* @title: GstAudioAggregator
* @short_description: Base class that manages a set of audio input pads
* with the purpose of aggregating or mixing their raw audio input buffers
- * @see_also: #GstAggregator
+ * @see_also: #GstAggregator, #GstAudioMixer
*
- * #GstAudioAggregator will perform conversion on the data arriving
- * on its sink pads, based on the format expected downstream.
+ * Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
+ * their source and sink pads,
+ * gst_element_class_add_static_pad_template_with_gtype() is a convenient
+ * helper.
*
- * Subclasses can opt out of the conversion behaviour by setting
- * #GstAudioAggregatorClass.convert_buffer() to %NULL.
+ * #GstAudioAggregator can perform conversion on the data arriving
+ * on its sink pads, based on the format expected downstream: in order
+ * to enable that behaviour, the GType of the sink pads must either be
+ * a (subclass of) #GstAudioAggregatorConvertPad to use the default
+ * #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
+ * implementing #GstAudioAggregatorPad.convert_buffer.
*
- * Subclasses that wish to use the default conversion implementation
- * should use a (subclass of) #GstAudioAggregatorConvertPad as their
- * #GstAggregatorClass.sinkpads_type, as it will cache the created
- * #GstAudioConverter and install a property allowing to configure it,
- * #GstAudioAggregatorPad:converter-config.
+ * To allow for the output caps to change, the mechanism is the same as
+ * above, with the GType of the source pad.
*
- * Subclasses that wish to perform custom conversion should override
- * #GstAudioAggregatorClass.convert_buffer().
+ * See #GstAudioMixer for an example.
*
* When conversion is enabled, #GstAudioAggregator will accept
* any type of raw audio caps and perform conversion
* the first configured sink pad to finish fixating its source pad
* caps.
*
- * Additionally, handling audio conversion directly in the element
- * means that this base class supports safely reconfiguring its
- * source pad.
- *
* A notable exception for now is the sample rate, sink pads must
* have the same sample rate as either the downstream requirement,
* or the first configured pad, or a combination of both (when
aaggcpad->priv->converter_config_changed = FALSE;
}
+static void
+gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad *
+ aaggpad)
+{
+ GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed =
+ TRUE;
+}
+
static GstBuffer *
-gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad *
- aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
+gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad *
+ aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
GstBuffer * input_buffer)
{
GstBuffer *res;
+ GstAudioAggregatorConvertPad *aaggcpad =
+ GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad);
gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
out_info);
klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstAudioAggregatorPadClass *aaggpad_class =
+ (GstAudioAggregatorPadClass *) klass;
g_type_class_add_private (klass,
sizeof (GstAudioAggregatorConvertPadPrivate));
"when converting this pad's audio buffers",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ aaggpad_class->convert_buffer =
+ gst_audio_aggregator_convert_pad_convert_buffer;
+
+ aaggpad_class->update_conversion_info =
+ gst_audio_aggregator_pad_update_conversion_info;
+
gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
}
}
static GstBuffer *
-gst_audio_aggregator_convert_once (GstAudioAggregator * aagg, GstPad * pad,
- GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
-{
- GstAudioConverter *converter =
- gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
- in_info, out_info, NULL);
- gint insize = gst_buffer_get_size (buffer);
- gsize insamples = insize / in_info->bpf;
- gsize outsamples = gst_audio_converter_get_out_frames (converter,
- insamples);
- gint outsize = outsamples * out_info->bpf;
- GstMapInfo inmap, outmap;
- GstBuffer *converted = gst_buffer_new_allocate (NULL, outsize, NULL);
-
- gst_buffer_copy_into (converted, buffer,
- GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
- GST_BUFFER_COPY_META, 0, -1);
-
- gst_buffer_map (buffer, &inmap, GST_MAP_READ);
- gst_buffer_map (converted, &outmap, GST_MAP_WRITE);
-
- gst_audio_converter_samples (converter,
- GST_AUDIO_CONVERTER_FLAG_NONE,
- (gpointer *) & inmap.data, insamples,
- (gpointer *) & outmap.data, outsamples);
-
- gst_buffer_unmap (buffer, &inmap);
- gst_buffer_unmap (converted, &outmap);
- gst_audio_converter_free (converter);
-
- return converted;
-}
-
-static GstBuffer *
-gst_audio_aggregator_default_convert_buffer (GstAudioAggregator * aagg,
- GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info,
- GstBuffer * buffer)
-{
- if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
- return
- gst_audio_aggregator_convert_pad_convert_buffer
- (GST_AUDIO_AGGREGATOR_CONVERT_PAD (pad),
- &GST_AUDIO_AGGREGATOR_PAD (pad)->info, out_info, buffer);
- else
- return gst_audio_aggregator_convert_once (aagg, pad, in_info, out_info,
- buffer);
-}
-
-static GstBuffer *
gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
{
- GstAudioAggregatorClass *klass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
+ GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad);
+ GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
g_assert (klass->convert_buffer);
- return klass->convert_buffer (aagg, pad, in_info, out_info, buffer);
+ return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
}
static void
gst_audio_aggregator_negotiated_src_caps;
klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
- klass->convert_buffer = gst_audio_aggregator_default_convert_buffer;
GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
ret = FALSE;
} else {
+ GstAudioAggregatorPadClass *klass =
+ GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
GST_OBJECT_LOCK (aaggpad);
gst_audio_info_from_caps (&aaggpad->info, caps);
- if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
- GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
- priv->converter_config_changed = TRUE;
+ if (klass->update_conversion_info)
+ klass->update_conversion_info (aaggpad);
GST_OBJECT_UNLOCK (aaggpad);
}
static GstCaps *
gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
{
- GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
GstAudioAggregatorPad *first_configured_pad;
- if (!aaggclass->convert_buffer)
+ if (!GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
return
GST_AGGREGATOR_CLASS
(gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps);
for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
GstAudioAggregatorPad *aaggpad = l->data;
+ GstAudioAggregatorPadClass *klass =
+ GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
- if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
- GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
- priv->converter_config_changed = TRUE;
+ if (klass->update_conversion_info)
+ klass->update_conversion_info (aaggpad);
/* If we currently were mixing a buffer, we need to convert it to the new
* format */
gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
- GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
GstAudioInfo info;
GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
- if (aaggclass->convert_buffer) {
+ if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) {
gst_audio_aggregator_update_converters (aagg, &info);
if (aagg->priv->current_buffer
&& !gst_audio_info_is_equal (&aagg->info, &info)) {
- GstBuffer *converted =
+ GstBuffer *converted;
+ GstAudioAggregatorPadClass *klass =
+ GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
+
+ if (klass->update_conversion_info)
+ klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));
+
+ converted =
gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info,
&info, aagg->priv->current_buffer);
gst_buffer_unref (aagg->priv->current_buffer);
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad)
{
- GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
- if (aaggclass->convert_buffer) {
+ if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
} else {
/* New buffer? */
if (!pad->priv->buffer) {
- if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
+ if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
pad->priv->buffer =
gst_audio_aggregator_convert_buffer
(aagg, GST_PAD (pad), &pad->info, &aagg->info,
GST_END_TEST;
+/* In this test, we create two input buffers with a duration of 1 second,
+ * and require the audiomixer to output 1.5 second long buffers.
+ *
+ * After we have input two buffers, we change the output format
+ * from S8 to S32, then push a last buffer.
+ *
+ * This makes audioaggregator convert its "half-mixed" current_buffer,
+ * we can then ensure that the second output buffer is as expected.
+ */
+GST_START_TEST (test_change_output_caps_mid_output_buffer)
+{
+ GstSegment segment;
+ GstElement *bin, *audiomixer, *capsfilter, *sink;
+ GstBus *bus;
+ GstPad *sinkpad;
+ gboolean res;
+ GstStateChangeReturn state_res;
+ GstFlowReturn ret;
+ GstEvent *event;
+ GstBuffer *buffer;
+ GstCaps *caps;
+ GstQuery *drain;
+ GstMapInfo inmap;
+ GstMapInfo outmap;
+ guint i;
+
+ bin = gst_pipeline_new ("pipeline");
+ bus = gst_element_get_bus (bin);
+ gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+ g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+ g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+ g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+ audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+ g_object_set (audiomixer, "output-buffer-duration", 1500 * GST_MSECOND, NULL);
+ capsfilter = gst_element_factory_make ("capsfilter", NULL);
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
+
+ res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
+ fail_unless (res == TRUE, NULL);
+
+ state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
+ fail_if (sinkpad == NULL, NULL);
+
+ gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
+
+ caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, "S8",
+ "layout", G_TYPE_STRING, "interleaved",
+ "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+
+ gst_pad_set_caps (sinkpad, caps);
+ g_object_set (capsfilter, "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ segment.start = 0;
+ segment.stop = 3 * GST_SECOND;
+ segment.time = 0;
+ event = gst_event_new_segment (&segment);
+ gst_pad_send_event (sinkpad, event);
+
+ buffer = new_buffer (10, 0, 0, 1 * GST_SECOND, 0);
+ ret = gst_pad_chain (sinkpad, buffer);
+ ck_assert_int_eq (ret, GST_FLOW_OK);
+
+ buffer = new_buffer (10, 0, 1 * GST_SECOND, 1 * GST_SECOND, 0);
+ gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
+ memset (inmap.data, 1, 10);
+ gst_buffer_unmap (buffer, &inmap);
+ ret = gst_pad_chain (sinkpad, buffer);
+ ck_assert_int_eq (ret, GST_FLOW_OK);
+
+ drain = gst_query_new_drain ();
+ gst_pad_query (sinkpad, drain);
+ gst_query_unref (drain);
+
+ caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
+ "layout", G_TYPE_STRING, "interleaved",
+ "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+ g_object_set (capsfilter, "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ gst_buffer_replace (&handoff_buffer, NULL);
+ g_object_set (sink, "signal-handoffs", TRUE, NULL);
+ g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
+
+ buffer = new_buffer (10, 0, 2 * GST_SECOND, 1 * GST_SECOND, 0);
+ gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
+ memset (inmap.data, 0, 10);
+ gst_buffer_unmap (buffer, &inmap);
+ ret = gst_pad_chain (sinkpad, buffer);
+ ck_assert_int_eq (ret, GST_FLOW_OK);
+
+ drain = gst_query_new_drain ();
+ gst_pad_query (sinkpad, drain);
+ gst_query_unref (drain);
+
+ fail_unless (handoff_buffer);
+ fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 60);
+
+ gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
+ for (i = 0; i < 15; i++) {
+ guint32 sample;
+
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
+#else
+ sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
+#endif
+
+ if (i < 5) {
+ fail_unless_equals_int (sample, 1 << 24);
+ } else {
+ fail_unless_equals_int (sample, 0);
+ }
+ }
+
+ gst_buffer_unmap (handoff_buffer, &outmap);
+
+ gst_element_release_request_pad (audiomixer, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_element_set_state (bin, GST_STATE_NULL);
+ gst_bus_remove_signal_watch (bus);
+ gst_object_unref (bus);
+ gst_object_unref (bin);
+}
+
+GST_END_TEST;
static Suite *
audiomixer_suite (void)
{
tcase_add_test (tc_chain, test_sinkpad_property_controller);
tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
tcase_add_test (tc_chain, test_change_output_caps);
+ tcase_add_test (tc_chain, test_change_output_caps_mid_output_buffer);
/* Use a longer timeout */
#ifdef HAVE_VALGRIND
projects / platform / upstream / gstreamer.git / commitdiff