226,9 @@ gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf)
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pbuf->context = NULL;
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pbuf->stream = NULL;
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#ifdef HAVE_PULSE_1_0
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pbuf->format = NULL;
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pbuf->channels = 0;
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pbuf->is_pcm = FALSE;
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#else
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pa_sample_spec_init (&pbuf->sample_spec);
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#endif
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pbuf->m_data = NULL;
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pbuf->m_towrite = 0;
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@ -265,14 +257,12 @@ gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
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pbuf->m_offset = 0;
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pbuf->m_lastoffset = 0;
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}
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#ifdef HAVE_PULSE_1_0
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if (pbuf->format) {
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pa_format_info_free (pbuf->format);
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pbuf->format = NULL;
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pbuf->channels = 0;
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pbuf->is_pcm = FALSE;
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}
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#endif
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pa_stream_disconnect (pbuf->stream);
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@ -424,7 +414,6 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
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if (idx != pa_stream_get_index (pbuf->stream))
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continue;
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#ifdef HAVE_PULSE_1_0
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if (psink->device && pbuf->is_pcm &&
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!g_str_equal (psink->device,
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pa_stream_get_device_name (pbuf->stream))) {
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@ -443,7 +432,6 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
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if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
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GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
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}
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#endif
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/* Actually this event is also triggered when other properties of
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* the stream change that are unrelated to the volume. However it is
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@ -738,7 +726,6 @@ gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
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gst_element_post_message (GST_ELEMENT_CAST (psink),
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gst_message_new_request_state (GST_OBJECT_CAST (psink),
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GST_STATE_PLAYING));
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#ifdef HAVE_PULSE_1_0
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} else if (!strcmp (name, PA_STREAM_EVENT_FORMAT_LOST)) {
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GstEvent *renego;
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@ -763,7 +750,6 @@ gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
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GST_ELEMENT_ERROR (psink, STREAM, FORMAT, ("Sink format changed"),
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("Sink format changed"));
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}
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#endif
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} else {
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GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
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}
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@ -804,18 +790,14 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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const pa_buffer_attr *actual;
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pa_channel_map channel_map;
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pa_operation *o = NULL;
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#ifdef HAVE_PULSE_0_9_20
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pa_cvolume v;
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#endif
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pa_cvolume *pv = NULL;
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pa_stream_flags_t flags;
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const gchar *name;
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GstAudioClock *clock;
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#ifdef HAVE_PULSE_1_0
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pa_format_info *formats[1];
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#ifndef GST_DISABLE_GST_DEBUG
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gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
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#endif
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#endif
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
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@ -823,14 +805,9 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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GST_LOG_OBJECT (psink, "creating sample spec");
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/* convert the gstreamer sample spec to the pulseaudio format */
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#ifdef HAVE_PULSE_1_0
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if (!gst_pulse_fill_format_info (spec, &pbuf->format, &pbuf->channels))
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goto invalid_spec;
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pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
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#else
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if (!gst_pulse_fill_sample_spec (spec, &pbuf->sample_spec))
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goto invalid_spec;
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#endif
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pa_threaded_mainloop_lock (mainloop);
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@ -847,12 +824,8 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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pa_operation_unref (o);
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/* initialize the channel map */
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#ifdef HAVE_PULSE_1_0
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if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
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pa_format_info_set_channel_map (pbuf->format, &channel_map);
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#else
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gst_pulse_gst_to_channel_map (&channel_map, spec);
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#endif
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/* find a good name for the stream */
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if (psink->stream_name)
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@ -861,17 +834,10 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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name = "Playback Stream";
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/* create a stream */
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#ifdef HAVE_PULSE_1_0
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formats[0] = pbuf->format;
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if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
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psink->proplist)))
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goto stream_failed;
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#else
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GST_LOG_OBJECT (psink, "creating stream with name %s", name);
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if (!(pbuf->stream = pa_stream_new_with_proplist (pbuf->context, name,
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&pbuf->sample_spec, &channel_map, psink->proplist)))
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goto stream_failed;
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#endif
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/* install essential callbacks */
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pa_stream_set_state_callback (pbuf->stream,
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@ -904,26 +870,19 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
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GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
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#ifdef HAVE_PULSE_0_9_20
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/* configure volume when we changed it, else we leave the default */
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if (psink->volume_set) {
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GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
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pv = &v;
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#ifdef HAVE_PULSE_1_0
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if (pbuf->is_pcm)
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gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
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else {
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GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
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pv = NULL;
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}
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#else
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gst_pulse_cvolume_from_linear (pv, pbuf->sample_spec.channels,
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psink->volume);
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#endif
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} else {
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pv = NULL;
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}
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#endif
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/* construct the flags */
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flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
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@ -949,7 +908,6 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
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goto connect_failed;
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#ifdef HAVE_PULSE_1_0
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g_free (psink->device);
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psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
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@ -957,7 +915,6 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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pa_format_info_snprint (print_buf, sizeof (print_buf),
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pa_stream_get_format_info (pbuf->stream));
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GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
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#endif
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#endif
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/* After we passed the volume off of to PA we never want to set it
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@ -1032,7 +989,6 @@ gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
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gst_pulsering_destroy_stream (pbuf);
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pa_threaded_mainloop_unlock (mainloop);
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#ifdef HAVE_PULSE_1_0
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{
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GstPulseSink *psink;
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@ -1040,7 +996,6 @@ gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
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g_atomic_int_set (&psink->format_lost, FALSE);
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psink->format_lost_time = GST_CLOCK_TIME_NONE;
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}
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#endif
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return TRUE;
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}
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@ -1063,12 +1018,10 @@ gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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#ifdef HAVE_PULSE_1_0
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if (g_atomic_int_get (&psink->format_lost)) {
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/* Sink format changed, stream's gone so fake being paused */
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return TRUE;
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}
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#endif
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GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
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if (pbuf->corked != corked) {
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@ -1251,13 +1204,11 @@ gst_pulseringbuffer_stop (GstAudioRingBuffer * buf)
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GST_DEBUG_OBJECT (psink, "signal commit thread");
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pa_threaded_mainloop_signal (mainloop, 0);
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}
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#ifdef HAVE_PULSE_1_0
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if (g_atomic_int_get (&psink->format_lost)) {
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/* Don't try to flush, the stream's probably gone by now */
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res = TRUE;
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goto cleanup;
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}
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#endif
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/* then try to flush, it's not fatal when this fails */
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GST_DEBUG_OBJECT (psink, "flushing");
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@ -1428,12 +1379,10 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
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* needed to properly handle reverse playback: it points to the last sample. */
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data_end = data + (bpf * inr);
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#ifdef HAVE_PULSE_1_0
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if (g_atomic_int_get (&psink->format_lost)) {
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/* Sink format changed, drop the data and hope upstream renegotiates */
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goto fake_done;
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}
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#endif
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if (pbuf->paused)
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goto was_paused;
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@ -1482,12 +1431,10 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
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for (;;) {
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pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
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#ifdef HAVE_PULSE_1_0
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if (g_atomic_int_get (&psink->format_lost)) {
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/* Sink format changed, give up and hope upstream renegotiates */
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goto fake_done;
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}
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#endif
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if (pbuf->m_writable == (size_t) - 1)
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goto writable_size_failed;
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@ -1538,13 +1485,11 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
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GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
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(guint) avail, offset);
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#ifdef HAVE_PULSE_1_0
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/* No trick modes for passthrough streams */
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if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
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GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
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goto unlock_and_fail;
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}
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#endif
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if (G_LIKELY (inr == outr && !reverse)) {
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/* no rate conversion, simply write out the samples */
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@ -1626,9 +1571,7 @@ gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
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}
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}
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#ifdef HAVE_PULSE_1_0
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fake_done:
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#endif
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/* we consumed all samples here */
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data = data_end + bpf;
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@ -1931,13 +1874,11 @@ gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
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pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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#ifdef HAVE_PULSE_1_0
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if (g_atomic_int_get (&psink->format_lost)) {
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/* Stream was lost in a format change, it'll get set up again once
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* upstream renegotiates */
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return psink->format_lost_time;
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}
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#endif
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pa_threaded_mainloop_lock (mainloop);
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if (gst_pulsering_is_dead (psink, pbuf, TRUE))
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@ -1973,10 +1914,8 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
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{
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GstPulseRingBuffer *pbuf;
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GstPulseSink *psink;
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#ifdef HAVE_PULSE_1_0
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GList *l;
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guint8 j;
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#endif
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pbuf = GST_PULSERING_BUFFER_CAST (userdata);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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@ -1987,7 +1926,6 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
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g_free (psink->device_description);
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psink->device_description = g_strdup (i->description);
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#ifdef HAVE_PULSE_1_0
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g_mutex_lock (psink->sink_formats_lock);
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for (l = g_list_first (psink->sink_formats); l; l = g_list_next (l))
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@ -2001,13 +1939,11 @@ gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
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pa_format_info_copy (i->formats[j]));
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g_mutex_unlock (psink->sink_formats_lock);
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#endif
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done:
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pa_threaded_mainloop_signal (mainloop, 0);
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}
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#ifdef HAVE_PULSE_1_0
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/* NOTE: If you're making changes here, see if pulseaudiosink acceptcaps also
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* needs to be changed accordingly. */
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static gboolean
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@ -2139,7 +2075,6 @@ info_failed:
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goto out;
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}
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}
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#endif
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static void
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gst_pulsesink_init (GstPulseSink * pulsesink)
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@ -2149,10 +2084,8 @@ gst_pulsesink_init (GstPulseSink * pulsesink)
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pulsesink->device_description = NULL;
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pulsesink->client_name = gst_pulse_client_name ();
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#ifdef HAVE_PULSE_1_0
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pulsesink->sink_formats_lock = g_mutex_new ();
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pulsesink->sink_formats = NULL;
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#endif
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pulsesink->volume = DEFAULT_VOLUME;
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pulsesink->volume_set = FALSE;
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@ -2162,10 +2095,8 @@ gst_pulsesink_init (GstPulseSink * pulsesink)
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pulsesink->notify = 0;
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#ifdef HAVE_PULSE_1_0
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g_atomic_int_set (&pulsesink->format_lost, FALSE);
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pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
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#endif
|
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pulsesink->properties = NULL;
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pulsesink->proplist = NULL;
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|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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__ */
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue