mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-18 14:26:43 +00:00
Code Issues 1 Projects Releases Wiki Activity

pulsesrc: Handle negotiation events

Browse source 
This makes sure that we:

a) Destroy an existing stream if a negotiate() request comes in: this is
required when receiving a downstream renegotiation request after a
stream has been created.

b) Create a new stream on prepare(): this is required since we do a
setcaps() in negotiate(), which causes the stream to be dropped by a
ringbuffer release() call (this does not happen during first negotiation
since the release is only done on a running ringbuffer). The subsequent
call to ringbuffer acquire() fails because the stream was lost on
release().

https://bugzilla.gnome.org/show_bug.cgi?id=681247
This commit is contained in:
Arun Raghavan 2012-08-14 15:40:31 +05:30 committed by Wim Taymans
parent 1a8512986a
commit ef9c81d495
1 changed files with 63 additions and 35 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

64
ext/pulse/pulsesrc.c
View file

@ -1136,22 +1136,39 @@ server_dead:
}
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps,
GstAudioRingBufferSpec * rspec)
{
pa_channel_map channel_map;
const pa_channel_map *m;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstAudioRingBufferSpec spec;
GstAudioRingBufferSpec new_spec, *spec;
const gchar *name;
int i;
/* If we already have a stream (renegotiation), free it first */
if (pulsesrc->stream)
gst_pulsesrc_destroy_stream (pulsesrc);
if (rspec) {
/* Post-negotiation, we already have a ringbuffer spec, so we just need to
* use it to create a stream. */
spec = rspec;
/* At this point, we expect the channel-mask to be set in caps, so we just
* use that */
if (!gst_pulse_gst_to_channel_map (&channel_map, spec))
goto invalid_spec;
} else if (caps) {
/* At negotiation time, we get a fixed caps and use it to set up a stream */
s = gst_caps_get_structure (*caps, 0);
gst_structure_get_int (s, "channels", &spec.info.channels);
gst_structure_get_int (s, "channels", &new_spec.info.channels);
if (!gst_structure_has_field (s, "channel-mask")) {
if (spec.info.channels == 1) {
if (new_spec.info.channels == 1) {
pa_channel_map_init_mono (&channel_map);
} else if (spec.info.channels == 2) {
} else if (new_spec.info.channels == 2) {
pa_channel_map_init_stereo (&channel_map);
} else {
need_channel_layout = TRUE;
@ -1160,24 +1177,30 @@ gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
}
}
memset (&spec, 0, sizeof (GstAudioRingBufferSpec));
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, *caps))
memset (&new_spec, 0, sizeof (GstAudioRingBufferSpec));
new_spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&new_spec, *caps))
goto invalid_caps;
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (spec.caps);
gst_caps_unref (new_spec.caps);
if (!need_channel_layout
&& !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
&& !gst_pulse_gst_to_channel_map (&channel_map, &new_spec)) {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
for (i = 0; i < G_N_ELEMENTS (spec.info.position); i++)
spec.info.position[i] = GST_AUDIO_CHANNEL_POSITION_INVALID;
for (i = 0; i < G_N_ELEMENTS (new_spec.info.position); i++)
new_spec.info.position[i] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec))
spec = &new_spec;
} else {
/* !rspec && !caps */
g_assert_not_reached ();
}
if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
@ -1198,12 +1221,14 @@ gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
(need_channel_layout) ? NULL : &channel_map)))
goto create_failed;
if (caps) {
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &spec);
gst_audio_channel_positions_to_valid_order (spec.info.position,
spec.info.channels);
gst_pulse_channel_map_to_gst (m, &new_spec);
gst_audio_channel_positions_to_valid_order (new_spec.info.position,
new_spec.info.channels);
gst_caps_unref (*caps);
*caps = gst_audio_info_to_caps (&spec.info);
*caps = gst_audio_info_to_caps (&new_spec.info);
}
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, *caps);
@ -1305,7 +1330,7 @@ gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
result = TRUE;
} else if (gst_caps_is_fixed (caps)) {
/* yay, fixed caps, use those then */
result = gst_pulsesrc_create_stream (pulsesrc, &caps);
result = gst_pulsesrc_create_stream (pulsesrc, &caps, NULL);
if (result)
result = gst_base_src_set_caps (basesrc, caps);
}
@ -1335,6 +1360,9 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->stream)
gst_pulsesrc_create_stream (pulsesrc, NULL, spec);
{
GstAudioRingBufferSpec s = *spec;
const pa_channel_map *m;