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gstreamer/gst-libs/gst/audio/gstaudioaggregator.c
Olivier Crête e3be1b8490 audio aggregator: Add QoS property to pad 
Add a property to emit a QoS message whenever any data is dropped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
2021-07-08 23:01:13 -04:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
* 2013 Sebastian Dröge <sebastian@centricular.com>
* 2014 Collabora
* Olivier Crete <olivier.crete@collabora.com>
*
* gstaudioaggregator.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION: gstaudioaggregator
* @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, #GstAudioMixer
*
* 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.
*
* #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 #GstAudioAggregatorPadClass.convert_buffer.
*
* To allow for the output caps to change, the mechanism is the same as
* above, with the GType of the source pad.
*
* See #GstAudioMixer for an example.
*
* When conversion is enabled, #GstAudioAggregator will accept
* any type of raw audio caps and perform conversion
* on the data arriving on its sink pads, with whatever downstream
* expects as the target format.
*
* In case downstream caps are not fully fixated, it will use
* the first configured sink pad to finish fixating its source pad
* caps.
*
* 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
* downstream specifies a range or a set of acceptable rates).
*
* The #GstAggregator::samples-selected signal is provided with some
* additional information about the output buffer:
* - "offset" G_TYPE_UINT64 Offset in samples since segment start
* for the position that is next to be filled in the output buffer.
* - "frames" G_TYPE_UINT Number of frames per output buffer.
*
* In addition the gst_aggregator_peek_next_sample() function returns
* additional information in the info #GstStructure of the returned sample:
* - "output-offset" G_TYPE_UINT64 Sample offset in output segment relative to
* the output segment's start where the current position of this input
* buffer would be placed
* - "position" G_TYPE_UINT current position in the input buffer in samples
* - "size" G_TYPE_UINT size of the input buffer in samples
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstaudioaggregator.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
#define GST_CAT_DEFAULT audio_aggregator_debug
enum
{
PROP_PAD_0,
PROP_PAD_QOS_MESSAGES,
};
struct _GstAudioAggregatorPadPrivate
{
/* All members are protected by the pad object lock */
GstBuffer *buffer; /* current buffer we're mixing, for
comparison with a new input buffer from
aggregator to see if we need to update our
cached values. */
guint position, size; /* position in the input buffer and size of the
input buffer in number of samples */
guint64 output_offset; /* Sample offset in output segment relative to
srcpad.segment.start where the current position
of this input_buffer would be placed. */
guint64 next_offset; /* Next expected sample offset relative to
pad.segment.start */
/* Last time we noticed a discont */
GstClockTime discont_time;
/* A new unhandled segment event has been received */
gboolean new_segment;
gboolean qos_messages; /* Property to decide to send QoS messages or not */
};
/*****************************************
* GstAudioAggregatorPad implementation *
*****************************************/
G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorPad, gst_audio_aggregator_pad,
GST_TYPE_AGGREGATOR_PAD);
static GstFlowReturn
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
GstAggregator * aggregator);
static void
gst_audio_aggregator_pad_finalize (GObject * object)
{
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object;
gst_buffer_replace (&pad->priv->buffer, NULL);
G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object);
}
static void
gst_audio_aggregator_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object);
switch (prop_id) {
case PROP_PAD_QOS_MESSAGES:
GST_OBJECT_LOCK (pad);
g_value_set_boolean (value, pad->priv->qos_messages);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object);
switch (prop_id) {
case PROP_PAD_QOS_MESSAGES:
GST_OBJECT_LOCK (pad);
pad->priv->qos_messages = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;
gobject_class->set_property = gst_audio_aggregator_pad_set_property;
gobject_class->get_property = gst_audio_aggregator_pad_get_property;
gobject_class->finalize = gst_audio_aggregator_pad_finalize;
aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad);
/**
* GstAudioAggregatorPad:qos-messages:
*
* Emit QoS messages when dropping buffers.
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class,
PROP_PAD_QOS_MESSAGES, g_param_spec_boolean ("qos-messages",
"Quality of Service Messages",
"Emit QoS messages when dropping buffers", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad)
{
pad->priv = gst_audio_aggregator_pad_get_instance_private (pad);
gst_audio_info_init (&pad->info);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
static GstFlowReturn
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
GstAggregator * aggregator)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (aggpad);
pad->priv->position = pad->priv->size = 0;
pad->priv->output_offset = pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_OBJECT_UNLOCK (aggpad);
return GST_FLOW_OK;
}
enum
{
PROP_CONVERT_PAD_0,
PROP_CONVERT_PAD_CONVERTER_CONFIG
};
struct _GstAudioAggregatorConvertPadPrivate
{
/* All members are protected by the pad object lock */
GstAudioConverter *converter;
GstStructure *converter_config;
gboolean converter_config_changed;
};
G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorConvertPad,
gst_audio_aggregator_convert_pad, GST_TYPE_AUDIO_AGGREGATOR_PAD);
static void
gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
* aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info)
{
GstStructure *config = aaggcpad->priv->converter_config;
GstAudioConverter *converter;
if (!aaggcpad->priv->converter_config_changed)
return;
g_clear_pointer (&aaggcpad->priv->converter, gst_audio_converter_free);
aaggcpad->priv->converter_config_changed = FALSE;
if (in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) {
/* If we haven't received caps yet, this pad should not have
* a buffer to convert anyway */
return;
}
converter =
gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE, in_info, out_info,
config ? gst_structure_copy (config) : NULL);
if (converter == NULL) {
/* FIXME: Not converting when we need to but the config is invalid (e.g.
* because the mix-matrix is not the right size) produces garbage. An
* invalid config should cause a GST_FLOW_NOT_NEGOTIATED. */
return;
}
if (!gst_audio_converter_is_passthrough (converter))
aaggcpad->priv->converter = converter;
else
gst_audio_converter_free (converter);
}
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 (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);
if (aaggcpad->priv->converter) {
gint insize = gst_buffer_get_size (input_buffer);
gsize insamples = insize / in_info->bpf;
gsize outsamples =
gst_audio_converter_get_out_frames (aaggcpad->priv->converter,
insamples);
gint outsize = outsamples * out_info->bpf;
GstMapInfo inmap, outmap;
res = gst_buffer_new_allocate (NULL, outsize, NULL);
/* We create a perfectly similar buffer, except obviously for
* its converted contents */
gst_buffer_copy_into (res, input_buffer,
GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
GST_BUFFER_COPY_META, 0, -1);
gst_buffer_map (input_buffer, &inmap, GST_MAP_READ);
gst_buffer_map (res, &outmap, GST_MAP_WRITE);
gst_audio_converter_samples (aaggcpad->priv->converter,
GST_AUDIO_CONVERTER_FLAG_NONE,
(gpointer *) & inmap.data, insamples,
(gpointer *) & outmap.data, outsamples);
gst_buffer_unmap (input_buffer, &inmap);
gst_buffer_unmap (res, &outmap);
} else {
res = gst_buffer_ref (input_buffer);
}
return res;
}
static void
gst_audio_aggregator_convert_pad_finalize (GObject * object)
{
GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object;
if (pad->priv->converter)
gst_audio_converter_free (pad->priv->converter);
if (pad->priv->converter_config)
gst_structure_free (pad->priv->converter_config);
G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize
(object);
}
static void
gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
switch (prop_id) {
case PROP_CONVERT_PAD_CONVERTER_CONFIG:
GST_OBJECT_LOCK (pad);
if (pad->priv->converter_config)
g_value_set_boxed (value, pad->priv->converter_config);
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
switch (prop_id) {
case PROP_CONVERT_PAD_CONVERTER_CONFIG:
GST_OBJECT_LOCK (pad);
if (pad->priv->converter_config)
gst_structure_free (pad->priv->converter_config);
pad->priv->converter_config = g_value_dup_boxed (value);
pad->priv->converter_config_changed = TRUE;
GST_OBJECT_UNLOCK (pad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAudioAggregatorPadClass *aaggpad_class =
(GstAudioAggregatorPadClass *) klass;
gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property;
gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property;
g_object_class_install_property (gobject_class,
PROP_CONVERT_PAD_CONVERTER_CONFIG,
g_param_spec_boxed ("converter-config", "Converter configuration",
"A GstStructure describing the configuration that should be used "
"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 void
gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad)
{
pad->priv = gst_audio_aggregator_convert_pad_get_instance_private (pad);
}
/**************************************
* GstAudioAggregator implementation *
**************************************/
struct _GstAudioAggregatorPrivate
{
GMutex mutex;
/* All three properties are unprotected, can't be modified while streaming */
/* Size in frames that is output per buffer */
GstClockTime alignment_threshold;
GstClockTime discont_wait;
gint output_buffer_duration_n;
gint output_buffer_duration_d;
guint samples_per_buffer;
guint error_per_buffer;
guint accumulated_error;
guint current_blocksize;
/* Protected by srcpad stream clock */
/* Output buffer starting at offset containing blocksize frames (calculated
* from output_buffer_duration) */
GstBuffer *current_buffer;
/* counters to keep track of timestamps */
/* Readable with object lock, writable with both aag lock and object lock */
/* Sample offset starting from 0 at aggregator.segment.start */
gint64 offset;
/* info structure passed to selected-samples signal, must only be accessed
* from the aggregate thread */
GstStructure *selected_samples_info;
};
#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);
#define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex);
static void gst_audio_aggregator_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_dispose (GObject * object);
static gboolean gst_audio_aggregator_src_event (GstAggregator * agg,
GstEvent * event);
static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg,
GstAggregatorPad * aggpad, GstEvent * event);
static gboolean gst_audio_aggregator_src_query (GstAggregator * agg,
GstQuery * query);
static gboolean
gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
GstQuery * query);
static gboolean gst_audio_aggregator_start (GstAggregator * agg);
static gboolean gst_audio_aggregator_stop (GstAggregator * agg);
static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg);
static GstBuffer *gst_audio_aggregator_create_output_buffer (GstAudioAggregator
* aagg, guint num_frames);
static GstBuffer *gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer);
static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg,
gboolean timeout);
static gboolean sync_pad_values (GstElement * aagg, GstPad * pad, gpointer ud);
static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg,
GstCaps * caps);
static GstFlowReturn
gst_audio_aggregator_update_src_caps (GstAggregator * agg,
GstCaps * caps, GstCaps ** ret);
static GstCaps *gst_audio_aggregator_fixate_src_caps (GstAggregator * agg,
GstCaps * caps);
static GstSample *gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
GstAggregatorPad * aggpad);
#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
#define DEFAULT_OUTPUT_BUFFER_DURATION_D (100)
enum
{
PROP_0,
PROP_OUTPUT_BUFFER_DURATION,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_OUTPUT_BUFFER_DURATION_FRACTION,
};
G_DEFINE_ABSTRACT_TYPE_WITH_PRIVATE (GstAudioAggregator, gst_audio_aggregator,
GST_TYPE_AGGREGATOR);
static GstBuffer *
gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
{
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 (aaggpad, in_info, out_info, buffer);
}
static void
gst_audio_aggregator_translate_output_buffer_duration (GstAudioAggregator *
aagg, GstClockTime duration)
{
gint gcd;
aagg->priv->output_buffer_duration_n = duration;
aagg->priv->output_buffer_duration_d = GST_SECOND;
gcd = gst_util_greatest_common_divisor (aagg->priv->output_buffer_duration_n,
aagg->priv->output_buffer_duration_d);
if (gcd) {
aagg->priv->output_buffer_duration_n /= gcd;
aagg->priv->output_buffer_duration_d /= gcd;
}
}
static gboolean
gst_audio_aggregator_update_samples_per_buffer (GstAudioAggregator * aagg)
{
gboolean ret = TRUE;
GstAudioAggregatorPad *srcpad =
GST_AUDIO_AGGREGATOR_PAD (GST_AGGREGATOR_SRC_PAD (aagg));
if (!srcpad->info.finfo
|| GST_AUDIO_INFO_FORMAT (&srcpad->info) == GST_AUDIO_FORMAT_UNKNOWN) {
ret = FALSE;
goto out;
}
aagg->priv->samples_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
aagg->priv->output_buffer_duration_n) /
aagg->priv->output_buffer_duration_d;
if (aagg->priv->samples_per_buffer == 0) {
ret = FALSE;
goto out;
}
aagg->priv->error_per_buffer =
(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
aagg->priv->output_buffer_duration_n) %
aagg->priv->output_buffer_duration_d;
aagg->priv->accumulated_error = 0;
GST_DEBUG_OBJECT (aagg, "Buffer duration: %u/%u",
aagg->priv->output_buffer_duration_n,
aagg->priv->output_buffer_duration_d);
GST_DEBUG_OBJECT (aagg, "Samples per buffer: %u (error: %u/%u)",
aagg->priv->samples_per_buffer, aagg->priv->error_per_buffer,
aagg->priv->output_buffer_duration_d);
out:
return ret;
}
static void
gst_audio_aggregator_recalculate_latency (GstAudioAggregator * aagg)
{
guint64 latency = gst_util_uint64_scale_int (GST_SECOND,
aagg->priv->output_buffer_duration_n,
aagg->priv->output_buffer_duration_d);
gst_aggregator_set_latency (GST_AGGREGATOR (aagg), latency, latency);
GST_OBJECT_LOCK (aagg);
/* Force recalculating in aggregate */
aagg->priv->samples_per_buffer = 0;
GST_OBJECT_UNLOCK (aagg);
}
static void
gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass;
gobject_class->set_property = gst_audio_aggregator_set_property;
gobject_class->get_property = gst_audio_aggregator_get_property;
gobject_class->dispose = gst_audio_aggregator_dispose;
gstaggregator_class->src_event =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event);
gstaggregator_class->sink_event =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event);
gstaggregator_class->src_query =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query);
gstaggregator_class->sink_query = gst_audio_aggregator_sink_query;
gstaggregator_class->start = gst_audio_aggregator_start;
gstaggregator_class->stop = gst_audio_aggregator_stop;
gstaggregator_class->flush = gst_audio_aggregator_flush;
gstaggregator_class->aggregate =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate);
gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip);
gstaggregator_class->get_next_time = gst_aggregator_simple_get_next_time;
gstaggregator_class->update_src_caps =
GST_DEBUG_FUNCPTR (gst_audio_aggregator_update_src_caps);
gstaggregator_class->fixate_src_caps = gst_audio_aggregator_fixate_src_caps;
gstaggregator_class->negotiated_src_caps =
gst_audio_aggregator_negotiated_src_caps;
gstaggregator_class->peek_next_sample = gst_audio_aggregator_peek_next_sample;
klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration",
"Output block size in nanoseconds", 1,
G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioAggregator:output-buffer-duration-fraction:
*
* Output block size in nanoseconds, expressed as a fraction.
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class,
PROP_OUTPUT_BUFFER_DURATION_FRACTION,
gst_param_spec_fraction ("output-buffer-duration-fraction",
"Output buffer duration fraction",
"Output block size in nanoseconds, expressed as a fraction", 1,
G_MAXINT, G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
DEFAULT_OUTPUT_BUFFER_DURATION_D,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 0,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_audio_aggregator_init (GstAudioAggregator * aagg)
{
aagg->priv = gst_audio_aggregator_get_instance_private (aagg);
g_mutex_init (&aagg->priv->mutex);
aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
gst_audio_aggregator_translate_output_buffer_duration (aagg,
DEFAULT_OUTPUT_BUFFER_DURATION);
gst_audio_aggregator_recalculate_latency (aagg);
aagg->current_caps = NULL;
aagg->priv->selected_samples_info =
gst_structure_new_empty ("GstAudioAggregatorSelectedSamplesInfo");
}
static void
gst_audio_aggregator_dispose (GObject * object)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
gst_caps_replace (&aagg->current_caps, NULL);
gst_clear_structure (&aagg->priv->selected_samples_info);
g_mutex_clear (&aagg->priv->mutex);
G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object);
}
static void
gst_audio_aggregator_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
switch (prop_id) {
case PROP_OUTPUT_BUFFER_DURATION:
gst_audio_aggregator_translate_output_buffer_duration (aagg,
g_value_get_uint64 (value));
g_object_notify (object, "output-buffer-duration-fraction");
gst_audio_aggregator_recalculate_latency (aagg);
break;
case PROP_ALIGNMENT_THRESHOLD:
aagg->priv->alignment_threshold = g_value_get_uint64 (value);
break;
case PROP_DISCONT_WAIT:
aagg->priv->discont_wait = g_value_get_uint64 (value);
break;
case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
aagg->priv->output_buffer_duration_n =
gst_value_get_fraction_numerator (value);
aagg->priv->output_buffer_duration_d =
gst_value_get_fraction_denominator (value);
g_object_notify (object, "output-buffer-duration");
gst_audio_aggregator_recalculate_latency (aagg);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_aggregator_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
switch (prop_id) {
case PROP_OUTPUT_BUFFER_DURATION:
g_value_set_uint64 (value, gst_util_uint64_scale_int (GST_SECOND,
aagg->priv->output_buffer_duration_n,
aagg->priv->output_buffer_duration_d));
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, aagg->priv->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, aagg->priv->discont_wait);
break;
case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
gst_value_set_fraction (value, aagg->priv->output_buffer_duration_n,
aagg->priv->output_buffer_duration_d);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* Caps negotiation */
/* Unref after usage */
static GstAudioAggregatorPad *
gst_audio_aggregator_get_first_configured_pad (GstAggregator * agg)
{
GstAudioAggregatorPad *res = NULL;
GList *l;
GST_OBJECT_LOCK (agg);
for (l = GST_ELEMENT (agg)->sinkpads; l; l = l->next) {
GstAudioAggregatorPad *aaggpad = l->data;
if (GST_AUDIO_INFO_FORMAT (&aaggpad->info) != GST_AUDIO_FORMAT_UNKNOWN) {
res = gst_object_ref (aaggpad);
break;
}
}
GST_OBJECT_UNLOCK (agg);
return res;
}
static GstCaps *
gst_audio_aggregator_sink_getcaps (GstPad * pad, GstAggregator * agg,
GstCaps * filter)
{
GstAudioAggregatorPad *first_configured_pad =
gst_audio_aggregator_get_first_configured_pad (agg);
GstCaps *sink_template_caps = gst_pad_get_pad_template_caps (pad);
GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
GstCaps *sink_caps;
GST_INFO_OBJECT (pad, "Getting caps with filter %" GST_PTR_FORMAT, filter);
GST_DEBUG_OBJECT (pad, "sink template caps : %" GST_PTR_FORMAT,
sink_template_caps);
GST_DEBUG_OBJECT (pad, "downstream caps %" GST_PTR_FORMAT, downstream_caps);
/* If we already have a configured pad, assume that we can only configure
* to the very same format filtered with the template caps and continue
* with the result of that as the template caps */
if (first_configured_pad) {
GstCaps *first_configured_caps =
gst_audio_info_to_caps (&first_configured_pad->info);
GstCaps *tmp;
tmp =
gst_caps_intersect_full (sink_template_caps, first_configured_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (first_configured_caps);
gst_caps_unref (sink_template_caps);
sink_template_caps = tmp;
gst_object_unref (first_configured_pad);
}
/* If we have downstream caps, filter them against our template caps or
* the filtered first configured pad caps from above */
if (downstream_caps) {
sink_caps =
gst_caps_intersect_full (sink_template_caps, downstream_caps,
GST_CAPS_INTERSECT_FIRST);
} else {
sink_caps = gst_caps_ref (sink_template_caps);
}
if (filter) {
GstCaps *tmp = gst_caps_intersect_full (sink_caps, filter,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (sink_caps);
sink_caps = tmp;
}
gst_caps_unref (sink_template_caps);
if (downstream_caps)
gst_caps_unref (downstream_caps);
GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps);
return sink_caps;
}
static GstCaps *
gst_audio_aggregator_convert_sink_getcaps (GstPad * pad, GstAggregator * agg,
GstCaps * filter)
{
GstAudioAggregatorPad *first_configured_pad =
gst_audio_aggregator_get_first_configured_pad (agg);
GstCaps *sink_template_caps = gst_pad_get_pad_template_caps (pad);
GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
GstCaps *sink_caps;
GST_INFO_OBJECT (pad, "Getting caps with filter %" GST_PTR_FORMAT, filter);
GST_DEBUG_OBJECT (pad, "sink template caps : %" GST_PTR_FORMAT,
sink_template_caps);
GST_DEBUG_OBJECT (pad, "downstream caps %" GST_PTR_FORMAT, downstream_caps);
/* We can convert between all formats except for the sample rate, which has
* to match. */
/* If we have a first configured pad, we can only convert everything except
* for the sample rate, so modify our template caps to have exactly that
* sample rate in all structures */
if (first_configured_pad) {
GST_INFO_OBJECT (pad, "first configured pad has sample rate %d",
first_configured_pad->info.rate);
sink_template_caps = gst_caps_make_writable (sink_template_caps);
gst_caps_set_simple (sink_template_caps, "rate", G_TYPE_INT,
first_configured_pad->info.rate, NULL);
gst_object_unref (first_configured_pad);
}
/* Now if we have downstream caps, filter against the template caps from
* above, i.e. with potentially fixated sample rate field already. This
* filters out any structures with unsupported rates.
*
* Afterwards we create new caps that only take over the rate fields of the
* remaining downstream caps, and filter that against the plain template
* caps to get the resulting allowed caps with conversion for everything but
* the rate */
if (downstream_caps) {
GstCaps *tmp;
guint i, n;
tmp =
gst_caps_intersect_full (sink_template_caps, downstream_caps,
GST_CAPS_INTERSECT_FIRST);
n = gst_caps_get_size (tmp);
sink_caps = gst_caps_new_empty ();
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (tmp, i);
GstStructure *new_s =
gst_structure_new_empty (gst_structure_get_name (s));
gst_structure_set_value (new_s, "rate", gst_structure_get_value (s,
"rate"));
sink_caps = gst_caps_merge_structure (sink_caps, new_s);
}
gst_caps_unref (tmp);
tmp = sink_caps;
sink_caps =
gst_caps_intersect_full (sink_template_caps, tmp,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (tmp);
} else {
sink_caps = gst_caps_ref (sink_template_caps);
}
/* And finally filter anything that remains against the filter caps */
if (filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, sink_caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (sink_caps);
sink_caps = tmp;
}
GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps);
gst_caps_unref (sink_template_caps);
if (downstream_caps)
gst_caps_unref (downstream_caps);
return sink_caps;
}
static gboolean
gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
GstAggregator * agg, GstCaps * caps)
{
GstAudioAggregatorPad *first_configured_pad =
gst_audio_aggregator_get_first_configured_pad (agg);
GstAudioInfo info;
gboolean ret = TRUE;
gboolean downstream_supports_rate = TRUE;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (aaggpad, "Rejecting invalid caps: %" GST_PTR_FORMAT,
caps);
return FALSE;
}
/* TODO: handle different rates on sinkpads, a bit complex
* because offsets will have to be updated, and audio resampling
* has a latency to take into account
*/
/* Only check against the downstream caps if we didn't configure any caps
* so far. Otherwise we already know that downstream supports the rate
* because we negotiated with downstream */
if (!first_configured_pad) {
GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
/* Returns NULL if there is no downstream peer */
if (downstream_caps) {
GstCaps *rate_caps =
gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, info.rate,
NULL);
gst_caps_set_features_simple (rate_caps,
gst_caps_features_copy (GST_CAPS_FEATURES_ANY));
downstream_supports_rate =
gst_caps_can_intersect (rate_caps, downstream_caps);
gst_caps_unref (rate_caps);
gst_caps_unref (downstream_caps);
}
}
if (!downstream_supports_rate || (first_configured_pad
&& info.rate != first_configured_pad->info.rate)) {
GST_WARNING_OBJECT (aaggpad,
"Sample rate %d can't be configured (downstream supported: %d, configured rate: %d)",
info.rate, downstream_supports_rate,
first_configured_pad ? first_configured_pad->info.rate : 0);
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);
aaggpad->info = info;
if (klass->update_conversion_info)
klass->update_conversion_info (aaggpad);
GST_OBJECT_UNLOCK (aaggpad);
}
if (first_configured_pad)
gst_object_unref (first_configured_pad);
return ret;
}
static GstFlowReturn
gst_audio_aggregator_update_src_caps (GstAggregator * agg,
GstCaps * caps, GstCaps ** ret)
{
GstCaps *src_template_caps = gst_pad_get_pad_template_caps (agg->srcpad);
GstCaps *downstream_caps =
gst_pad_peer_query_caps (agg->srcpad, src_template_caps);
gst_caps_unref (src_template_caps);
*ret = gst_caps_intersect (caps, downstream_caps);
GST_INFO ("Updated src caps to %" GST_PTR_FORMAT, *ret);
if (downstream_caps)
gst_caps_unref (downstream_caps);
return GST_FLOW_OK;
}
/* At that point if the caps are not fixed, this means downstream
* didn't have fully specified requirements, we'll just go ahead
* and fixate raw audio fields using our first configured pad, we don't for
* now need a more complicated heuristic
*/
static GstCaps *
gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
{
GstAudioAggregatorPad *first_configured_pad = NULL;
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
first_configured_pad = gst_audio_aggregator_get_first_configured_pad (agg);
caps = gst_caps_make_writable (caps);
if (first_configured_pad) {
GstStructure *s, *s2;
GstCaps *first_configured_caps =
gst_audio_info_to_caps (&first_configured_pad->info);
gint first_configured_rate, first_configured_channels;
gint channels;
s = gst_caps_get_structure (caps, 0);
s2 = gst_caps_get_structure (first_configured_caps, 0);
gst_structure_get_int (s2, "rate", &first_configured_rate);
gst_structure_get_int (s2, "channels", &first_configured_channels);
gst_structure_fixate_field_string (s, "format",
gst_structure_get_string (s2, "format"));
gst_structure_fixate_field_string (s, "layout",
gst_structure_get_string (s2, "layout"));
gst_structure_fixate_field_nearest_int (s, "rate", first_configured_rate);
gst_structure_fixate_field_nearest_int (s, "channels",
first_configured_channels);
gst_structure_get_int (s, "channels", &channels);
if (!gst_structure_has_field (s, "channel-mask") && channels > 2) {
guint64 mask;
if (!gst_structure_get (s2, "channel-mask", GST_TYPE_BITMASK, &mask,
NULL)) {
mask = gst_audio_channel_get_fallback_mask (channels);
}
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
}
gst_caps_unref (first_configured_caps);
gst_object_unref (first_configured_pad);
} else {
GstStructure *s;
gint channels;
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", GST_AUDIO_DEF_RATE);
gst_structure_fixate_field_string (s, "format", GST_AUDIO_NE ("S16"));
gst_structure_fixate_field_string (s, "layout", "interleaved");
gst_structure_fixate_field_nearest_int (s, "channels", 2);
if (gst_structure_get_int (s, "channels", &channels) && channels > 2) {
if (!gst_structure_has_field_typed (s, "channel-mask", GST_TYPE_BITMASK))
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, 0ULL, NULL);
}
}
if (!gst_caps_is_fixed (caps))
caps = gst_caps_fixate (caps);
GST_INFO_OBJECT (agg, "Fixated src caps to %" GST_PTR_FORMAT, caps);
return caps;
}
/* Must be called with OBJECT_LOCK taken */
static void
gst_audio_aggregator_update_converters (GstAudioAggregator * aagg,
GstAudioInfo * new_info, GstAudioInfo * old_info)
{
GList *l;
for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
GstAudioAggregatorPad *aaggpad = l->data;
GstAudioAggregatorPadClass *klass =
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
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 */
if (aaggpad->priv->buffer) {
GstBuffer *new_converted_buffer =
gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad),
old_info, new_info, aaggpad->priv->buffer);
gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer);
gst_buffer_unref (new_converted_buffer);
}
}
}
/* We now have our final output caps, we can create the required converters */
static gboolean
gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GstAudioInfo info;
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
return FALSE;
}
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
if (!gst_audio_info_is_equal (&info, &srcpad->info)) {
GstAudioInfo old_info = srcpad->info;
GstAudioAggregatorPadClass *srcpad_klass =
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
gst_caps_replace (&aagg->current_caps, caps);
if (old_info.rate != info.rate)
aagg->priv->offset = -1;
memcpy (&srcpad->info, &info, sizeof (info));
gst_audio_aggregator_update_converters (aagg, &info, &old_info);
if (srcpad_klass->update_conversion_info)
srcpad_klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->
srcpad));
if (aagg->priv->current_buffer) {
GstBuffer *converted;
converted =
gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &old_info,
&info, aagg->priv->current_buffer);
gst_buffer_unref (aagg->priv->current_buffer);
aagg->priv->current_buffer = converted;
}
/* Force recalculating in aggregate */
aagg->priv->samples_per_buffer = 0;
}
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return
GST_AGGREGATOR_CLASS
(gst_audio_aggregator_parent_class)->negotiated_src_caps (agg, caps);
}
/* event handling */
static gboolean
gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event)
{
gboolean result;
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GST_DEBUG_OBJECT (agg->srcpad, "Got %s event on src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_QOS:
/* QoS might be tricky */
gst_event_unref (event);
return FALSE;
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
gst_event_unref (event);
return FALSE;
break;
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
gdouble rate;
GstSeekType start_type, stop_type;
gint64 start, stop;
GstFormat seek_format, dest_format;
/* parse the seek parameters */
gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
&start, &stop_type, &stop);
/* Check the seeking parameters before linking up */
if ((start_type != GST_SEEK_TYPE_NONE)
&& (start_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (aagg,
"seeking failed, unhandled seek type for start: %d", start_type);
goto done;
}
if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (aagg,
"seeking failed, unhandled seek type for end: %d", stop_type);
goto done;
}
GST_OBJECT_LOCK (agg);
dest_format = GST_AGGREGATOR_PAD (agg->srcpad)->segment.format;
GST_OBJECT_UNLOCK (agg);
if (seek_format != dest_format) {
result = FALSE;
GST_DEBUG_OBJECT (aagg,
"seeking failed, unhandled seek format: %s",
gst_format_get_name (seek_format));
goto done;
}
}
break;
default:
break;
}
return
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg,
event);
done:
return result;
}
static gboolean
gst_audio_aggregator_sink_event (GstAggregator * agg,
GstAggregatorPad * aggpad, GstEvent * event)
{
GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
gboolean res = TRUE;
GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->format != GST_FORMAT_TIME) {
GST_ERROR_OBJECT (agg, "Segment of type %s are not supported,"
" only TIME segments are supported",
gst_format_get_name (segment->format));
gst_event_unref (event);
event = NULL;
res = FALSE;
break;
}
GST_OBJECT_LOCK (agg);
if (segment->rate != GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate) {
GST_ERROR_OBJECT (aggpad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
} else if (segment->rate < 0.0) {
GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet");
res = FALSE;
gst_event_unref (event);
event = NULL;
} else {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (pad);
pad->priv->new_segment = TRUE;
GST_OBJECT_UNLOCK (pad);
}
GST_OBJECT_UNLOCK (agg);
break;
}
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
GST_INFO_OBJECT (aggpad, "Got caps %" GST_PTR_FORMAT, caps);
res = gst_audio_aggregator_sink_setcaps (aaggpad, agg, caps);
gst_event_unref (event);
event = NULL;
break;
}
default:
break;
}
if (!res) {
if (event)
gst_event_unref (event);
return res;
}
if (event != NULL)
return
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event
(agg, aggpad, event);
return res;
}
static gboolean
gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
GstQuery * query)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aggpad)) {
caps =
gst_audio_aggregator_convert_sink_getcaps (GST_PAD (aggpad), agg,
filter);
} else {
caps =
gst_audio_aggregator_sink_getcaps (GST_PAD (aggpad), agg, filter);
}
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
res =
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_query
(agg, aggpad, query);
break;
}
return res;
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* each other which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_audio_aggregator_query_duration (GstAudioAggregator * aagg,
GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
res = gst_audio_aggregator_query_duration (aagg, query);
break;
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
GST_OBJECT_LOCK (aagg);
switch (format) {
case GST_FORMAT_TIME:
gst_query_set_position (query, format,
gst_segment_to_stream_time (&GST_AGGREGATOR_PAD (agg->srcpad)->
segment, GST_FORMAT_TIME,
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position));
res = TRUE;
break;
case GST_FORMAT_BYTES:
if (GST_AUDIO_INFO_BPF (&srcpad->info)) {
gst_query_set_position (query, format, aagg->priv->offset *
GST_AUDIO_INFO_BPF (&srcpad->info));
res = TRUE;
}
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, aagg->priv->offset);
res = TRUE;
break;
default:
break;
}
GST_OBJECT_UNLOCK (aagg);
break;
}
default:
res =
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query
(agg, query);
break;
}
return res;
}
void
gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstCaps * caps)
{
#ifndef G_DISABLE_ASSERT
gboolean valid;
GST_OBJECT_LOCK (pad);
valid = gst_audio_info_from_caps (&pad->info, caps);
g_assert (valid);
GST_OBJECT_UNLOCK (pad);
#else
GST_OBJECT_LOCK (pad);
(void) gst_audio_info_from_caps (&pad->info, caps);
GST_OBJECT_UNLOCK (pad);
#endif
}
/* Must hold object lock and aagg lock to call */
static void
gst_audio_aggregator_reset (GstAudioAggregator * aagg)
{
GstAggregator *agg = GST_AGGREGATOR (aagg);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
aagg->priv->offset = -1;
gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
gst_caps_replace (&aagg->current_caps, NULL);
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
aagg->priv->accumulated_error = 0;
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
}
static gboolean
gst_audio_aggregator_start (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gst_audio_aggregator_reset (aagg);
return TRUE;
}
static gboolean
gst_audio_aggregator_stop (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gst_audio_aggregator_reset (aagg);
return TRUE;
}
static GstFlowReturn
gst_audio_aggregator_flush (GstAggregator * agg)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
aagg->priv->offset = -1;
aagg->priv->accumulated_error = 0;
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
static GstBuffer *
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
/* Guard against invalid audio info, we just don't clip here then */
if (!GST_AUDIO_INFO_IS_VALID (&pad->info))
return buffer;
GST_OBJECT_LOCK (bpad);
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return buffer;
}
/* Called with the object lock for both the element and pad held,
* as well as the aagg lock
*
* Replace the current buffer with input and update GstAudioAggregatorPadPrivate
* values.
*/
static gboolean
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad)
{
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
gint rate, bpf;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
} else {
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
}
pad->priv->position = 0;
pad->priv->size = gst_buffer_get_size (pad->priv->buffer) / bpf;
if (pad->priv->size == 0) {
if (!GST_BUFFER_DURATION_IS_VALID (pad->priv->buffer) ||
!GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_GAP)) {
GST_WARNING_OBJECT (pad, "Dropping 0-sized buffer missing either a"
" duration or a GAP flag: %" GST_PTR_FORMAT, pad->priv->buffer);
return FALSE;
}
pad->priv->size =
gst_util_uint64_scale (GST_BUFFER_DURATION (pad->priv->buffer), rate,
GST_SECOND);
}
if (!GST_BUFFER_PTS_IS_VALID (pad->priv->buffer)) {
if (pad->priv->output_offset == -1)
pad->priv->output_offset = aagg->priv->offset;
if (pad->priv->next_offset == -1)
pad->priv->next_offset = pad->priv->size;
else
pad->priv->next_offset += pad->priv->size;
goto done;
}
start_time = GST_BUFFER_PTS (pad->priv->buffer);
end_time =
start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
rate);
/* Clipping should've ensured this */
g_assert (start_time >= aggpad->segment.start);
start_offset =
gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
GST_SECOND);
end_offset = start_offset + pad->priv->size;
if (GST_BUFFER_IS_DISCONT (pad->priv->buffer)
|| GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_RESYNC)
|| pad->priv->new_segment || pad->priv->next_offset == -1) {
discont = TRUE;
pad->priv->new_segment = FALSE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= pad->priv->next_offset)
diff = pad->priv->next_offset - start_offset;
else
diff = start_offset - pad->priv->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (aagg->priv->discont_wait > 0) {
if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
pad->priv->discont_time = start_time;
} else if (start_time - pad->priv->discont_time >=
aagg->priv->discont_wait) {
discont = TRUE;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (pad->priv->next_offset != -1)
GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
pad->priv->next_offset, start_offset);
pad->priv->next_offset = end_offset;
} else {
pad->priv->next_offset += pad->priv->size;
}
if (pad->priv->output_offset == -1 || discont) {
GstClockTime start_running_time;
GstClockTime end_running_time;
GstClockTime segment_pos;
guint64 start_output_offset = -1;
guint64 end_output_offset = -1;
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
start_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, start_time);
end_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, end_time);
/* Convert to position in the output segment */
segment_pos =
gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
start_running_time);
if (GST_CLOCK_TIME_IS_VALID (segment_pos))
start_output_offset =
gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
GST_SECOND);
segment_pos =
gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
end_running_time);
if (GST_CLOCK_TIME_IS_VALID (segment_pos))
end_output_offset =
gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
GST_SECOND);
if (start_output_offset == -1 && end_output_offset == -1) {
/* Outside output segment, drop */
pad->priv->position = 0;
pad->priv->size = 0;
GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
return FALSE;
}
/* Calculate end_output_offset if it was outside the output segment */
if (end_output_offset == -1)
end_output_offset = start_output_offset + pad->priv->size;
if (end_output_offset < aagg->priv->offset) {
pad->priv->position = 0;
pad->priv->size = 0;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
if (start_output_offset == -1 ||
start_output_offset < aagg->priv->offset ||
(pad->priv->output_offset != -1 &&
start_output_offset < pad->priv->output_offset)) {
guint diff;
if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
diff = pad->priv->size - end_output_offset + aagg->priv->offset;
} else if (start_output_offset == -1) {
start_output_offset = end_output_offset - pad->priv->size;
if (start_output_offset < aagg->priv->offset)
diff = aagg->priv->offset - start_output_offset;
else
diff = 0;
} else if (pad->priv->output_offset != -1 &&
start_output_offset < pad->priv->output_offset) {
diff = pad->priv->output_offset - start_output_offset;
} else {
diff = aagg->priv->offset - start_output_offset;
}
pad->priv->position += diff;
if (pad->priv->position >= pad->priv->size) {
/* Empty buffer, drop */
pad->priv->position = 0;
pad->priv->size = 0;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
}
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
pad->priv->output_offset = aagg->priv->offset;
else if (pad->priv->output_offset != -1)
pad->priv->output_offset = MAX (pad->priv->output_offset,
start_output_offset);
else
pad->priv->output_offset = start_output_offset;
GST_DEBUG_OBJECT (pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current audio aggregator offset %" G_GINT64_FORMAT,
pad->priv->output_offset, aagg->priv->offset);
}
done:
GST_LOG_OBJECT (pad,
"Queued new buffer at offset %" G_GUINT64_FORMAT,
pad->priv->output_offset);
return TRUE;
}
/* Called with pad object lock held */
static gboolean
gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf,
guint blocksize)
{
guint overlap;
guint out_start;
gboolean filled;
guint in_offset;
gboolean pad_changed = FALSE;
/* Overlap => mix */
if (aagg->priv->offset < pad->priv->output_offset)
out_start = pad->priv->output_offset - aagg->priv->offset;
else
out_start = 0;
overlap = pad->priv->size - pad->priv->position;
if (overlap > blocksize - out_start)
overlap = blocksize - out_start;
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
/* skip gap buffer */
GST_LOG_OBJECT (pad, "skipping GAP buffer");
pad->priv->output_offset += pad->priv->size - pad->priv->position;
pad->priv->position = pad->priv->size;
gst_buffer_replace (&pad->priv->buffer, NULL);
return FALSE;
}
gst_buffer_ref (inbuf);
in_offset = pad->priv->position;
GST_OBJECT_UNLOCK (pad);
GST_OBJECT_UNLOCK (aagg);
filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg,
pad, inbuf, in_offset, outbuf, out_start, overlap);
GST_OBJECT_LOCK (aagg);
GST_OBJECT_LOCK (pad);
pad_changed = (inbuf != pad->priv->buffer);
gst_buffer_unref (inbuf);
if (filled)
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);
if (pad_changed)
return FALSE;
pad->priv->position += overlap;
pad->priv->output_offset += overlap;
if (pad->priv->position == pad->priv->size) {
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
return FALSE;
}
return TRUE;
}
static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
guint num_frames)
{
GstAllocator *allocator;
GstAllocationParams params;
GstBuffer *outbuf;
GstMapInfo outmap;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);
GST_DEBUG ("Creating output buffer with size %d",
num_frames * GST_AUDIO_INFO_BPF (&srcpad->info));
outbuf = gst_buffer_new_allocate (allocator, num_frames *
GST_AUDIO_INFO_BPF (&srcpad->info), &params);
if (allocator)
gst_object_unref (allocator);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_info_fill_silence (srcpad->info.finfo, outmap.data,
outmap.size);
gst_buffer_unmap (outbuf, &outmap);
return outbuf;
}
static gboolean
sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data)
{
GstAudioAggregatorPad *aapad = GST_AUDIO_AGGREGATOR_PAD (pad);
GstAggregatorPad *bpad = GST_AGGREGATOR_PAD_CAST (pad);
GstClockTime timestamp, stream_time;
if (aapad->priv->buffer == NULL)
return TRUE;
timestamp = GST_BUFFER_PTS (aapad->priv->buffer);
GST_OBJECT_LOCK (bpad);
stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
timestamp);
GST_OBJECT_UNLOCK (bpad);
/* sync object properties on stream time */
/* TODO: Ideally we would want to do that on every sample */
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT_CAST (pad), stream_time);
return TRUE;
}
static GstSample *
gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
GstAggregatorPad * aggpad)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GstSample *sample = NULL;
if (pad->priv->buffer && pad->priv->output_offset >= aagg->priv->offset
&& pad->priv->output_offset <
aagg->priv->offset + aagg->priv->samples_per_buffer) {
GstCaps *caps = gst_pad_get_current_caps (GST_PAD (aggpad));
GstStructure *info =
gst_structure_new ("GstAudioAggregatorPadNextSampleInfo",
"output-offset", G_TYPE_UINT64, pad->priv->output_offset,
"position", G_TYPE_UINT, pad->priv->position,
"size", G_TYPE_UINT, pad->priv->size,
NULL);
sample = gst_sample_new (pad->priv->buffer, caps, &aggpad->segment, info);
gst_caps_unref (caps);
gst_structure_free (info);
}
return sample;
}
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
/* Calculate the current output offset/timestamp and offset_end/timestamp_end.
* Allocate a silence buffer for this and store it.
*
* For all pads:
* 1) Once per input buffer (cached)
* 1) Check discont (flag and timestamp with tolerance)
* 2) If discont or new, resync. That means:
* 1) Drop all start data of the buffer that comes before
* the current position/offset.
* 2) Calculate the offset (output segment!) that the first
* frame of the input buffer corresponds to. Base this on
* the running time.
*
* 2) If the current pad's offset/offset_end overlaps with the output
* offset/offset_end, mix it at the appropriate position in the output
* buffer and advance the pad's position. Remember if this pad needs
* a new buffer to advance behind the output offset_end.
*
* If we had no pad with a buffer, go EOS.
*
* If we had at least one pad that did not advance behind output
* offset_end, let aggregate be called again for the current
* output offset/offset_end.
*/
GstElement *element;
GstAudioAggregator *aagg;
GList *iter;
GstFlowReturn ret;
GstBuffer *outbuf = NULL;
gint64 next_offset;
gint64 next_timestamp;
gint rate, bpf;
gboolean dropped = FALSE;
gboolean is_eos = TRUE;
gboolean is_done = TRUE;
guint blocksize;
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
/* Sync pad properties to the stream time */
gst_element_foreach_sink_pad (element, sync_pad_values, NULL);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
if (aagg->priv->samples_per_buffer == 0) {
if (!gst_audio_aggregator_update_samples_per_buffer (aagg)) {
GST_ERROR_OBJECT (aagg,
"Failed to calculate the number of samples per buffer");
GST_OBJECT_UNLOCK (agg);
goto not_negotiated;
}
}
/* Update position from the segment start/stop if needed */
if (agg_segment->position == -1) {
if (agg_segment->rate > 0.0)
agg_segment->position = agg_segment->start;
else
agg_segment->position = agg_segment->stop;
}
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (timeout) {
GstClockTime output_buffer_duration;
GST_DEBUG_OBJECT (aagg,
"Got timeout before receiving any caps, don't output anything");
blocksize = aagg->priv->samples_per_buffer;
if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
aagg->priv->output_buffer_duration_d)
blocksize += 1;
aagg->priv->accumulated_error =
(aagg->priv->accumulated_error +
aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
output_buffer_duration =
gst_util_uint64_scale (blocksize, GST_SECOND, rate);
/* Advance position */
if (agg_segment->rate > 0.0)
agg_segment->position += output_buffer_duration;
else if (agg_segment->position > output_buffer_duration)
agg_segment->position -= output_buffer_duration;
else
agg_segment->position = 0;
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_AGGREGATOR_FLOW_NEED_DATA;
} else {
GST_OBJECT_UNLOCK (agg);
goto not_negotiated;
}
}
if (aagg->priv->offset == -1) {
aagg->priv->offset =
gst_util_uint64_scale (agg_segment->position - agg_segment->start, rate,
GST_SECOND);
GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
aagg->priv->offset);
}
if (aagg->priv->current_buffer == NULL) {
blocksize = aagg->priv->samples_per_buffer;
if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
aagg->priv->output_buffer_duration_d)
blocksize += 1;
aagg->priv->current_blocksize = blocksize;
aagg->priv->accumulated_error =
(aagg->priv->accumulated_error +
aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
GST_OBJECT_UNLOCK (agg);
aagg->priv->current_buffer =
GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
blocksize);
/* Be careful, some things could have changed ? */
GST_OBJECT_LOCK (agg);
GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
} else {
blocksize = aagg->priv->current_blocksize;
}
/* FIXME: Reverse mixing does not work at all yet */
if (agg_segment->rate > 0.0) {
next_offset = aagg->priv->offset + blocksize;
} else {
next_offset = aagg->priv->offset - blocksize;
}
/* Use the sample counter, which will never accumulate rounding errors */
next_timestamp =
agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
outbuf = aagg->priv->current_buffer;
GST_LOG_OBJECT (agg,
"Starting to mix %u samples for offset %" G_GINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, blocksize,
aagg->priv->offset, GST_TIME_ARGS (agg_segment->position));
for (iter = element->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);
GstBuffer *input_buffer;
if (!pad_eos)
is_eos = FALSE;
input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
GST_OBJECT_LOCK (pad);
if (!input_buffer) {
if (timeout) {
if (pad->priv->output_offset < next_offset) {
gint64 diff = next_offset - pad->priv->output_offset;
GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
" frames (%" GST_TIME_FORMAT ")", diff,
GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
GST_AUDIO_INFO_RATE (&srcpad->info))));
}
} else if (!pad_eos) {
is_done = FALSE;
}
GST_OBJECT_UNLOCK (pad);
continue;
} else if (!GST_AUDIO_INFO_IS_VALID (&pad->info)) {
GST_OBJECT_UNLOCK (pad);
GST_OBJECT_UNLOCK (agg);
goto not_negotiated;
}
/* New buffer? */
if (!pad->priv->buffer) {
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
pad->priv->buffer =
gst_audio_aggregator_convert_buffer
(aagg, GST_PAD (pad), &pad->info, &srcpad->info, input_buffer);
else
pad->priv->buffer = gst_buffer_ref (input_buffer);
if (!gst_audio_aggregator_fill_buffer (aagg, pad)) {
gst_buffer_replace (&pad->priv->buffer, NULL);
gst_buffer_unref (input_buffer);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
}
gst_buffer_unref (input_buffer);
if (!pad->priv->buffer && !dropped && pad_eos) {
GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
GST_OBJECT_UNLOCK (pad);
continue;
}
g_assert (pad->priv->buffer);
/* This pad is lagging behind, we need to update the offset
* and maybe drop the current buffer */
if (pad->priv->output_offset < aagg->priv->offset) {
gint64 diff = aagg->priv->offset - pad->priv->output_offset;
gint64 odiff = diff;
if (pad->priv->position + diff > pad->priv->size)
diff = pad->priv->size - pad->priv->position;
pad->priv->position += diff;
pad->priv->output_offset += diff;
if (pad->priv->position == pad->priv->size) {
GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
", dropping %" GST_PTR_FORMAT,
GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
GST_AUDIO_INFO_RATE (&srcpad->info))), pad->priv->buffer);
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
}
g_assert (pad->priv->buffer);
GST_OBJECT_UNLOCK (pad);
}
GST_OBJECT_UNLOCK (agg);
{
gst_structure_set (aagg->priv->selected_samples_info, "offset",
G_TYPE_UINT64, aagg->priv->offset, "frames", G_TYPE_UINT, blocksize,
NULL);
gst_aggregator_selected_samples (agg, agg_segment->position,
GST_CLOCK_TIME_NONE, next_timestamp - agg_segment->position,
aagg->priv->selected_samples_info);
}
GST_OBJECT_LOCK (agg);
for (iter = element->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
GST_OBJECT_LOCK (pad);
if (pad->priv->buffer && pad->priv->output_offset >= aagg->priv->offset
&& pad->priv->output_offset < aagg->priv->offset + blocksize) {
gboolean drop_buf;
GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
outbuf, blocksize);
if (pad->priv->output_offset >= next_offset) {
GST_LOG_OBJECT (pad,
"Pad is at or after current offset: %" G_GUINT64_FORMAT " >= %"
G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
} else {
is_done = FALSE;
}
if (drop_buf) {
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
}
GST_OBJECT_UNLOCK (pad);
}
GST_OBJECT_UNLOCK (agg);
if (dropped) {
/* We dropped a buffer, retry */
GST_LOG_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_AGGREGATOR_FLOW_NEED_DATA;
}
if (!is_done && !is_eos) {
/* Get more buffers */
GST_LOG_OBJECT (aagg,
"We're not done yet for the current offset, waiting for more data");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_AGGREGATOR_FLOW_NEED_DATA;
}
if (is_eos) {
gint64 max_offset = 0;
GST_DEBUG_OBJECT (aagg, "We're EOS");
GST_OBJECT_LOCK (agg);
for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset);
}
GST_OBJECT_UNLOCK (agg);
/* This means EOS or nothing mixed in at all */
if (aagg->priv->offset == max_offset) {
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_EOS;
}
if (max_offset <= next_offset) {
GST_DEBUG_OBJECT (aagg,
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
next_timestamp =
agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (next_offset > aagg->priv->offset)
gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf);
}
}
/* set timestamps on the output buffer */
GST_OBJECT_LOCK (agg);
if (agg_segment->rate > 0.0) {
GST_BUFFER_PTS (outbuf) = agg_segment->position;
GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
GST_BUFFER_DURATION (outbuf) = next_timestamp - agg_segment->position;
} else {
GST_BUFFER_PTS (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset;
GST_BUFFER_DURATION (outbuf) = agg_segment->position - next_timestamp;
}
GST_OBJECT_UNLOCK (agg);
/* send it out */
GST_LOG_OBJECT (aagg,
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
GST_BUFFER_OFFSET (outbuf));
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
ret = gst_aggregator_finish_buffer (agg, outbuf);
aagg->priv->current_buffer = NULL;
GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
aagg->priv->offset = next_offset;
agg_segment->position = next_timestamp;
/* If there was a timeout and there was a gap in data in out of the streams,
* then it's a very good time to for a resync with the timestamps.
*/
if (timeout) {
for (iter = element->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
GST_OBJECT_LOCK (pad);
if (pad->priv->output_offset < aagg->priv->offset)
pad->priv->output_offset = -1;
GST_OBJECT_UNLOCK (pad);
}
}
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return ret;
/* ERRORS */
not_negotiated:
{
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
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