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audioaggregator: refactor conversion API
For the rationale, see: https://bugzilla.gnome.org/show_bug.cgi?id=793917 Also test audiomixer conversion of current output buffer
This commit is contained in:
Mathieu Duponchelle
parent
c920d994ab
commit
10835e9919
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@ -28,22 +28,24 @@
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* @title: GstAudioAggregator
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* @short_description: Base class that manages a set of audio input pads
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* with the purpose of aggregating or mixing their raw audio input buffers
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* @see_also: #GstAggregator
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* @see_also: #GstAggregator, #GstAudioMixer
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*
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* #GstAudioAggregator will perform conversion on the data arriving
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* on its sink pads, based on the format expected downstream.
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* Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
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* their source and sink pads,
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* gst_element_class_add_static_pad_template_with_gtype() is a convenient
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* helper.
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*
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* Subclasses can opt out of the conversion behaviour by setting
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* #GstAudioAggregatorClass.convert_buffer() to %NULL.
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* #GstAudioAggregator can perform conversion on the data arriving
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* on its sink pads, based on the format expected downstream: in order
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* to enable that behaviour, the GType of the sink pads must either be
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* a (subclass of) #GstAudioAggregatorConvertPad to use the default
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* #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
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* implementing #GstAudioAggregatorPad.convert_buffer.
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*
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* Subclasses that wish to use the default conversion implementation
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* should use a (subclass of) #GstAudioAggregatorConvertPad as their
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* #GstAggregatorClass.sinkpads_type, as it will cache the created
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* #GstAudioConverter and install a property allowing to configure it,
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* #GstAudioAggregatorPad:converter-config.
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* To allow for the output caps to change, the mechanism is the same as
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* above, with the GType of the source pad.
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*
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* Subclasses that wish to perform custom conversion should override
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* #GstAudioAggregatorClass.convert_buffer().
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* See #GstAudioMixer for an example.
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*
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* When conversion is enabled, #GstAudioAggregator will accept
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* any type of raw audio caps and perform conversion
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@ -54,10 +56,6 @@
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* the first configured sink pad to finish fixating its source pad
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* caps.
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*
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* Additionally, handling audio conversion directly in the element
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* means that this base class supports safely reconfiguring its
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* source pad.
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*
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* A notable exception for now is the sample rate, sink pads must
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* have the same sample rate as either the downstream requirement,
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* or the first configured pad, or a combination of both (when
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@ -223,12 +221,22 @@ gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
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aaggcpad->priv->converter_config_changed = FALSE;
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}
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static void
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gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad *
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aaggpad)
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{
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed =
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TRUE;
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}
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static GstBuffer *
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gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad *
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aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
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gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad *
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aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
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GstBuffer * input_buffer)
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{
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GstBuffer *res;
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GstAudioAggregatorConvertPad *aaggcpad =
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad);
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gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
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out_info);
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@ -327,6 +335,8 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
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klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAudioAggregatorPadClass *aaggpad_class =
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(GstAudioAggregatorPadClass *) klass;
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g_type_class_add_private (klass,
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sizeof (GstAudioAggregatorConvertPadPrivate));
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@ -339,6 +349,12 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
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"when converting this pad's audio buffers",
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GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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aaggpad_class->convert_buffer =
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gst_audio_aggregator_convert_pad_convert_buffer;
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aaggpad_class->update_conversion_info =
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gst_audio_aggregator_pad_update_conversion_info;
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gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
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}
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@ -449,64 +465,16 @@ gst_audio_aggregator_get_next_time (GstAggregator * agg)
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return next_time;
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}
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static GstBuffer *
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gst_audio_aggregator_convert_once (GstAudioAggregator * aagg, GstPad * pad,
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GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
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{
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GstAudioConverter *converter =
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gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
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in_info, out_info, NULL);
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gint insize = gst_buffer_get_size (buffer);
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gsize insamples = insize / in_info->bpf;
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gsize outsamples = gst_audio_converter_get_out_frames (converter,
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insamples);
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gint outsize = outsamples * out_info->bpf;
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GstMapInfo inmap, outmap;
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GstBuffer *converted = gst_buffer_new_allocate (NULL, outsize, NULL);
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gst_buffer_copy_into (converted, buffer,
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GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
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GST_BUFFER_COPY_META, 0, -1);
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gst_buffer_map (buffer, &inmap, GST_MAP_READ);
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gst_buffer_map (converted, &outmap, GST_MAP_WRITE);
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gst_audio_converter_samples (converter,
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GST_AUDIO_CONVERTER_FLAG_NONE,
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(gpointer *) & inmap.data, insamples,
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(gpointer *) & outmap.data, outsamples);
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gst_buffer_unmap (buffer, &inmap);
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gst_buffer_unmap (converted, &outmap);
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gst_audio_converter_free (converter);
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return converted;
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}
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static GstBuffer *
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gst_audio_aggregator_default_convert_buffer (GstAudioAggregator * aagg,
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GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info,
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GstBuffer * buffer)
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{
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if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
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return
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gst_audio_aggregator_convert_pad_convert_buffer
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(GST_AUDIO_AGGREGATOR_CONVERT_PAD (pad),
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&GST_AUDIO_AGGREGATOR_PAD (pad)->info, out_info, buffer);
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else
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return gst_audio_aggregator_convert_once (aagg, pad, in_info, out_info,
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buffer);
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}
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static GstBuffer *
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gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
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GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
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{
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GstAudioAggregatorClass *klass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
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GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad);
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GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
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g_assert (klass->convert_buffer);
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return klass->convert_buffer (aagg, pad, in_info, out_info, buffer);
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return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
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}
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static void
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@ -542,7 +510,6 @@ gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
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gst_audio_aggregator_negotiated_src_caps;
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klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
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klass->convert_buffer = gst_audio_aggregator_default_convert_buffer;
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GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
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GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
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@ -746,11 +713,12 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
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gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
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ret = FALSE;
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} else {
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GstAudioAggregatorPadClass *klass =
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GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
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GST_OBJECT_LOCK (aaggpad);
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gst_audio_info_from_caps (&aaggpad->info, caps);
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if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
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priv->converter_config_changed = TRUE;
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if (klass->update_conversion_info)
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klass->update_conversion_info (aaggpad);
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GST_OBJECT_UNLOCK (aaggpad);
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}
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@ -792,10 +760,9 @@ gst_audio_aggregator_update_src_caps (GstAggregator * agg,
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static GstCaps *
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gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
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{
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GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
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GstAudioAggregatorPad *first_configured_pad;
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if (!aaggclass->convert_buffer)
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if (!GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
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return
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GST_AGGREGATOR_CLASS
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(gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps);
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@ -844,10 +811,11 @@ gst_audio_aggregator_update_converters (GstAudioAggregator * aagg,
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for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
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GstAudioAggregatorPad *aaggpad = l->data;
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GstAudioAggregatorPadClass *klass =
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GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
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if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
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priv->converter_config_changed = TRUE;
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if (klass->update_conversion_info)
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klass->update_conversion_info (aaggpad);
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/* If we currently were mixing a buffer, we need to convert it to the new
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* format */
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@ -865,7 +833,6 @@ static gboolean
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gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
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{
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GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
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GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
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GstAudioInfo info;
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GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
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@ -878,12 +845,19 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
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GST_AUDIO_AGGREGATOR_LOCK (aagg);
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GST_OBJECT_LOCK (aagg);
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if (aaggclass->convert_buffer) {
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if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) {
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gst_audio_aggregator_update_converters (aagg, &info);
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if (aagg->priv->current_buffer
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&& !gst_audio_info_is_equal (&aagg->info, &info)) {
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GstBuffer *converted =
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GstBuffer *converted;
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GstAudioAggregatorPadClass *klass =
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GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
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if (klass->update_conversion_info)
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klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));
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converted =
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gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info,
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&info, aagg->priv->current_buffer);
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gst_buffer_unref (aagg->priv->current_buffer);
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@ -1324,7 +1298,6 @@ static gboolean
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gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
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GstAudioAggregatorPad * pad)
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{
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GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
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GstClockTime start_time, end_time;
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gboolean discont = FALSE;
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guint64 start_offset, end_offset;
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@ -1333,7 +1306,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
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GstAggregator *agg = GST_AGGREGATOR (aagg);
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GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
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if (aaggclass->convert_buffer) {
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if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
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rate = GST_AUDIO_INFO_RATE (&aagg->info);
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bpf = GST_AUDIO_INFO_BPF (&aagg->info);
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} else {
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@ -1802,7 +1775,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
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/* New buffer? */
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if (!pad->priv->buffer) {
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if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
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if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
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pad->priv->buffer =
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gst_audio_aggregator_convert_buffer
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(aagg, GST_PAD (pad), &pad->info, &aagg->info,
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@ -79,12 +79,21 @@ struct _GstAudioAggregatorPad
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/**
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* GstAudioAggregatorPadClass:
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*
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* @convert_buffer: Convert a buffer from one format to another.
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* @update_conversion_info: Called when either the input or output
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* formats have changed.
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*/
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struct _GstAudioAggregatorPadClass
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{
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GstAggregatorPadClass parent_class;
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GstBuffer * (* convert_buffer) (GstAudioAggregatorPad * pad,
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GstAudioInfo *in_info,
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GstAudioInfo *out_info,
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GstBuffer * buffer);
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void (* update_conversion_info) (GstAudioAggregatorPad *pad);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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@ -181,10 +190,6 @@ struct _GstAudioAggregator
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* buffer. The in_offset and out_offset are in "frames", which is
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* the size of a sample times the number of channels. Returns TRUE if
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* any non-silence was added to the buffer
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* @convert_buffer: Convert a buffer from one format to another. The pad
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* is either a sinkpad, when converting an input buffer, or the source pad,
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* when converting the output buffer after a downstream format change is
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* requested.
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*/
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struct _GstAudioAggregatorClass {
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GstAggregatorClass parent_class;
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@ -194,11 +199,6 @@ struct _GstAudioAggregatorClass {
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gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg,
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GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset,
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GstBuffer * outbuf, guint out_offset, guint num_frames);
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GstBuffer * (* convert_buffer) (GstAudioAggregator *aagg,
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GstPad * pad,
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GstAudioInfo *in_info,
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GstAudioInfo *out_info,
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GstBuffer * buffer);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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@ -560,8 +560,8 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass)
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gobject_class->get_property = gst_audio_interleave_get_property;
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gobject_class->finalize = gst_audio_interleave_finalize;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_audio_interleave_src_template);
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
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&gst_audio_interleave_src_template, GST_TYPE_AUDIO_AGGREGATOR_PAD);
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
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&gst_audio_interleave_sink_template, GST_TYPE_AUDIO_INTERLEAVE_PAD);
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gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave",
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@ -580,7 +580,6 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass)
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agg_class->negotiated_src_caps = gst_audio_interleave_negotiated_src_caps;
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aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer;
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aagg_class->convert_buffer = NULL;
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/**
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* GstInterleave:channel-positions
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@ -224,8 +224,8 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_audiomixer_src_template);
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
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&gst_audiomixer_src_template, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD);
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
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&gst_audiomixer_sink_template, GST_TYPE_AUDIO_MIXER_PAD);
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gst_element_class_set_static_metadata (gstelement_class, "AudioMixer",
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@ -1849,6 +1849,141 @@ GST_START_TEST (test_change_output_caps)
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GST_END_TEST;
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/* In this test, we create two input buffers with a duration of 1 second,
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* and require the audiomixer to output 1.5 second long buffers.
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*
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* After we have input two buffers, we change the output format
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* from S8 to S32, then push a last buffer.
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*
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* This makes audioaggregator convert its "half-mixed" current_buffer,
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* we can then ensure that the second output buffer is as expected.
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*/
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GST_START_TEST (test_change_output_caps_mid_output_buffer)
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{
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GstSegment segment;
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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)
|
||||
{
|
||||
|
|
@ -1876,6 +2011,7 @@ 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
|
||||
|
|
|
|||
Loading…
Reference in a new issue