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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.
This commit is contained in:
Sebastian Dröge
parent
7431491470
commit
4ab89b397e
4 changed files with 271 additions and 78 deletions
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2008-08-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
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(gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
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(gst_pulsesrc_prepare):
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* ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
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(gst_pulse_channel_map_to_gst):
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* ext/pulse/pulseutil.h:
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If downstream provides no channel layout and >2 channels should be
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used use the default layout that pulseaudio chooses and also
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add this layout to the caps. Fixes bug #547258.
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2008-08-20 Wim Taymans <wim.taymans@collabora.co.uk>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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@ -84,12 +84,15 @@ 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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@ -200,11 +203,9 @@ gst_pulsesrc_base_init (gpointer g_class)
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static void
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gst_pulsesrc_class_init (gpointer g_class, gpointer class_data)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
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GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (g_class);
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GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (g_class);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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parent_class = g_type_class_peek_parent (g_class);
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@ -217,6 +218,8 @@ gst_pulsesrc_class_init (gpointer g_class, gpointer class_data)
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_set_property);
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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);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_pulsesrc_close);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_prepare);
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@ -489,80 +492,6 @@ gst_pulsesrc_close (GstAudioSrc * asrc)
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return TRUE;
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}
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static gboolean
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gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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pa_buffer_attr buf_attr;
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pa_channel_map channel_map;
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GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
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if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec)) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
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("Invalid sample specification."), (NULL));
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goto unlock_and_fail;
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}
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pa_threaded_mainloop_lock (pulsesrc->mainloop);
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if (!pulsesrc->context
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|| pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context state: %s",
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pulsesrc->
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context ? pa_strerror (pa_context_errno (pulsesrc->context)) :
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NULL), (NULL));
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goto unlock_and_fail;
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}
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if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
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"Record Stream",
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&pulsesrc->sample_spec,
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gst_pulse_gst_to_channel_map (&channel_map, spec)))) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to create stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
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pulsesrc);
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pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
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pulsesrc);
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memset (&buf_attr, 0, sizeof (buf_attr));
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buf_attr.maxlength = spec->segtotal * spec->segsize * 2;
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buf_attr.fragsize = spec->segsize;
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if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &buf_attr,
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PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
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PA_STREAM_NOT_MONOTONOUS) < 0) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to connect stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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/* Wait until the stream is ready */
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pa_threaded_mainloop_wait (pulsesrc->mainloop);
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if (pa_stream_get_state (pulsesrc->stream) != PA_STREAM_READY) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to connect stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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return TRUE;
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unlock_and_fail:
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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return FALSE;
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}
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static gboolean
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gst_pulsesrc_unprepare (GstAudioSrc * asrc)
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{
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@ -695,6 +624,194 @@ unlock_and_fail:
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return 0;
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}
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static gboolean
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gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps * caps)
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{
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pa_channel_map channel_map;
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GstStructure *s;
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gboolean need_channel_layout = FALSE;
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GstRingBufferSpec spec;
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memset (&spec, 0, sizeof (GstRingBufferSpec));
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spec.latency_time = GST_SECOND;
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if (!gst_ring_buffer_parse_caps (&spec, caps)) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
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("Can't parse caps."), (NULL));
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goto fail;
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}
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/* Keep the refcount of the caps at 1 to make them writable */
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gst_caps_unref (spec.caps);
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if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec)) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
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("Invalid sample specification."), (NULL));
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goto fail;
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}
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pa_threaded_mainloop_lock (pulsesrc->mainloop);
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if (!pulsesrc->context
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|| pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context state: %s",
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pulsesrc->context ? pa_strerror (pa_context_errno (pulsesrc->
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context)) : NULL), (NULL));
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goto unlock_and_fail;
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}
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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_has_field (s, "channel-layout") ||
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!gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
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if (spec.channels == 1)
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pa_channel_map_init_mono (&channel_map);
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else if (spec.channels == 2)
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pa_channel_map_init_stereo (&channel_map);
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else
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need_channel_layout = TRUE;
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}
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if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
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"Record Stream",
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&pulsesrc->sample_spec,
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(need_channel_layout) ? NULL : &channel_map))) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to create stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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if (need_channel_layout) {
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const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);
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gst_pulse_channel_map_to_gst (m, &spec);
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caps = spec.caps;
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}
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GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, caps);
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pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
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pulsesrc);
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pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
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pulsesrc);
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return TRUE;
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unlock_and_fail:
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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fail:
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return FALSE;
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}
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/* This is essentially gst_base_src_negotiate_default() but the caps
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* are guaranteed to have a channel layout for > 2 channels
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*/
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static gboolean
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gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
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{
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GstCaps *thiscaps;
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GstCaps *caps = NULL;
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GstCaps *peercaps = NULL;
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gboolean result = FALSE;
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/* first see what is possible on our source pad */
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thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc));
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GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
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/* nothing or anything is allowed, we're done */
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if (thiscaps == NULL || gst_caps_is_any (thiscaps))
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goto no_nego_needed;
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/* get the peer caps */
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peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
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GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
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if (peercaps) {
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GstCaps *icaps;
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/* get intersection */
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icaps = gst_caps_intersect (thiscaps, peercaps);
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GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, icaps);
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gst_caps_unref (thiscaps);
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gst_caps_unref (peercaps);
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if (icaps) {
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/* take first (and best, since they are sorted) possibility */
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caps = gst_caps_copy_nth (icaps, 0);
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gst_caps_unref (icaps);
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}
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} else {
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/* no peer, work with our own caps then */
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caps = thiscaps;
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}
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if (caps) {
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caps = gst_caps_make_writable (caps);
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gst_caps_truncate (caps);
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/* now fixate */
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if (!gst_caps_is_empty (caps)) {
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gst_pad_fixate_caps (GST_BASE_SRC_PAD (basesrc), caps);
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GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
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if (gst_caps_is_any (caps)) {
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/* hmm, still anything, so element can do anything and
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* nego is not needed */
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result = TRUE;
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} else if (gst_caps_is_fixed (caps)) {
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/* yay, fixed caps, use those then */
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result = gst_pulsesrc_create_stream (GST_PULSESRC (basesrc), caps);
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if (result)
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gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
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result = TRUE;
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}
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}
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gst_caps_unref (caps);
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}
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return result;
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no_nego_needed:
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{
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GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
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if (thiscaps)
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gst_caps_unref (thiscaps);
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return TRUE;
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}
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}
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static gboolean
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gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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pa_buffer_attr buf_attr;
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GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
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memset (&buf_attr, 0, sizeof (buf_attr));
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buf_attr.maxlength = spec->segtotal * spec->segsize * 2;
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buf_attr.fragsize = spec->segsize;
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if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &buf_attr,
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PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
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PA_STREAM_NOT_MONOTONOUS) < 0) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to connect stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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/* Wait until the stream is ready */
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pa_threaded_mainloop_wait (pulsesrc->mainloop);
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if (pa_stream_get_state (pulsesrc->stream) != PA_STREAM_READY) {
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GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
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("Failed to connect stream: %s",
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pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
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goto unlock_and_fail;
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}
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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return TRUE;
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unlock_and_fail:
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pa_threaded_mainloop_unlock (pulsesrc->mainloop);
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return FALSE;
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}
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static GstStateChangeReturn
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gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
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{
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|
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@ -45,6 +45,30 @@ static const pa_channel_position_t gst_pos_to_pa[GST_AUDIO_CHANNEL_POSITION_NUM]
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[GST_AUDIO_CHANNEL_POSITION_NONE] = PA_CHANNEL_POSITION_INVALID
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};
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/* All index are increased by one because PA_CHANNEL_POSITION_INVALID == -1 */
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static const GstAudioChannelPosition
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pa_to_gst_pos[GST_AUDIO_CHANNEL_POSITION_NUM]
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= {
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[PA_CHANNEL_POSITION_MONO + 1] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
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[PA_CHANNEL_POSITION_FRONT_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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[PA_CHANNEL_POSITION_FRONT_RIGHT + 1] =
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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[PA_CHANNEL_POSITION_REAR_CENTER + 1] =
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
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[PA_CHANNEL_POSITION_REAR_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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[PA_CHANNEL_POSITION_REAR_RIGHT + 1] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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[PA_CHANNEL_POSITION_LFE + 1] = GST_AUDIO_CHANNEL_POSITION_LFE,
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[PA_CHANNEL_POSITION_FRONT_CENTER + 1] =
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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[PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER + 1] =
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
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[PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER + 1] =
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
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[PA_CHANNEL_POSITION_SIDE_LEFT + 1] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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[PA_CHANNEL_POSITION_SIDE_RIGHT + 1] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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[PA_CHANNEL_POSITION_INVALID + 1] = GST_AUDIO_CHANNEL_POSITION_NONE,
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};
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gboolean
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gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss)
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{
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|
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@ -95,7 +119,8 @@ gst_pulse_client_name (void)
|
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}
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|
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pa_channel_map *
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gst_pulse_gst_to_channel_map (pa_channel_map * map, GstRingBufferSpec * spec)
|
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gst_pulse_gst_to_channel_map (pa_channel_map * map,
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const GstRingBufferSpec * spec)
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{
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int i;
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|
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@ -136,3 +161,39 @@ gst_pulse_gst_to_channel_map (pa_channel_map * map, GstRingBufferSpec * spec)
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|
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return map;
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}
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GstRingBufferSpec *
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gst_pulse_channel_map_to_gst (const pa_channel_map * map,
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GstRingBufferSpec * spec)
|
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{
|
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int i;
|
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GstAudioChannelPosition *pos;
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gboolean invalid = FALSE;
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|
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g_return_val_if_fail (map->channels == spec->channels, NULL);
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pos = g_new0 (GstAudioChannelPosition, spec->channels + 1);
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|
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for (i = 0; i < spec->channels; i++) {
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if (map->map[i] == PA_CHANNEL_POSITION_INVALID) {
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invalid = TRUE;
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break;
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} else if (map->map[i] < GST_AUDIO_CHANNEL_POSITION_NUM) {
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pos[i] = pa_to_gst_pos[map->map[i] + 1];
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} else {
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invalid = TRUE;
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break;
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}
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}
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if (invalid) {
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for (i = 0; i < spec->channels; i++)
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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
|
||||
|
|
|
|||
Loading…
Reference in a new issue