Commit 4ab89b39 authored by Sebastian Dröge's avatar Sebastian Dröge

ext/pulse/: If downstream provides no channel layout and >2 channels should be...

ext/pulse/: If downstream provides no channel layout and >2 channels should be used use the default layout that pulse...

Original commit message from CVS:
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
(gst_pulsesrc_prepare):
* ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
(gst_pulse_channel_map_to_gst):
* ext/pulse/pulseutil.h:
If downstream provides no channel layout and >2 channels should be
used use the default layout that pulseaudio chooses and also
add this layout to the caps. Fixes bug #547258.
parent 74314914
2008-08-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
(gst_pulsesrc_prepare):
* ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
(gst_pulse_channel_map_to_gst):
* ext/pulse/pulseutil.h:
If downstream provides no channel layout and >2 channels should be
used use the default layout that pulseaudio chooses and also
add this layout to the caps. Fixes bug #547258.
2008-08-20 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: Peter Kjellerstedt <pkj at axis com>
......
......@@ -84,12 +84,15 @@ static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
guint length);
static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_negotiate (GstBaseSrc * basesrc);
static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
element, GstStateChange transition);
......@@ -200,11 +203,9 @@ gst_pulsesrc_base_init (gpointer g_class)
static void
gst_pulsesrc_class_init (gpointer g_class, gpointer class_data)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (g_class);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (g_class);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
parent_class = g_type_class_peek_parent (g_class);
......@@ -217,6 +218,8 @@ gst_pulsesrc_class_init (gpointer g_class, gpointer class_data)
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_get_property);
gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_pulsesrc_negotiate);
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);
......@@ -489,80 +492,6 @@ gst_pulsesrc_close (GstAudioSrc * asrc)
return TRUE;
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr buf_attr;
pa_channel_map channel_map;
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto unlock_and_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",
pulsesrc->
context ? pa_strerror (pa_context_errno (pulsesrc->context)) :
NULL), (NULL));
goto unlock_and_fail;
}
if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
"Record Stream",
&pulsesrc->sample_spec,
gst_pulse_gst_to_channel_map (&channel_map, spec)))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
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);
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;
}
static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
......@@ -695,6 +624,194 @@ unlock_and_fail:
return 0;
}
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",
pulsesrc->context ? pa_strerror (pa_context_errno (pulsesrc->
context)) : NULL), (NULL));
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;
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
......
......@@ -45,6 +45,30 @@ static const pa_channel_position_t gst_pos_to_pa[GST_AUDIO_CHANNEL_POSITION_NUM]
[GST_AUDIO_CHANNEL_POSITION_NONE] = PA_CHANNEL_POSITION_INVALID
};
/* All index are increased by one because PA_CHANNEL_POSITION_INVALID == -1 */
static const GstAudioChannelPosition
pa_to_gst_pos[GST_AUDIO_CHANNEL_POSITION_NUM]
= {
[PA_CHANNEL_POSITION_MONO + 1] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
[PA_CHANNEL_POSITION_FRONT_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
[PA_CHANNEL_POSITION_FRONT_RIGHT + 1] =
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
[PA_CHANNEL_POSITION_REAR_CENTER + 1] =
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
[PA_CHANNEL_POSITION_REAR_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
[PA_CHANNEL_POSITION_REAR_RIGHT + 1] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
[PA_CHANNEL_POSITION_LFE + 1] = GST_AUDIO_CHANNEL_POSITION_LFE,
[PA_CHANNEL_POSITION_FRONT_CENTER + 1] =
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
[PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER + 1] =
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
[PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER + 1] =
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
[PA_CHANNEL_POSITION_SIDE_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
[PA_CHANNEL_POSITION_SIDE_RIGHT + 1] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
[PA_CHANNEL_POSITION_INVALID + 1] = GST_AUDIO_CHANNEL_POSITION_NONE,
};
gboolean
gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss)
{
......@@ -95,7 +119,8 @@ gst_pulse_client_name (void)
}
pa_channel_map *
gst_pulse_gst_to_channel_map (pa_channel_map * map, GstRingBufferSpec * spec)
gst_pulse_gst_to_channel_map (pa_channel_map * map,
const GstRingBufferSpec * spec)
{
int i;
......@@ -136,3 +161,39 @@ gst_pulse_gst_to_channel_map (pa_channel_map * map, GstRingBufferSpec * spec)
return map;
}
GstRingBufferSpec *
gst_pulse_channel_map_to_gst (const pa_channel_map * map,
GstRingBufferSpec * spec)
{
int i;
GstAudioChannelPosition *pos;
gboolean invalid = FALSE;
g_return_val_if_fail (map->channels == spec->channels, NULL);
pos = g_new0 (GstAudioChannelPosition, spec->channels + 1);
for (i = 0; i < spec->channels; i++) {
if (map->map[i] == PA_CHANNEL_POSITION_INVALID) {
invalid = TRUE;
break;
} else if (map->map[i] < GST_AUDIO_CHANNEL_POSITION_NUM) {
pos[i] = pa_to_gst_pos[map->map[i] + 1];
} else {
invalid = TRUE;
break;
}
}
if (invalid) {
for (i = 0; i < spec->channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
gst_audio_set_channel_positions (gst_caps_get_structure (spec->caps, 0), pos);
g_free (pos);
return spec;
}
......@@ -32,6 +32,9 @@ gboolean gst_pulse_fill_sample_spec (GstRingBufferSpec * spec,
gchar *gst_pulse_client_name (void);
pa_channel_map *gst_pulse_gst_to_channel_map (pa_channel_map * map,
const GstRingBufferSpec * spec);
GstRingBufferSpec *gst_pulse_channel_map_to_gst (const pa_channel_map * map,
GstRingBufferSpec * spec);
#endif
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