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gstreamer/gst-libs/gst/audio/gstaudiodecoder.c

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/* GStreamer
* Copyright (C) 2009 Igalia S.L.
* Author: Iago Toral Quiroga <itoral@igalia.com>
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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:gstaudiodecoder
* @title: GstAudioDecoder
* @short_description: Base class for audio decoders
* @see_also: #GstBaseTransform
*
* This base class is for audio decoders turning encoded data into
* raw audio samples.
*
* GstAudioDecoder and subclass should cooperate as follows.
*
* ## Configuration
*
* * Initially, GstAudioDecoder calls @start when the decoder element
* is activated, which allows subclass to perform any global setup.
* Base class (context) parameters can already be set according to subclass
* capabilities (or possibly upon receive more information in subsequent
* @set_format).
* * GstAudioDecoder calls @set_format to inform subclass of the format
* of input audio data that it is about to receive.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
* * GstAudioDecoder calls @stop at end of all processing.
*
* As of configuration stage, and throughout processing, GstAudioDecoder
* provides various (context) parameters, e.g. describing the format of
* output audio data (valid when output caps have been set) or current parsing state.
* Conversely, subclass can and should configure context to inform
* base class of its expectation w.r.t. buffer handling.
*
* ## Data processing
* * Base class gathers input data, and optionally allows subclass
* to parse this into subsequently manageable (as defined by subclass)
* chunks. Such chunks are subsequently referred to as 'frames',
* though they may or may not correspond to 1 (or more) audio format frame.
* * Input frame is provided to subclass' @handle_frame.
* * If codec processing results in decoded data, subclass should call
* @gst_audio_decoder_finish_frame to have decoded data pushed
* downstream.
* * Just prior to actually pushing a buffer downstream,
* it is passed to @pre_push. Subclass should either use this callback
* to arrange for additional downstream pushing or otherwise ensure such
* custom pushing occurs after at least a method call has finished since
* setting src pad caps.
* * During the parsing process GstAudioDecoderClass will handle both
* srcpad and sinkpad events. Sink events will be passed to subclass
* if @event callback has been provided.
*
* ## Shutdown phase
*
* * GstAudioDecoder class calls @stop to inform the subclass that data
* parsing will be stopped.
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* needs to set the fixed caps on srcpad, when the format is ensured. This
* is typically when base class calls subclass' @set_format function, though
* it might be delayed until calling @gst_audio_decoder_finish_frame.
*
* In summary, above process should have subclass concentrating on
* codec data processing while leaving other matters to base class,
* such as most notably timestamp handling. While it may exert more control
* in this area (see e.g. @pre_push), it is very much not recommended.
*
* In particular, base class will try to arrange for perfect output timestamps
* as much as possible while tracking upstream timestamps.
* To this end, if deviation between the next ideal expected perfect timestamp
* and upstream exceeds #GstAudioDecoder:tolerance, then resync to upstream
* occurs (which would happen always if the tolerance mechanism is disabled).
*
* In non-live pipelines, baseclass can also (configurably) arrange for
* output buffer aggregation which may help to redue large(r) numbers of
* small(er) buffers being pushed and processed downstream. Note that this
* feature is only available if the buffer layout is interleaved. For planar
* buffers, the decoder implementation is fully responsible for the output
* buffer size.
*
* On the other hand, it should be noted that baseclass only provides limited
* seeking support (upon explicit subclass request), as full-fledged support
* should rather be left to upstream demuxer, parser or alike. This simple
* approach caters for seeking and duration reporting using estimated input
* bitrates.
*
* Things that subclass need to take care of:
*
* * Provide pad templates
* * Set source pad caps when appropriate
* * Set user-configurable properties to sane defaults for format and
* implementing codec at hand, and convey some subclass capabilities and
* expectations in context.
*
* * Accept data in @handle_frame and provide encoded results to
* @gst_audio_decoder_finish_frame. If it is prepared to perform
* PLC, it should also accept NULL data in @handle_frame and provide for
* data for indicated duration.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiodecoder.h"
#include "gstaudioutilsprivate.h"
#include <gst/pbutils/descriptions.h>
#include <string.h>
GST_DEBUG_CATEGORY (audiodecoder_debug);
#define GST_CAT_DEFAULT audiodecoder_debug
enum
{
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LATENCY,
PROP_TOLERANCE,
PROP_PLC,
PROP_MAX_ERRORS
};
#define DEFAULT_LATENCY 0
#define DEFAULT_TOLERANCE 0
#define DEFAULT_PLC FALSE
#define DEFAULT_DRAINABLE TRUE
#define DEFAULT_NEEDS_FORMAT FALSE
#define DEFAULT_MAX_ERRORS GST_AUDIO_DECODER_MAX_ERRORS
typedef struct _GstAudioDecoderContext
{
/* last negotiated input caps */
GstCaps *input_caps;
/* (output) audio format */
GstAudioInfo info;
GstCaps *caps;
gboolean output_format_changed;
/* parsing state */
gboolean eos;
gboolean sync;
gboolean had_output_data;
gboolean had_input_data;
/* misc */
gint delay;
/* output */
gboolean do_plc;
gboolean do_estimate_rate;
GstCaps *allocation_caps;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
GstAllocator *allocator;
GstAllocationParams params;
} GstAudioDecoderContext;
struct _GstAudioDecoderPrivate
{
/* activation status */
gboolean active;
/* input base/first ts as basis for output ts */
GstClockTime base_ts;
/* input samples processed and sent downstream so far (w.r.t. base_ts) */
guint64 samples;
/* collected input data */
GstAdapter *adapter;
/* tracking input ts for changes */
GstClockTime prev_ts;
guint64 prev_distance;
/* frames obtained from input */
GQueue frames;
/* collected output data */
GstAdapter *adapter_out;
/* ts and duration for output data collected above */
GstClockTime out_ts, out_dur;
/* mark outgoing discont */
gboolean discont;
/* subclass gave all it could already */
gboolean drained;
/* subclass currently being forcibly drained */
gboolean force;
/* input_segment are output_segment identical */
gboolean in_out_segment_sync;
/* TRUE if we have an active set of instant rate flags */
gboolean decode_flags_override;
GstSegmentFlags decode_flags;
/* expecting the buffer with DISCONT flag */
gboolean expecting_discont_buf;
/* number of samples pushed out via _finish_subframe(), resets on _finish_frame() */
guint subframe_samples;
/* input bps estimatation */
/* global in bytes seen */
guint64 bytes_in;
/* global samples sent out */
guint64 samples_out;
/* bytes flushed during parsing */
guint sync_flush;
/* error count */
gint error_count;
/* max errors */
gint max_errors;
/* upstream stream tags (global tags are passed through as-is) */
GstTagList *upstream_tags;
/* subclass tags */
GstTagList *taglist; /* FIXME: rename to decoder_tags */
GstTagMergeMode decoder_tags_merge_mode;
gboolean taglist_changed; /* FIXME: rename to tags_changed */
/* whether circumstances allow output aggregation */
gint agg;
/* reverse playback queues */
/* collect input */
GList *gather;
/* to-be-decoded */
GList *decode;
/* reversed output */
GList *queued;
/* context storage */
GstAudioDecoderContext ctx;
/* properties */
GstClockTime latency;
GstClockTime tolerance;
gboolean plc;
gboolean drainable;
gboolean needs_format;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
/* flags */
gboolean use_default_pad_acceptcaps;
};
/* cached quark to avoid contention on the global quark table lock */
#define META_TAG_AUDIO meta_tag_audio_quark
static GQuark meta_tag_audio_quark;
static void gst_audio_decoder_finalize (GObject * object);
static void gst_audio_decoder_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_decoder_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static void gst_audio_decoder_clear_queues (GstAudioDecoder * dec);
static GstFlowReturn gst_audio_decoder_chain_reverse (GstAudioDecoder *
dec, GstBuffer * buf);
static GstStateChangeReturn gst_audio_decoder_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec,
GstEvent * event);
static gboolean gst_audio_decoder_src_eventfunc (GstAudioDecoder * dec,
GstEvent * event);
static gboolean gst_audio_decoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_decoder_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_decoder_sink_setcaps (GstAudioDecoder * dec,
GstCaps * caps);
static GstFlowReturn gst_audio_decoder_chain (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static gboolean gst_audio_decoder_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_audio_decoder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full);
static gboolean gst_audio_decoder_decide_allocation_default (GstAudioDecoder *
dec, GstQuery * query);
static gboolean gst_audio_decoder_propose_allocation_default (GstAudioDecoder *
dec, GstQuery * query);
static gboolean gst_audio_decoder_negotiate_default (GstAudioDecoder * dec);
static gboolean gst_audio_decoder_negotiate_unlocked (GstAudioDecoder * dec);
static gboolean gst_audio_decoder_handle_gap (GstAudioDecoder * dec,
GstEvent * event);
static gboolean gst_audio_decoder_sink_query_default (GstAudioDecoder * dec,
GstQuery * query);
static gboolean gst_audio_decoder_src_query_default (GstAudioDecoder * dec,
GstQuery * query);
static gboolean gst_audio_decoder_transform_meta_default (GstAudioDecoder *
decoder, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);
static GstFlowReturn
gst_audio_decoder_finish_frame_or_subframe (GstAudioDecoder * dec,
GstBuffer * buf, gint frames);
static GstElementClass *parent_class = NULL;
static gint private_offset = 0;
static void gst_audio_decoder_class_init (GstAudioDecoderClass * klass);
static void gst_audio_decoder_init (GstAudioDecoder * dec,
GstAudioDecoderClass * klass);
GType
gst_audio_decoder_get_type (void)
{
static gsize audio_decoder_type = 0;
if (g_once_init_enter (&audio_decoder_type)) {
GType _type;
static const GTypeInfo audio_decoder_info = {
sizeof (GstAudioDecoderClass),
NULL,
NULL,
(GClassInitFunc) gst_audio_decoder_class_init,
NULL,
NULL,
sizeof (GstAudioDecoder),
0,
(GInstanceInitFunc) gst_audio_decoder_init,
};
_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAudioDecoder", &audio_decoder_info, G_TYPE_FLAG_ABSTRACT);
private_offset =
g_type_add_instance_private (_type, sizeof (GstAudioDecoderPrivate));
g_once_init_leave (&audio_decoder_type, _type);
}
return audio_decoder_type;
}
static inline GstAudioDecoderPrivate *
gst_audio_decoder_get_instance_private (GstAudioDecoder * self)
{
return (G_STRUCT_MEMBER_P (self, private_offset));
}
static void
gst_audio_decoder_class_init (GstAudioDecoderClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
GstAudioDecoderClass *audiodecoder_class;
gobject_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
audiodecoder_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
if (private_offset != 0)
g_type_class_adjust_private_offset (klass, &private_offset);
GST_DEBUG_CATEGORY_INIT (audiodecoder_debug, "audiodecoder", 0,
"audio decoder base class");
gobject_class->set_property = gst_audio_decoder_set_property;
gobject_class->get_property = gst_audio_decoder_get_property;
gobject_class->finalize = gst_audio_decoder_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_audio_decoder_change_state);
/* Properties */
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_int64 ("min-latency", "Minimum Latency",
"Aggregate output data to a minimum of latency time (ns)",
0, G_MAXINT64, DEFAULT_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TOLERANCE,
g_param_spec_int64 ("tolerance", "Tolerance",
"Perfect ts while timestamp jitter/imperfection within tolerance (ns)",
0, G_MAXINT64, DEFAULT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PLC,
g_param_spec_boolean ("plc", "Packet Loss Concealment",
"Perform packet loss concealment (if supported)",
DEFAULT_PLC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioDecoder:max-errors:
*
* Maximum number of tolerated consecutive decode errors. See
* gst_audio_decoder_set_max_errors() for more details.
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class, PROP_MAX_ERRORS,
g_param_spec_int ("max-errors", "Max errors",
"Max consecutive decoder errors before returning flow error",
-1, G_MAXINT, DEFAULT_MAX_ERRORS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
audiodecoder_class->sink_event =
GST_DEBUG_FUNCPTR (gst_audio_decoder_sink_eventfunc);
audiodecoder_class->src_event =
GST_DEBUG_FUNCPTR (gst_audio_decoder_src_eventfunc);
audiodecoder_class->propose_allocation =
GST_DEBUG_FUNCPTR (gst_audio_decoder_propose_allocation_default);
audiodecoder_class->decide_allocation =
GST_DEBUG_FUNCPTR (gst_audio_decoder_decide_allocation_default);
audiodecoder_class->negotiate =
GST_DEBUG_FUNCPTR (gst_audio_decoder_negotiate_default);
audiodecoder_class->sink_query =
GST_DEBUG_FUNCPTR (gst_audio_decoder_sink_query_default);
audiodecoder_class->src_query =
GST_DEBUG_FUNCPTR (gst_audio_decoder_src_query_default);
audiodecoder_class->transform_meta =
GST_DEBUG_FUNCPTR (gst_audio_decoder_transform_meta_default);
meta_tag_audio_quark = g_quark_from_static_string (GST_META_TAG_AUDIO_STR);
}
static void
gst_audio_decoder_init (GstAudioDecoder * dec, GstAudioDecoderClass * klass)
{
GstPadTemplate *pad_template;
GST_DEBUG_OBJECT (dec, "gst_audio_decoder_init");
dec->priv = gst_audio_decoder_get_instance_private (dec);
/* Setup sink pad */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
g_return_if_fail (pad_template != NULL);
dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_sink_event));
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_chain));
gst_pad_set_query_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_sink_query));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
GST_DEBUG_OBJECT (dec, "sinkpad created");
/* Setup source pad */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
g_return_if_fail (pad_template != NULL);
dec->srcpad = gst_pad_new_from_template (pad_template, "src");
gst_pad_set_event_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_src_event));
gst_pad_set_query_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_src_query));
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
GST_DEBUG_OBJECT (dec, "srcpad created");
dec->priv->adapter = gst_adapter_new ();
dec->priv->adapter_out = gst_adapter_new ();
g_queue_init (&dec->priv->frames);
g_rec_mutex_init (&dec->stream_lock);
/* property default */
dec->priv->latency = DEFAULT_LATENCY;
dec->priv->tolerance = DEFAULT_TOLERANCE;
dec->priv->plc = DEFAULT_PLC;
dec->priv->drainable = DEFAULT_DRAINABLE;
dec->priv->needs_format = DEFAULT_NEEDS_FORMAT;
dec->priv->max_errors = GST_AUDIO_DECODER_MAX_ERRORS;
/* init state */
dec->priv->ctx.min_latency = 0;
dec->priv->ctx.max_latency = 0;
gst_audio_decoder_reset (dec, TRUE);
GST_DEBUG_OBJECT (dec, "init ok");
}
static void
gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
{
GST_DEBUG_OBJECT (dec, "gst_audio_decoder_reset");
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (full) {
dec->priv->active = FALSE;
GST_OBJECT_LOCK (dec);
dec->priv->bytes_in = 0;
dec->priv->samples_out = 0;
GST_OBJECT_UNLOCK (dec);
dec->priv->agg = -1;
dec->priv->error_count = 0;
gst_audio_decoder_clear_queues (dec);
if (dec->priv->taglist) {
gst_tag_list_unref (dec->priv->taglist);
dec->priv->taglist = NULL;
}
dec->priv->decoder_tags_merge_mode = GST_TAG_MERGE_KEEP_ALL;
if (dec->priv->upstream_tags) {
gst_tag_list_unref (dec->priv->upstream_tags);
dec->priv->upstream_tags = NULL;
}
dec->priv->taglist_changed = FALSE;
gst_segment_init (&dec->input_segment, GST_FORMAT_TIME);
gst_segment_init (&dec->output_segment, GST_FORMAT_TIME);
dec->priv->in_out_segment_sync = TRUE;
g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (dec->priv->pending_events);
dec->priv->pending_events = NULL;
if (dec->priv->ctx.allocator)
gst_object_unref (dec->priv->ctx.allocator);
GST_OBJECT_LOCK (dec);
dec->priv->decode_flags_override = FALSE;
gst_caps_replace (&dec->priv->ctx.input_caps, NULL);
gst_caps_replace (&dec->priv->ctx.caps, NULL);
gst_caps_replace (&dec->priv->ctx.allocation_caps, NULL);
memset (&dec->priv->ctx, 0, sizeof (dec->priv->ctx));
gst_audio_info_init (&dec->priv->ctx.info);
GST_OBJECT_UNLOCK (dec);
dec->priv->ctx.had_output_data = FALSE;
dec->priv->ctx.had_input_data = FALSE;
}
g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (&dec->priv->frames);
gst_adapter_clear (dec->priv->adapter);
gst_adapter_clear (dec->priv->adapter_out);
dec->priv->out_ts = GST_CLOCK_TIME_NONE;
dec->priv->out_dur = 0;
dec->priv->prev_ts = GST_CLOCK_TIME_NONE;
dec->priv->prev_distance = 0;
dec->priv->drained = TRUE;
dec->priv->base_ts = GST_CLOCK_TIME_NONE;
dec->priv->samples = 0;
dec->priv->discont = TRUE;
dec->priv->sync_flush = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
}
static void
gst_audio_decoder_finalize (GObject * object)
{
GstAudioDecoder *dec;
g_return_if_fail (GST_IS_AUDIO_DECODER (object));
dec = GST_AUDIO_DECODER (object);
if (dec->priv->adapter) {
g_object_unref (dec->priv->adapter);
}
if (dec->priv->adapter_out) {
g_object_unref (dec->priv->adapter_out);
}
g_rec_mutex_clear (&dec->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstEvent *
gst_audio_decoder_create_merged_tags_event (GstAudioDecoder * dec)
{
GstTagList *merged_tags;
GST_LOG_OBJECT (dec, "upstream : %" GST_PTR_FORMAT, dec->priv->upstream_tags);
GST_LOG_OBJECT (dec, "decoder : %" GST_PTR_FORMAT, dec->priv->taglist);
GST_LOG_OBJECT (dec, "mode : %d", dec->priv->decoder_tags_merge_mode);
merged_tags =
gst_tag_list_merge (dec->priv->upstream_tags,
dec->priv->taglist, dec->priv->decoder_tags_merge_mode);
GST_DEBUG_OBJECT (dec, "merged : %" GST_PTR_FORMAT, merged_tags);
if (merged_tags == NULL)
return NULL;
if (gst_tag_list_is_empty (merged_tags)) {
gst_tag_list_unref (merged_tags);
return NULL;
}
return gst_event_new_tag (merged_tags);
}
static gboolean
gst_audio_decoder_push_event (GstAudioDecoder * dec, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_copy_segment (event, &seg);
GST_DEBUG_OBJECT (dec, "starting segment %" GST_SEGMENT_FORMAT, &seg);
dec->output_segment = seg;
dec->priv->in_out_segment_sync =
gst_segment_is_equal (&dec->input_segment, &seg);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
default:
break;
}
return gst_pad_push_event (dec->srcpad, event);
}
static gboolean
gst_audio_decoder_negotiate_default (GstAudioDecoder * dec)
{
GstAudioDecoderClass *klass;
gboolean res = TRUE;
GstCaps *caps;
GstCaps *prevcaps;
GstQuery *query = NULL;
GstAllocator *allocator;
GstAllocationParams params;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (&dec->priv->ctx.info), FALSE);
g_return_val_if_fail (GST_IS_CAPS (dec->priv->ctx.caps), FALSE);
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
caps = dec->priv->ctx.caps;
if (dec->priv->ctx.allocation_caps == NULL)
dec->priv->ctx.allocation_caps = gst_caps_ref (caps);
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
if (dec->priv->pending_events) {
GList **pending_events, *l;
pending_events = &dec->priv->pending_events;
GST_DEBUG_OBJECT (dec, "Pushing pending events");
for (l = *pending_events; l;) {
GstEvent *event = GST_EVENT (l->data);
GList *tmp;
if (GST_EVENT_TYPE (event) < GST_EVENT_CAPS) {
gst_audio_decoder_push_event (dec, l->data);
tmp = l;
l = l->next;
*pending_events = g_list_delete_link (*pending_events, tmp);
} else {
l = l->next;
}
}
}
prevcaps = gst_pad_get_current_caps (dec->srcpad);
if (!prevcaps || !gst_caps_is_equal (prevcaps, caps))
res = gst_pad_set_caps (dec->srcpad, caps);
if (prevcaps)
gst_caps_unref (prevcaps);
if (!res)
goto done;
dec->priv->ctx.output_format_changed = FALSE;
query = gst_query_new_allocation (dec->priv->ctx.allocation_caps, TRUE);
if (!gst_pad_peer_query (dec->srcpad, query)) {
GST_DEBUG_OBJECT (dec, "didn't get downstream ALLOCATION hints");
}
g_assert (klass->decide_allocation != NULL);
res = klass->decide_allocation (dec, query);
GST_DEBUG_OBJECT (dec, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, res,
query);
if (!res)
goto no_decide_allocation;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
gst_query_parse_nth_allocation_param (query, 0, &allocator, &params);
} else {
allocator = NULL;
gst_allocation_params_init (&params);
}
if (dec->priv->ctx.allocator)
gst_object_unref (dec->priv->ctx.allocator);
dec->priv->ctx.allocator = allocator;
dec->priv->ctx.params = params;
done:
if (query)
gst_query_unref (query);
return res;
/* ERRORS */
no_decide_allocation:
{
GST_WARNING_OBJECT (dec, "Subclass failed to decide allocation");
goto done;
}
}
static gboolean
gst_audio_decoder_negotiate_unlocked (GstAudioDecoder * dec)
{
GstAudioDecoderClass *klass = GST_AUDIO_DECODER_GET_CLASS (dec);
gboolean ret = TRUE;
if (G_LIKELY (klass->negotiate))
ret = klass->negotiate (dec);
return ret;
}
/**
* gst_audio_decoder_negotiate:
* @dec: a #GstAudioDecoder
*
* Negotiate with downstream elements to currently configured #GstAudioInfo.
* Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
* negotiate fails.
*
* Returns: %TRUE if the negotiation succeeded, else %FALSE.
*/
gboolean
gst_audio_decoder_negotiate (GstAudioDecoder * dec)
{
GstAudioDecoderClass *klass;
gboolean res = TRUE;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_pad_check_reconfigure (dec->srcpad);
if (klass->negotiate) {
res = klass->negotiate (dec);
if (!res)
gst_pad_mark_reconfigure (dec->srcpad);
}
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return res;
}
/**
* gst_audio_decoder_set_output_format:
* @dec: a #GstAudioDecoder
* @info: #GstAudioInfo
*
* Configure output info on the srcpad of @dec.
*
* Returns: %TRUE on success.
**/
gboolean
gst_audio_decoder_set_output_format (GstAudioDecoder * dec,
const GstAudioInfo * info)
{
gboolean res = TRUE;
GstCaps *caps = NULL;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (info), FALSE);
/* If the audio info can't be converted to caps,
* it was invalid */
caps = gst_audio_info_to_caps (info);
if (!caps) {
GST_WARNING_OBJECT (dec, "invalid output format");
return FALSE;
}
res = gst_audio_decoder_set_output_caps (dec, caps);
gst_caps_unref (caps);
return res;
}
/**
* gst_audio_decoder_set_output_caps:
* @dec: a #GstAudioDecoder
* @caps: (transfer none): (fixed) #GstCaps
*
* Configure output caps on the srcpad of @dec. Similar to
* gst_audio_decoder_set_output_format(), but allows subclasses to specify
* output caps that can't be expressed via #GstAudioInfo e.g. caps that have
* caps features.
*
* Returns: %TRUE on success.
*
* Since: 1.16
**/
gboolean
gst_audio_decoder_set_output_caps (GstAudioDecoder * dec, GstCaps * caps)
{
gboolean res = TRUE;
guint old_rate;
GstCaps *templ_caps;
GstAudioInfo info;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
GST_DEBUG_OBJECT (dec, "Setting srcpad caps %" GST_PTR_FORMAT, caps);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (!gst_caps_is_fixed (caps))
goto refuse_caps;
/* check if caps can be parsed */
if (!gst_audio_info_from_caps (&info, caps))
goto refuse_caps;
/* Only allow caps that are a subset of the template caps */
templ_caps = gst_pad_get_pad_template_caps (dec->srcpad);
if (!gst_caps_is_subset (caps, templ_caps)) {
GST_WARNING_OBJECT (dec, "Requested output format %" GST_PTR_FORMAT
" do not match template %" GST_PTR_FORMAT, caps, templ_caps);
gst_caps_unref (templ_caps);
goto refuse_caps;
}
gst_caps_unref (templ_caps);
/* adjust ts tracking to new sample rate */
old_rate = GST_AUDIO_INFO_RATE (&dec->priv->ctx.info);
if (GST_CLOCK_TIME_IS_VALID (dec->priv->base_ts) && old_rate) {
dec->priv->base_ts +=
GST_FRAMES_TO_CLOCK_TIME (dec->priv->samples, old_rate);
dec->priv->samples = 0;
}
/* copy the GstAudioInfo */
GST_OBJECT_LOCK (dec);
dec->priv->ctx.info = info;
GST_OBJECT_UNLOCK (dec);
gst_caps_replace (&dec->priv->ctx.caps, caps);
dec->priv->ctx.output_format_changed = TRUE;
done:
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (dec, "invalid output format");
res = FALSE;
goto done;
}
}
static gboolean
gst_audio_decoder_sink_setcaps (GstAudioDecoder * dec, GstCaps * caps)
{
GstAudioDecoderClass *klass;
gboolean res = TRUE;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (dec->priv->ctx.input_caps
&& gst_caps_is_equal (dec->priv->ctx.input_caps, caps)) {
GST_DEBUG_OBJECT (dec, "Caps did not change, not setting again");
goto done;
}
/* NOTE pbutils only needed here */
/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
#if 0
if (!dec->priv->taglist)
dec->priv->taglist = gst_tag_list_new ();
dec->priv->taglist = gst_tag_list_make_writable (dec->priv->taglist);
gst_pb_utils_add_codec_description_to_tag_list (dec->priv->taglist,
GST_TAG_AUDIO_CODEC, caps);
dec->priv->taglist_changed = TRUE;
#endif
if (klass->set_format)
res = klass->set_format (dec, caps);
if (res)
gst_caps_replace (&dec->priv->ctx.input_caps, caps);
done:
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return res;
}
static void
gst_audio_decoder_setup (GstAudioDecoder * dec)
{
GstQuery *query;
gboolean res;
/* check if in live pipeline, then latency messing is no-no */
query = gst_query_new_latency ();
res = gst_pad_peer_query (dec->sinkpad, query);
if (res) {
gst_query_parse_latency (query, &res, NULL, NULL);
res = !res;
}
gst_query_unref (query);
/* normalize to bool */
dec->priv->agg = ! !res;
}
static GstFlowReturn
gst_audio_decoder_push_forward (GstAudioDecoder * dec, GstBuffer * buf)
{
GstAudioDecoderClass *klass;
GstAudioDecoderPrivate *priv;
GstAudioDecoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime ts;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
priv = dec->priv;
ctx = &dec->priv->ctx;
g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
if (G_UNLIKELY (!buf)) {
g_assert_not_reached ();
return GST_FLOW_OK;
}
ctx->had_output_data = TRUE;
ts = GST_BUFFER_PTS (buf);
GST_LOG_OBJECT (dec,
"clipping buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* clip buffer */
buf = gst_audio_buffer_clip (buf, &dec->output_segment, ctx->info.rate,
ctx->info.bpf);
if (G_UNLIKELY (!buf)) {
GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
/* only check and return EOS if upstream still
* in the same segment and interested as such */
if (dec->priv->in_out_segment_sync) {
if (dec->output_segment.rate >= 0) {
if (ts >= dec->output_segment.stop)
ret = GST_FLOW_EOS;
} else if (ts < dec->output_segment.start) {
ret = GST_FLOW_EOS;
}
}
goto exit;
}
/* decorate */
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (dec, "marking discont");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* track where we are */
if (G_LIKELY (GST_BUFFER_PTS_IS_VALID (buf))) {
/* duration should always be valid for raw audio */
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
dec->output_segment.position =
GST_BUFFER_PTS (buf) + GST_BUFFER_DURATION (buf);
}
if (klass->pre_push) {
/* last chance for subclass to do some dirty stuff */
ret = klass->pre_push (dec, &buf);
if (ret != GST_FLOW_OK || !buf) {
GST_DEBUG_OBJECT (dec, "subclass returned %s, buf %p",
gst_flow_get_name (ret), buf);
if (buf)
gst_buffer_unref (buf);
goto exit;
}
}
GST_LOG_OBJECT (dec,
"pushing buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
ret = gst_pad_push (dec->srcpad, buf);
exit:
return ret;
}
/* mini aggregator combining output buffers into fewer larger ones,
* if so allowed/configured */
static GstFlowReturn
gst_audio_decoder_output (GstAudioDecoder * dec, GstBuffer * buf)
{
GstAudioDecoderPrivate *priv;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *inbuf = NULL;
priv = dec->priv;
if (G_UNLIKELY (priv->agg < 0))
gst_audio_decoder_setup (dec);
if (G_LIKELY (buf)) {
GST_LOG_OBJECT (dec,
"output buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
again:
inbuf = NULL;
if (priv->agg && dec->priv->latency > 0 &&
priv->ctx.info.layout == GST_AUDIO_LAYOUT_INTERLEAVED) {
gint av;
gboolean assemble = FALSE;
const GstClockTimeDiff tol = 10 * GST_MSECOND;
GstClockTimeDiff diff = -100 * GST_MSECOND;
av = gst_adapter_available (priv->adapter_out);
if (G_UNLIKELY (!buf)) {
/* forcibly send current */
assemble = TRUE;
GST_LOG_OBJECT (dec, "forcing fragment flush");
} else if (av && (!GST_BUFFER_PTS_IS_VALID (buf) ||
!GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
((diff = GST_CLOCK_DIFF (GST_BUFFER_PTS (buf),
priv->out_ts + priv->out_dur)) > tol) || diff < -tol)) {
assemble = TRUE;
GST_LOG_OBJECT (dec, "buffer %d ms apart from current fragment",
(gint) (diff / GST_MSECOND));
} else {
/* add or start collecting */
if (!av) {
GST_LOG_OBJECT (dec, "starting new fragment");
priv->out_ts = GST_BUFFER_PTS (buf);
} else {
GST_LOG_OBJECT (dec, "adding to fragment");
}
gst_adapter_push (priv->adapter_out, buf);
priv->out_dur += GST_BUFFER_DURATION (buf);
av += gst_buffer_get_size (buf);
buf = NULL;
}
if (priv->out_dur > dec->priv->latency)
assemble = TRUE;
if (av && assemble) {
GST_LOG_OBJECT (dec, "assembling fragment");
inbuf = buf;
buf = gst_adapter_take_buffer (priv->adapter_out, av);
GST_BUFFER_PTS (buf) = priv->out_ts;
GST_BUFFER_DURATION (buf) = priv->out_dur;
priv->out_ts = GST_CLOCK_TIME_NONE;
priv->out_dur = 0;
}
}
if (G_LIKELY (buf)) {
if (dec->output_segment.rate > 0.0) {
ret = gst_audio_decoder_push_forward (dec, buf);
GST_LOG_OBJECT (dec, "buffer pushed: %s", gst_flow_get_name (ret));
} else {
ret = GST_FLOW_OK;
priv->queued = g_list_prepend (priv->queued, buf);
GST_LOG_OBJECT (dec, "buffer queued");
}
if (inbuf) {
buf = inbuf;
goto again;
}
}
return ret;
}
static void
send_pending_events (GstAudioDecoder * dec)
{
GstAudioDecoderPrivate *priv = dec->priv;
GList *pending_events, *l;
pending_events = priv->pending_events;
priv->pending_events = NULL;
GST_DEBUG_OBJECT (dec, "Pushing pending events");
for (l = pending_events; l; l = l->next)
gst_audio_decoder_push_event (dec, l->data);
g_list_free (pending_events);
}
/* Iterate the list of pending events, and ensure
* the current output segment is up to date for
* decoding */
static void
apply_pending_events (GstAudioDecoder * dec)
{
GstAudioDecoderPrivate *priv = dec->priv;
GList *l;
GST_DEBUG_OBJECT (dec, "Applying pending segments");
for (l = priv->pending_events; l; l = l->next) {
GstEvent *event = GST_EVENT (l->data);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_copy_segment (event, &seg);
GST_DEBUG_OBJECT (dec, "starting segment %" GST_SEGMENT_FORMAT, &seg);
dec->output_segment = seg;
dec->priv->in_out_segment_sync =
gst_segment_is_equal (&dec->input_segment, &seg);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
default:
break;
}
}
}
static GstFlowReturn
check_pending_reconfigure (GstAudioDecoder * dec)
{
GstFlowReturn ret = GST_FLOW_OK;
GstAudioDecoderContext *ctx;
gboolean needs_reconfigure;
ctx = &dec->priv->ctx;
needs_reconfigure = gst_pad_check_reconfigure (dec->srcpad);
if (G_UNLIKELY (ctx->output_format_changed ||
(GST_AUDIO_INFO_IS_VALID (&ctx->info)
&& needs_reconfigure))) {
if (!gst_audio_decoder_negotiate_unlocked (dec)) {
gst_pad_mark_reconfigure (dec->srcpad);
if (GST_PAD_IS_FLUSHING (dec->srcpad))
ret = GST_FLOW_FLUSHING;
else
ret = GST_FLOW_NOT_NEGOTIATED;
}
}
return ret;
}
static gboolean
gst_audio_decoder_transform_meta_default (GstAudioDecoder *
decoder, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf)
{
const GstMetaInfo *info = meta->info;
const gchar *const *tags;
const gchar *const supported_tags[] = {
GST_META_TAG_AUDIO_STR,
GST_META_TAG_AUDIO_CHANNELS_STR,
NULL,
};
tags = gst_meta_api_type_get_tags (info->api);
if (!tags)
return TRUE;
while (*tags) {
if (!g_strv_contains (supported_tags, *tags))
return FALSE;
tags++;
}
return TRUE;
}
typedef struct
{
GstAudioDecoder *decoder;
GstBuffer *outbuf;
} CopyMetaData;
static gboolean
foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
{
CopyMetaData *data = user_data;
GstAudioDecoder *decoder = data->decoder;
GstAudioDecoderClass *klass = GST_AUDIO_DECODER_GET_CLASS (decoder);
GstBuffer *outbuf = data->outbuf;
const GstMetaInfo *info = (*meta)->info;
gboolean do_copy = FALSE;
if (gst_meta_api_type_has_tag (info->api, _gst_meta_tag_memory)) {
/* never call the transform_meta with memory specific metadata */
GST_DEBUG_OBJECT (decoder, "not copying memory specific metadata %s",
g_type_name (info->api));
do_copy = FALSE;
} else if (klass->transform_meta) {
do_copy = klass->transform_meta (decoder, outbuf, *meta, inbuf);
GST_DEBUG_OBJECT (decoder, "transformed metadata %s: copy: %d",
g_type_name (info->api), do_copy);
}
/* we only copy metadata when the subclass implemented a transform_meta
* function and when it returns %TRUE */
if (do_copy && info->transform_func) {
GstMetaTransformCopy copy_data = { FALSE, 0, -1 };
GST_DEBUG_OBJECT (decoder, "copy metadata %s", g_type_name (info->api));
/* simply copy then */
info->transform_func (outbuf, *meta, inbuf,
_gst_meta_transform_copy, &copy_data);
}
return TRUE;
}
/**
* gst_audio_decoder_finish_subframe:
* @dec: a #GstAudioDecoder
* @buf: (transfer full) (allow-none): decoded data
*
* Collects decoded data and pushes it downstream. This function may be called
* multiple times for a given input frame.
*
* @buf may be NULL in which case it is assumed that the current input frame is
* finished. This is equivalent to calling gst_audio_decoder_finish_subframe()
* with a NULL buffer and frames=1 after having pushed out all decoded audio
* subframes using this function.
*
* When called with valid data in @buf the source pad caps must have been set
* already.
*
* Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
* invalidated by a call to this function.
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*
* Since: 1.16
*/
GstFlowReturn
gst_audio_decoder_finish_subframe (GstAudioDecoder * dec, GstBuffer * buf)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), GST_FLOW_ERROR);
if (buf == NULL)
return gst_audio_decoder_finish_frame_or_subframe (dec, NULL, 1);
else
return gst_audio_decoder_finish_frame_or_subframe (dec, buf, 0);
}
/**
* gst_audio_decoder_finish_frame:
* @dec: a #GstAudioDecoder
* @buf: (transfer full) (allow-none): decoded data
* @frames: number of decoded frames represented by decoded data
*
* Collects decoded data and pushes it downstream.
*
* @buf may be NULL in which case the indicated number of frames
* are discarded and considered to have produced no output
* (e.g. lead-in or setup frames).
* Otherwise, source pad caps must be set when it is called with valid
* data in @buf.
*
* Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
* invalidated by a call to this function.
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*/
GstFlowReturn
gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
gint frames)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), GST_FLOW_ERROR);
/* no dummy calls please */
g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
return gst_audio_decoder_finish_frame_or_subframe (dec, buf, frames);
}
/* frames == 0 indicates that this is a sub-frame and further sub-frames may
* follow for the current input frame. */
static GstFlowReturn
gst_audio_decoder_finish_frame_or_subframe (GstAudioDecoder * dec,
GstBuffer * buf, gint frames)
{
GstAudioDecoderPrivate *priv;
GstAudioDecoderContext *ctx;
GstAudioDecoderClass *klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GstAudioMeta *meta;
GstClockTime ts, next_ts;
gsize size, samples = 0;
GstFlowReturn ret = GST_FLOW_OK;
GQueue inbufs = G_QUEUE_INIT;
gboolean is_subframe = (frames == 0);
gboolean do_check_resync;
/* subclass should not hand us no data */
g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
GST_FLOW_ERROR);
/* if it's a subframe (frames == 0) we must have a valid buffer */
g_assert (!is_subframe || buf != NULL);
priv = dec->priv;
ctx = &dec->priv->ctx;
meta = buf ? gst_buffer_get_audio_meta (buf) : NULL;
size = buf ? gst_buffer_get_size (buf) : 0;
samples = buf ? (meta ? meta->samples : size / ctx->info.bpf) : 0;
/* must know the output format by now */
g_return_val_if_fail (buf == NULL || GST_AUDIO_INFO_IS_VALID (&ctx->info),
GST_FLOW_ERROR);
GST_LOG_OBJECT (dec,
"accepting %" G_GSIZE_FORMAT " bytes == %" G_GSIZE_FORMAT
" samples for %d frames", buf ? size : 0, samples, frames);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (buf != NULL && priv->subframe_samples == 0) {
ret = check_pending_reconfigure (dec);
if (ret == GST_FLOW_FLUSHING || ret == GST_FLOW_NOT_NEGOTIATED) {
gst_buffer_unref (buf);
goto exit;
}
if (priv->pending_events)
send_pending_events (dec);
}
/* sanity checking */
if (G_LIKELY (buf && ctx->info.bpf)) {
if (!meta || meta->info.layout == GST_AUDIO_LAYOUT_INTERLEAVED) {
/* output should be whole number of sample frames */
if (size % ctx->info.bpf)
goto wrong_buffer;
/* output should have no additional padding */
if (samples != size / ctx->info.bpf)
goto wrong_samples;
} else {
/* can't have more samples than what the buffer fits */
if (samples > size / ctx->info.bpf)
goto wrong_samples;
}
}
/* frame and ts book-keeping */
if (G_UNLIKELY (frames < 0)) {
if (G_UNLIKELY (-frames - 1 > priv->frames.length)) {
GST_ELEMENT_WARNING (dec, STREAM, DECODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
frames = 0;
} else {
frames = priv->frames.length + frames + 1;
}
} else if (G_UNLIKELY (frames > priv->frames.length)) {
if (G_LIKELY (!priv->force)) {
GST_ELEMENT_WARNING (dec, STREAM, DECODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
}
frames = priv->frames.length;
}
if (G_LIKELY (priv->frames.length))
ts = GST_BUFFER_PTS (priv->frames.head->data);
else
ts = GST_CLOCK_TIME_NONE;
GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (ts));
if (is_subframe && priv->frames.length == 0)
goto subframe_without_pending_input_frame;
/* this will be skipped in the is_subframe case because frames will be 0 */
while (priv->frames.length && frames) {
g_queue_push_tail (&inbufs, g_queue_pop_head (&priv->frames));
dec->priv->ctx.delay = dec->priv->frames.length;
frames--;
}
if (G_UNLIKELY (!buf))
goto exit;
/* lock on */
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
priv->base_ts = ts;
GST_DEBUG_OBJECT (dec, "base_ts now %" GST_TIME_FORMAT, GST_TIME_ARGS (ts));
}
/* still no valid ts, track the segment one */
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) &&
dec->output_segment.rate > 0.0) {
priv->base_ts = dec->output_segment.start;
}
/* only check for resync at the beginning of an input/output frame */
do_check_resync = !is_subframe || priv->subframe_samples == 0;
/* slightly convoluted approach caters for perfect ts if subclass desires. */
if (do_check_resync && GST_CLOCK_TIME_IS_VALID (ts)) {
if (dec->priv->tolerance > 0) {
GstClockTimeDiff diff;
g_assert (GST_CLOCK_TIME_IS_VALID (priv->base_ts));
next_ts = priv->base_ts +
gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
GST_LOG_OBJECT (dec,
"buffer is %" G_GUINT64_FORMAT " samples past base_ts %"
GST_TIME_FORMAT ", expected ts %" GST_TIME_FORMAT, priv->samples,
GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
diff = GST_CLOCK_DIFF (next_ts, ts);
GST_LOG_OBJECT (dec, "ts diff %d ms", (gint) (diff / GST_MSECOND));
/* if within tolerance,
* discard buffer ts and carry on producing perfect stream,
* otherwise resync to ts */
if (G_UNLIKELY (diff < (gint64) - dec->priv->tolerance ||
diff > (gint64) dec->priv->tolerance)) {
GST_DEBUG_OBJECT (dec, "base_ts resync");
priv->base_ts = ts;
priv->samples = 0;
}
} else {
GST_DEBUG_OBJECT (dec, "base_ts resync");
priv->base_ts = ts;
priv->samples = 0;
}
}
/* delayed one-shot stuff until confirmed data */
if (priv->taglist && priv->taglist_changed) {
GstEvent *tags_event;
tags_event = gst_audio_decoder_create_merged_tags_event (dec);
if (tags_event != NULL)
gst_audio_decoder_push_event (dec, tags_event);
priv->taglist_changed = FALSE;
}
buf = gst_buffer_make_writable (buf);
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
GST_BUFFER_PTS (buf) =
priv->base_ts +
GST_FRAMES_TO_CLOCK_TIME (priv->samples, ctx->info.rate);
GST_BUFFER_DURATION (buf) = priv->base_ts +
GST_FRAMES_TO_CLOCK_TIME (priv->samples + samples, ctx->info.rate) -
GST_BUFFER_PTS (buf);
} else {
GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buf) =
GST_FRAMES_TO_CLOCK_TIME (samples, ctx->info.rate);
}
if (klass->transform_meta) {
if (inbufs.length) {
GList *l;
for (l = inbufs.head; l; l = l->next) {
CopyMetaData data;
data.decoder = dec;
data.outbuf = buf;
gst_buffer_foreach_meta (l->data, foreach_metadata, &data);
}
} else if (is_subframe) {
CopyMetaData data;
GstBuffer *in_buf;
/* For subframes we assume a 1:N relationship for now, so we just take
* metas from the first pending input buf */
in_buf = g_queue_peek_head (&priv->frames);
data.decoder = dec;
data.outbuf = buf;
gst_buffer_foreach_meta (in_buf, foreach_metadata, &data);
} else {
GST_WARNING_OBJECT (dec,
"Can't copy metadata because input buffers disappeared");
}
}
GST_OBJECT_LOCK (dec);
priv->samples += samples;
priv->samples_out += samples;
GST_OBJECT_UNLOCK (dec);
/* we got data, so note things are looking up */
if (G_UNLIKELY (dec->priv->error_count))
dec->priv->error_count = 0;
ret = gst_audio_decoder_output (dec, buf);
exit:
g_queue_foreach (&inbufs, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (&inbufs);
if (is_subframe)
dec->priv->subframe_samples += samples;
else
dec->priv->subframe_samples = 0;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return ret;
/* ERRORS */
wrong_buffer:
{
/* arguably more of a programming error? */
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("buffer size %" G_GSIZE_FORMAT " not a multiple of %d", size,
ctx->info.bpf));
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
wrong_samples:
{
/* arguably more of a programming error? */
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("GstAudioMeta samples (%" G_GSIZE_FORMAT ") are inconsistent with "
"the buffer size and layout (size/bpf = %" G_GSIZE_FORMAT ")",
meta->samples, size / ctx->info.bpf));
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
subframe_without_pending_input_frame:
{
/* arguably more of a programming error? */
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Received decoded subframe, but no pending frame"));
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
}
static GstFlowReturn
gst_audio_decoder_handle_frame (GstAudioDecoder * dec,
GstAudioDecoderClass * klass, GstBuffer * buffer)
{
/* Skip decoding and send a GAP instead if
* GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO is set and we have timestamps
* FIXME: We only do this for forward playback atm, because reverse
* playback would require accumulating GAP events and pushing them
* out in reverse order as for normal audio samples
*/
if (G_UNLIKELY (dec->input_segment.rate > 0.0
&& dec->input_segment.flags & GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO)) {
if (buffer) {
GstClockTime ts = GST_BUFFER_PTS (buffer);
if (GST_CLOCK_TIME_IS_VALID (ts)) {
GstEvent *event = gst_event_new_gap (ts, GST_BUFFER_DURATION (buffer));
gst_buffer_unref (buffer);
GST_LOG_OBJECT (dec, "Skipping decode in trickmode and sending gap");
gst_audio_decoder_handle_gap (dec, event);
return GST_FLOW_OK;
}
}
}
if (G_LIKELY (buffer)) {
gsize size = gst_buffer_get_size (buffer);
/* keep around for admin */
GST_LOG_OBJECT (dec,
"tracking frame size %" G_GSIZE_FORMAT ", ts %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
g_queue_push_tail (&dec->priv->frames, buffer);
dec->priv->ctx.delay = dec->priv->frames.length;
GST_OBJECT_LOCK (dec);
dec->priv->bytes_in += size;
GST_OBJECT_UNLOCK (dec);
} else {
GST_LOG_OBJECT (dec, "providing subclass with NULL frame");
}
return klass->handle_frame (dec, buffer);
}
/* maybe subclass configurable instead, but this allows for a whole lot of
* raw samples, so at least quite some encoded ... */
#define GST_AUDIO_DECODER_MAX_SYNC 10 * 8 * 2 * 1024
static GstFlowReturn
gst_audio_decoder_push_buffers (GstAudioDecoder * dec, gboolean force)
{
GstAudioDecoderClass *klass;
GstAudioDecoderPrivate *priv;
GstAudioDecoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *buffer;
gint av, flush;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
priv = dec->priv;
ctx = &dec->priv->ctx;
g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
av = gst_adapter_available (priv->adapter);
GST_DEBUG_OBJECT (dec, "available: %d", av);
while (ret == GST_FLOW_OK) {
flush = 0;
ctx->eos = force;
if (G_LIKELY (av)) {
gint len;
GstClockTime ts;
guint64 distance;
/* parse if needed */
if (klass->parse) {
gint offset = 0;
/* limited (legacy) parsing; avoid whole of baseparse */
GST_DEBUG_OBJECT (dec, "parsing available: %d", av);
/* piggyback sync state on discont */
ctx->sync = !priv->discont;
ret = klass->parse (dec, priv->adapter, &offset, &len);
g_assert (offset <= av);
if (offset) {
/* jumped a bit */
GST_DEBUG_OBJECT (dec, "skipped %d; setting DISCONT", offset);
gst_adapter_flush (priv->adapter, offset);
flush = offset;
/* avoid parsing indefinitely */
priv->sync_flush += offset;
if (priv->sync_flush > GST_AUDIO_DECODER_MAX_SYNC)
goto parse_failed;
}
if (ret == GST_FLOW_EOS) {
GST_LOG_OBJECT (dec, "no frame yet");
ret = GST_FLOW_OK;
break;
} else if (ret == GST_FLOW_OK) {
GST_LOG_OBJECT (dec, "frame at offset %d of length %d", offset, len);
g_assert (len);
g_assert (offset + len <= av);
priv->sync_flush = 0;
} else {
break;
}
} else {
len = av;
}
/* track upstream ts, but do not get stuck if nothing new upstream */
ts = gst_adapter_prev_pts (priv->adapter, &distance);
if (ts != priv->prev_ts || distance <= priv->prev_distance) {
priv->prev_ts = ts;
priv->prev_distance = distance;
} else {
GST_LOG_OBJECT (dec, "ts == prev_ts; discarding");
ts = GST_CLOCK_TIME_NONE;
}
buffer = gst_adapter_take_buffer (priv->adapter, len);
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_PTS (buffer) = ts;
flush += len;
priv->force = FALSE;
} else {
if (!force)
break;
if (!priv->drainable) {
priv->drained = TRUE;
break;
}
buffer = NULL;
priv->force = TRUE;
}
ret = gst_audio_decoder_handle_frame (dec, klass, buffer);
/* do not keep pushing it ... */
if (G_UNLIKELY (!av)) {
priv->drained = TRUE;
break;
}
av -= flush;
g_assert (av >= 0);
}
GST_LOG_OBJECT (dec, "done pushing to subclass");
return ret;
/* ERRORS */
parse_failed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("failed to parse stream"));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_audio_decoder_drain (GstAudioDecoder * dec)
{
GstFlowReturn ret;
if (dec->priv->drained && !dec->priv->gather)
return GST_FLOW_OK;
/* Apply any pending events before draining, as that
* may update the pending segment info */
apply_pending_events (dec);
/* dispatch reverse pending buffers */
/* chain eventually calls upon drain as well, but by that time
* gather list should be clear, so ok ... */
if (dec->output_segment.rate < 0.0 && dec->priv->gather)
gst_audio_decoder_chain_reverse (dec, NULL);
/* have subclass give all it can */
ret = gst_audio_decoder_push_buffers (dec, TRUE);
if (ret != GST_FLOW_OK) {
GST_WARNING_OBJECT (dec, "audio decoder push buffers failed");
goto drain_failed;
}
/* ensure all output sent */
ret = gst_audio_decoder_output (dec, NULL);
if (ret != GST_FLOW_OK)
GST_WARNING_OBJECT (dec, "audio decoder output failed");
drain_failed:
/* everything should be away now */
if (dec->priv->frames.length) {
/* not fatal/impossible though if subclass/codec eats stuff */
GST_WARNING_OBJECT (dec, "still %d frames left after draining",
dec->priv->frames.length);
g_queue_foreach (&dec->priv->frames, (GFunc) gst_buffer_unref, NULL);
g_queue_clear (&dec->priv->frames);
}
/* discard (unparsed) leftover */
gst_adapter_clear (dec->priv->adapter);
return ret;
}
/* hard == FLUSH, otherwise discont */
static GstFlowReturn
gst_audio_decoder_flush (GstAudioDecoder * dec, gboolean hard)
{
GstAudioDecoderClass *klass;
GstFlowReturn ret = GST_FLOW_OK;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_LOG_OBJECT (dec, "flush hard %d", hard);
if (!hard) {
ret = gst_audio_decoder_drain (dec);
} else {
gst_audio_decoder_clear_queues (dec);
gst_segment_init (&dec->input_segment, GST_FORMAT_TIME);
gst_segment_init (&dec->output_segment, GST_FORMAT_TIME);
dec->priv->error_count = 0;
}
/* only bother subclass with flushing if known it is already alive
* and kicking out stuff */
if (klass->flush && dec->priv->samples_out > 0)
klass->flush (dec, hard);
/* and get (re)set for the sequel */
gst_audio_decoder_reset (dec, FALSE);
return ret;
}
static GstFlowReturn
gst_audio_decoder_chain_forward (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
/* discard silly case, though maybe ts may be of value ?? */
if (G_UNLIKELY (gst_buffer_get_size (buffer) == 0)) {
GST_DEBUG_OBJECT (dec, "discarding empty buffer");
gst_buffer_unref (buffer);
goto exit;
}
/* grab buffer */
gst_adapter_push (dec->priv->adapter, buffer);
buffer = NULL;
/* new stuff, so we can push subclass again */
dec->priv->drained = FALSE;
/* hand to subclass */
ret = gst_audio_decoder_push_buffers (dec, FALSE);
exit:
GST_LOG_OBJECT (dec, "chain-done");
return ret;
}
static void
gst_audio_decoder_clear_queues (GstAudioDecoder * dec)
{
GstAudioDecoderPrivate *priv = dec->priv;
g_list_foreach (priv->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (priv->queued);
priv->queued = NULL;
g_list_foreach (priv->gather, (GFunc) gst_mini_object_unref, NULL);
g_list_free (priv->gather);
priv->gather = NULL;
g_list_foreach (priv->decode, (GFunc) gst_mini_object_unref, NULL);
g_list_free (priv->decode);
priv->decode = NULL;
}
/*
* Input:
* Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS
* Discont flag: D D D D
*
* - Each Discont marks a discont in the decoding order.
*
* for vorbis, each buffer is a keyframe when we have the previous
* buffer. This means that to decode buffer 7, we need buffer 6, which
* arrives out of order.
*
* we first gather buffers in the gather queue until we get a DISCONT. We
* prepend each incoming buffer so that they are in reversed order.
*
* gather queue: 9 8 7
* decode queue:
* output queue:
*
* When a DISCONT is received (buffer 4), we move the gather queue to the
* decode queue. This is simply done be taking the head of the gather queue
* and prepending it to the decode queue. This yields:
*
* gather queue:
* decode queue: 7 8 9
* output queue:
*
* Then we decode each buffer in the decode queue in order and put the output
* buffer in the output queue. The first buffer (7) will not produce any output
* because it needs the previous buffer (6) which did not arrive yet. This
* yields:
*
* gather queue:
* decode queue: 7 8 9
* output queue: 9 8
*
* Then we remove the consumed buffers from the decode queue. Buffer 7 is not
* completely consumed, we need to keep it around for when we receive buffer
* 6. This yields:
*
* gather queue:
* decode queue: 7
* output queue: 9 8
*
* Then we accumulate more buffers:
*
* gather queue: 6 5 4
* decode queue: 7
* output queue:
*
* prepending to the decode queue on DISCONT yields:
*
* gather queue:
* decode queue: 4 5 6 7
* output queue:
*
* after decoding and keeping buffer 4:
*
* gather queue:
* decode queue: 4
* output queue: 7 6 5
*
* Etc..
*/
static GstFlowReturn
gst_audio_decoder_flush_decode (GstAudioDecoder * dec)
{
GstAudioDecoderPrivate *priv = dec->priv;
GstFlowReturn res = GST_FLOW_OK;
GstClockTime timestamp;
GList *walk;
walk = priv->decode;
GST_DEBUG_OBJECT (dec, "flushing buffers to decoder");
/* clear buffer and decoder state */
gst_audio_decoder_flush (dec, FALSE);
while (walk) {
GList *next;
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
buf, GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
next = g_list_next (walk);
/* decode buffer, resulting data prepended to output queue */
gst_buffer_ref (buf);
res = gst_audio_decoder_chain_forward (dec, buf);
/* if we generated output, we can discard the buffer, else we
* keep it in the queue */
if (priv->queued) {
GST_DEBUG_OBJECT (dec, "decoded buffer to %p", priv->queued->data);
priv->decode = g_list_delete_link (priv->decode, walk);
gst_buffer_unref (buf);
} else {
GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping");
}
walk = next;
}
/* drain any aggregation (or otherwise) leftover */
gst_audio_decoder_drain (dec);
/* now send queued data downstream */
timestamp = GST_CLOCK_TIME_NONE;
while (priv->queued) {
GstBuffer *buf = GST_BUFFER_CAST (priv->queued->data);
GstClockTime duration;
duration = GST_BUFFER_DURATION (buf);
/* duration should always be valid for raw audio */
g_assert (GST_CLOCK_TIME_IS_VALID (duration));
/* interpolate (backward) if needed */
if (G_LIKELY (timestamp != -1)) {
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
}
if (!GST_BUFFER_PTS_IS_VALID (buf)) {
GST_LOG_OBJECT (dec, "applying reverse interpolated ts %"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
GST_BUFFER_PTS (buf) = timestamp;
} else {
/* track otherwise */
timestamp = GST_BUFFER_PTS (buf);
GST_LOG_OBJECT (dec, "tracking ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
}
if (G_LIKELY (res == GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %" G_GSIZE_FORMAT ", "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* should be already, but let's be sure */
buf = gst_buffer_make_writable (buf);
/* avoid stray DISCONT from forward processing,
* which have no meaning in reverse pushing */
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
res = gst_audio_decoder_push_forward (dec, buf);
} else {
gst_buffer_unref (buf);
}
priv->queued = g_list_delete_link (priv->queued, priv->queued);
}
return res;
}
static GstFlowReturn
gst_audio_decoder_chain_reverse (GstAudioDecoder * dec, GstBuffer * buf)
{
GstAudioDecoderPrivate *priv = dec->priv;
GstFlowReturn result = GST_FLOW_OK;
/* if we have a discont, move buffers to the decode list */
if (!buf || GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
GST_DEBUG_OBJECT (dec, "received discont");
while (priv->gather) {
GstBuffer *gbuf;
gbuf = GST_BUFFER_CAST (priv->gather->data);
/* remove from the gather list */
priv->gather = g_list_delete_link (priv->gather, priv->gather);
/* copy to decode queue */
priv->decode = g_list_prepend (priv->decode, gbuf);
}
/* decode stuff in the decode queue */
gst_audio_decoder_flush_decode (dec);
}
if (G_LIKELY (buf)) {
GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %" G_GSIZE_FORMAT ", "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_PTS (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* add buffer to gather queue */
priv->gather = g_list_prepend (priv->gather, buf);
}
return result;
}
static GstFlowReturn
gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstAudioDecoder *dec;
GstFlowReturn ret;
dec = GST_AUDIO_DECODER (parent);
GST_LOG_OBJECT (dec,
"received buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buffer),
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (G_UNLIKELY (dec->priv->ctx.input_caps == NULL && dec->priv->needs_format))
goto not_negotiated;
dec->priv->ctx.had_input_data = TRUE;
if (!dec->priv->expecting_discont_buf &&
GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gint64 samples, ts;
/* track present position */
ts = dec->priv->base_ts;
samples = dec->priv->samples;
GST_DEBUG_OBJECT (dec, "handling discont");
gst_audio_decoder_flush (dec, FALSE);
dec->priv->discont = TRUE;
/* buffer may claim DISCONT loudly, if it can't tell us where we are now,
* we'll stick to where we were ...
* Particularly useful/needed for upstream BYTE based */
if (dec->input_segment.rate > 0.0 && !GST_BUFFER_PTS_IS_VALID (buffer)) {
GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
dec->priv->base_ts = ts;
dec->priv->samples = samples;
}
}
dec->priv->expecting_discont_buf = FALSE;
if (dec->input_segment.rate > 0.0)
ret = gst_audio_decoder_chain_forward (dec, buffer);
else
ret = gst_audio_decoder_chain_reverse (dec, buffer);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return ret;
/* ERRORS */
not_negotiated:
{
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
}
/* perform upstream byte <-> time conversion (duration, seeking)
* if subclass allows and if enough data for moderately decent conversion */
static inline gboolean
gst_audio_decoder_do_byte (GstAudioDecoder * dec)
{
gboolean ret;
GST_OBJECT_LOCK (dec);
ret = dec->priv->ctx.do_estimate_rate && dec->priv->ctx.info.bpf &&
dec->priv->ctx.info.rate <= dec->priv->samples_out;
GST_OBJECT_UNLOCK (dec);
return ret;
}
/* Must be called holding the GST_AUDIO_DECODER_STREAM_LOCK */
static gboolean
gst_audio_decoder_negotiate_default_caps (GstAudioDecoder * dec)
{
GstCaps *caps, *templcaps;
gint i;
gint channels = 0;
gint rate;
guint64 channel_mask = 0;
gint caps_size;
GstStructure *structure;
GstAudioInfo info;
templcaps = gst_pad_get_pad_template_caps (dec->srcpad);
caps = gst_pad_peer_query_caps (dec->srcpad, templcaps);
if (caps)
gst_caps_unref (templcaps);
else
caps = templcaps;
templcaps = NULL;
if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps))
goto caps_error;
GST_LOG_OBJECT (dec, "peer caps %" GST_PTR_FORMAT, caps);
/* before fixating, try to use whatever upstream provided */
caps = gst_caps_make_writable (caps);
caps_size = gst_caps_get_size (caps);
if (dec->priv->ctx.input_caps) {
GstCaps *sinkcaps = dec->priv->ctx.input_caps;
GstStructure *structure = gst_caps_get_structure (sinkcaps, 0);
if (gst_structure_get_int (structure, "rate", &rate)) {
for (i = 0; i < caps_size; i++) {
gst_structure_set (gst_caps_get_structure (caps, i), "rate",
G_TYPE_INT, rate, NULL);
}
}
if (gst_structure_get_int (structure, "channels", &channels)) {
for (i = 0; i < caps_size; i++) {
gst_structure_set (gst_caps_get_structure (caps, i), "channels",
G_TYPE_INT, channels, NULL);
}
}
if (gst_structure_get (structure, "channel-mask", GST_TYPE_BITMASK,
&channel_mask, NULL)) {
for (i = 0; i < caps_size; i++) {
gst_structure_set (gst_caps_get_structure (caps, i), "channel-mask",
GST_TYPE_BITMASK, channel_mask, NULL);
}
}
}
for (i = 0; i < caps_size; i++) {
structure = gst_caps_get_structure (caps, i);
if (gst_structure_has_field (structure, "channels"))
gst_structure_fixate_field_nearest_int (structure,
"channels", GST_AUDIO_DEF_CHANNELS);
else
gst_structure_set (structure, "channels", G_TYPE_INT,
GST_AUDIO_DEF_CHANNELS, NULL);
if (gst_structure_has_field (structure, "rate"))
gst_structure_fixate_field_nearest_int (structure,
"rate", GST_AUDIO_DEF_RATE);
else
gst_structure_set (structure, "rate", G_TYPE_INT, GST_AUDIO_DEF_RATE,
NULL);
}
caps = gst_caps_fixate (caps);
structure = gst_caps_get_structure (caps, 0);
/* Need to add a channel-mask if channels > 2 */
gst_structure_get_int (structure, "channels", &channels);
if (channels > 2 && !gst_structure_has_field (structure, "channel-mask")) {
channel_mask = gst_audio_channel_get_fallback_mask (channels);
if (channel_mask != 0) {
gst_structure_set (structure, "channel-mask",
GST_TYPE_BITMASK, channel_mask, NULL);
} else {
GST_WARNING_OBJECT (dec, "No default channel-mask for %d channels",
channels);
}
}
if (!caps || !gst_audio_info_from_caps (&info, caps))
goto caps_error;
GST_OBJECT_LOCK (dec);
dec->priv->ctx.info = info;
dec->priv->ctx.caps = caps;
GST_OBJECT_UNLOCK (dec);
GST_INFO_OBJECT (dec,
"Chose default caps %" GST_PTR_FORMAT " for initial gap", caps);
return TRUE;
caps_error:
{
if (caps)
gst_caps_unref (caps);
return FALSE;
}
}
static gboolean
gst_audio_decoder_handle_gap (GstAudioDecoder * dec, GstEvent * event)
{
gboolean ret;
GstClockTime timestamp, duration;
gboolean needs_reconfigure = FALSE;
/* Ensure we have caps first */
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (!GST_AUDIO_INFO_IS_VALID (&dec->priv->ctx.info)) {
if (!gst_audio_decoder_negotiate_default_caps (dec)) {
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Decoder output not negotiated before GAP event."));
gst_event_unref (event);
return FALSE;
}
needs_reconfigure = TRUE;
}
needs_reconfigure = gst_pad_check_reconfigure (dec->srcpad)
|| needs_reconfigure;
if (G_UNLIKELY (dec->priv->ctx.output_format_changed || needs_reconfigure)) {
if (!gst_audio_decoder_negotiate_unlocked (dec)) {
GST_WARNING_OBJECT (dec, "Failed to negotiate with downstream");
gst_pad_mark_reconfigure (dec->srcpad);
}
}
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
gst_event_parse_gap (event, &timestamp, &duration);
/* time progressed without data, see if we can fill the gap with
* some concealment data */
GST_DEBUG_OBJECT (dec,
"gap event: plc %d, do_plc %d, position %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT,
dec->priv->plc, dec->priv->ctx.do_plc,
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration));
if (dec->priv->plc && dec->priv->ctx.do_plc && dec->input_segment.rate > 0.0) {
GstAudioDecoderClass *klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GstBuffer *buf;
/* hand subclass empty frame with duration that needs covering */
buf = gst_buffer_new ();
GST_BUFFER_PTS (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
/* best effort, not much error handling */
gst_audio_decoder_handle_frame (dec, klass, buf);
ret = TRUE;
dec->priv->expecting_discont_buf = TRUE;
gst_event_unref (event);
} else {
GstFlowReturn flowret;
/* sub-class doesn't know how to handle empty buffers,
* so just try sending GAP downstream */
flowret = check_pending_reconfigure (dec);
if (flowret == GST_FLOW_OK) {
send_pending_events (dec);
ret = gst_audio_decoder_push_event (dec, event);
} else {
ret = FALSE;
gst_event_unref (event);
}
}
return ret;
}
static GList *
_flush_events (GstPad * pad, GList * events)
{
GList *tmp;
for (tmp = events; tmp; tmp = tmp->next) {
if (GST_EVENT_TYPE (tmp->data) != GST_EVENT_EOS &&
GST_EVENT_TYPE (tmp->data) != GST_EVENT_SEGMENT &&
GST_EVENT_IS_STICKY (tmp->data)) {
gst_pad_store_sticky_event (pad, GST_EVENT_CAST (tmp->data));
}
gst_event_unref (tmp->data);
}
g_list_free (events);
return NULL;
}
static gboolean
gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
{
gboolean ret;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_STREAM_START:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* finish any data in current segment and clear the decoder
* to be ready for new stream data */
gst_audio_decoder_drain (dec);
gst_audio_decoder_flush (dec, FALSE);
GST_DEBUG_OBJECT (dec, "received STREAM_START. Clearing taglist");
/* Flush upstream tags after a STREAM_START */
if (dec->priv->upstream_tags) {
gst_tag_list_unref (dec->priv->upstream_tags);
dec->priv->upstream_tags = NULL;
dec->priv->taglist_changed = TRUE;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_SEGMENT:
{
GstSegment seg;
GstFormat format;
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_copy_segment (event, &seg);
format = seg.format;
if (format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (dec, "received TIME SEGMENT %" GST_SEGMENT_FORMAT,
&seg);
} else {
gint64 nstart;
GST_DEBUG_OBJECT (dec, "received SEGMENT %" GST_SEGMENT_FORMAT, &seg);
/* handle newsegment resulting from legacy simple seeking */
/* note that we need to convert this whether or not enough data
* to handle initial newsegment */
if (dec->priv->ctx.do_estimate_rate &&
gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, seg.start,
GST_FORMAT_TIME, &nstart)) {
/* best attempt convert */
/* as these are only estimates, stop is kept open-ended to avoid
* premature cutting */
GST_DEBUG_OBJECT (dec, "converted to TIME start %" GST_TIME_FORMAT,
GST_TIME_ARGS (nstart));
seg.format = GST_FORMAT_TIME;
seg.start = nstart;
seg.time = nstart;
seg.stop = GST_CLOCK_TIME_NONE;
/* replace event */
gst_event_unref (event);
event = gst_event_new_segment (&seg);
} else {
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
gst_event_unref (event);
ret = FALSE;
break;
}
}
/* prepare for next segment */
/* Use the segment start as a base timestamp
* in case upstream does not come up with anything better
* (e.g. upstream BYTE) */
if (format != GST_FORMAT_TIME) {
dec->priv->base_ts = seg.start;
dec->priv->samples = 0;
}
/* Update the decode flags in the segment if we have an instant-rate
* override active */
GST_OBJECT_LOCK (dec);
if (dec->priv->decode_flags_override) {
seg.flags &= ~GST_SEGMENT_INSTANT_FLAGS;
seg.flags |= dec->priv->decode_flags & GST_SEGMENT_INSTANT_FLAGS;
}
/* and follow along with segment */
dec->priv->in_out_segment_sync = FALSE;
dec->input_segment = seg;
GST_OBJECT_UNLOCK (dec);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
ret = TRUE;
break;
}
case GST_EVENT_INSTANT_RATE_CHANGE:
{
GstSegmentFlags flags;
GstSegment *seg;
gst_event_parse_instant_rate_change (event, NULL, &flags);
GST_OBJECT_LOCK (dec);
dec->priv->decode_flags_override = TRUE;
dec->priv->decode_flags = flags;
/* Update the input segment flags */
seg = &dec->input_segment;
seg->flags &= ~GST_SEGMENT_INSTANT_FLAGS;
seg->flags |= dec->priv->decode_flags & GST_SEGMENT_INSTANT_FLAGS;
GST_OBJECT_UNLOCK (dec);
/* Forward downstream */
ret = gst_pad_event_default (dec->sinkpad, GST_OBJECT_CAST (dec), event);
break;
}
case GST_EVENT_GAP:
ret = gst_audio_decoder_handle_gap (dec, event);
break;
case GST_EVENT_FLUSH_STOP:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* prepare for fresh start */
gst_audio_decoder_flush (dec, TRUE);
dec->priv->pending_events = _flush_events (dec->srcpad,
dec->priv->pending_events);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
/* Forward FLUSH_STOP, it is expected to be forwarded immediately
* and no buffers are queued anyway. */
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_SEGMENT_DONE:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_audio_decoder_drain (dec);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
/* Forward SEGMENT_DONE because no buffer or serialized event might come after
* SEGMENT_DONE and nothing could trigger another _finish_frame() call. */
if (dec->priv->pending_events)
send_pending_events (dec);
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_EOS:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_audio_decoder_drain (dec);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
if (dec->priv->ctx.had_input_data && !dec->priv->ctx.had_output_data) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
("No valid frames decoded before end of stream"),
("no valid frames found"));
}
/* Forward EOS because no buffer or serialized event will come after
* EOS and nothing could trigger another _finish_frame() call. */
if (dec->priv->pending_events)
send_pending_events (dec);
ret = gst_audio_decoder_push_event (dec, event);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_audio_decoder_sink_setcaps (dec, caps);
gst_event_unref (event);
break;
}
case GST_EVENT_TAG:
{
GstTagList *tags;
gst_event_parse_tag (event, &tags);
if (gst_tag_list_get_scope (tags) == GST_TAG_SCOPE_STREAM) {
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (dec->priv->upstream_tags != tags) {
if (dec->priv->upstream_tags)
gst_tag_list_unref (dec->priv->upstream_tags);
dec->priv->upstream_tags = gst_tag_list_ref (tags);
GST_INFO_OBJECT (dec, "upstream stream tags: %" GST_PTR_FORMAT, tags);
}
gst_event_unref (event);
event = gst_audio_decoder_create_merged_tags_event (dec);
dec->priv->taglist_changed = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
/* No tags, go out of here instead of fall through */
if (!event) {
ret = TRUE;
break;
}
}
/* fall through */
}
default:
if (!GST_EVENT_IS_SERIALIZED (event)) {
ret =
gst_pad_event_default (dec->sinkpad, GST_OBJECT_CAST (dec), event);
} else {
GST_DEBUG_OBJECT (dec, "Enqueuing event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
GST_AUDIO_DECODER_STREAM_LOCK (dec);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
ret = TRUE;
}
break;
}
return ret;
}
static gboolean
gst_audio_decoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstAudioDecoder *dec;
GstAudioDecoderClass *klass;
gboolean ret;
dec = GST_AUDIO_DECODER (parent);
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
if (klass->sink_event)
ret = klass->sink_event (dec, event);
else {
gst_event_unref (event);
ret = FALSE;
}
return ret;
}
static gboolean
gst_audio_decoder_do_seek (GstAudioDecoder * dec, GstEvent * event)
{
GstSeekFlags flags;
GstSeekType start_type, end_type;
GstFormat format;
gdouble rate;
gint64 start, start_time, end_time;
GstSegment seek_segment;
guint32 seqnum;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type,
&start_time, &end_type, &end_time);
/* we'll handle plain open-ended flushing seeks with the simple approach */
if (rate != 1.0) {
GST_DEBUG_OBJECT (dec, "unsupported seek: rate");
return FALSE;
}
if (start_type != GST_SEEK_TYPE_SET) {
GST_DEBUG_OBJECT (dec, "unsupported seek: start time");
return FALSE;
}
if ((end_type != GST_SEEK_TYPE_SET && end_type != GST_SEEK_TYPE_NONE) ||
(end_type == GST_SEEK_TYPE_SET && end_time != GST_CLOCK_TIME_NONE)) {
GST_DEBUG_OBJECT (dec, "unsupported seek: end time");
return FALSE;
}
if (!(flags & GST_SEEK_FLAG_FLUSH)) {
GST_DEBUG_OBJECT (dec, "unsupported seek: not flushing");
return FALSE;
}
memcpy (&seek_segment, &dec->output_segment, sizeof (seek_segment));
gst_segment_do_seek (&seek_segment, rate, format, flags, start_type,
start_time, end_type, end_time, NULL);
start_time = seek_segment.position;
if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
GST_FORMAT_BYTES, &start)) {
GST_DEBUG_OBJECT (dec, "conversion failed");
return FALSE;
}
seqnum = gst_event_get_seqnum (event);
event = gst_event_new_seek (1.0, GST_FORMAT_BYTES, flags,
GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE, -1);
gst_event_set_seqnum (event, seqnum);
GST_DEBUG_OBJECT (dec, "seeking to %" GST_TIME_FORMAT " at byte offset %"
G_GINT64_FORMAT, GST_TIME_ARGS (start_time), start);
return gst_pad_push_event (dec->sinkpad, event);
}
static gboolean
gst_audio_decoder_src_eventfunc (GstAudioDecoder * dec, GstEvent * event)
{
gboolean res;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
GstFormat format;
gdouble rate;
GstSeekFlags flags;
GstSeekType start_type, stop_type;
gint64 start, stop;
gint64 tstart, tstop;
guint32 seqnum;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
seqnum = gst_event_get_seqnum (event);
/* upstream gets a chance first */
if ((res = gst_pad_push_event (dec->sinkpad, event)))
break;
/* if upstream fails for a time seek, maybe we can help if allowed */
if (format == GST_FORMAT_TIME) {
if (gst_audio_decoder_do_byte (dec))
res = gst_audio_decoder_do_seek (dec, event);
break;
}
/* ... though a non-time seek can be aided as well */
/* First bring the requested format to time */
if (!(res =
gst_pad_query_convert (dec->srcpad, format, start,
GST_FORMAT_TIME, &tstart)))
goto convert_error;
if (!(res =
gst_pad_query_convert (dec->srcpad, format, stop, GST_FORMAT_TIME,
&tstop)))
goto convert_error;
/* then seek with time on the peer */
event = gst_event_new_seek (rate, GST_FORMAT_TIME,
flags, start_type, tstart, stop_type, tstop);
gst_event_set_seqnum (event, seqnum);
res = gst_pad_push_event (dec->sinkpad, event);
break;
}
default:
res = gst_pad_event_default (dec->srcpad, GST_OBJECT_CAST (dec), event);
break;
}
done:
return res;
/* ERRORS */
convert_error:
{
GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
goto done;
}
}
static gboolean
gst_audio_decoder_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAudioDecoder *dec;
GstAudioDecoderClass *klass;
gboolean ret;
dec = GST_AUDIO_DECODER (parent);
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
if (klass->src_event)
ret = klass->src_event (dec, event);
else {
gst_event_unref (event);
ret = FALSE;
}
return ret;
}
static gboolean
gst_audio_decoder_decide_allocation_default (GstAudioDecoder * dec,
GstQuery * query)
{
GstAllocator *allocator = NULL;
GstAllocationParams params;
gboolean update_allocator;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
/* try the allocator */
gst_query_parse_nth_allocation_param (query, 0, &allocator, &params);
update_allocator = TRUE;
} else {
allocator = NULL;
gst_allocation_params_init (&params);
update_allocator = FALSE;
}
if (update_allocator)
gst_query_set_nth_allocation_param (query, 0, allocator, &params);
else
gst_query_add_allocation_param (query, allocator, &params);
if (allocator)
gst_object_unref (allocator);
return TRUE;
}
static gboolean
gst_audio_decoder_propose_allocation_default (GstAudioDecoder * dec,
GstQuery * query)
{
return TRUE;
}
/**
* gst_audio_decoder_proxy_getcaps:
* @decoder: a #GstAudioDecoder
* @caps: (allow-none): initial caps
* @filter: (allow-none): filter caps
*
* Returns caps that express @caps (or sink template caps if @caps == NULL)
* restricted to rate/channels/... combinations supported by downstream
* elements.
*
* Returns: (transfer full): a #GstCaps owned by caller
*
* Since: 1.6
*/
GstCaps *
gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder, GstCaps * caps,
GstCaps * filter)
{
return __gst_audio_element_proxy_getcaps (GST_ELEMENT_CAST (decoder),
GST_AUDIO_DECODER_SINK_PAD (decoder),
GST_AUDIO_DECODER_SRC_PAD (decoder), caps, filter);
}
static GstCaps *
gst_audio_decoder_sink_getcaps (GstAudioDecoder * decoder, GstCaps * filter)
{
GstAudioDecoderClass *klass;
GstCaps *caps;
klass = GST_AUDIO_DECODER_GET_CLASS (decoder);
if (klass->getcaps)
caps = klass->getcaps (decoder, filter);
else
caps = gst_audio_decoder_proxy_getcaps (decoder, NULL, filter);
GST_LOG_OBJECT (decoder, "Returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_audio_decoder_sink_query_default (GstAudioDecoder * dec, GstQuery * query)
{
GstPad *pad = GST_AUDIO_DECODER_SINK_PAD (dec);
gboolean res = FALSE;
GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_FORMATS:
{
gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES);
res = TRUE;
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
GST_OBJECT_LOCK (dec);
res = __gst_audio_encoded_audio_convert (&dec->priv->ctx.info,
dec->priv->bytes_in, dec->priv->samples_out,
src_fmt, src_val, &dest_fmt, &dest_val);
GST_OBJECT_UNLOCK (dec);
if (!res)
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
case GST_QUERY_ALLOCATION:
{
GstAudioDecoderClass *klass = GST_AUDIO_DECODER_GET_CLASS (dec);
if (klass->propose_allocation)
res = klass->propose_allocation (dec, query);
break;
}
case GST_QUERY_CAPS:{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_audio_decoder_sink_getcaps (dec, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
case GST_QUERY_ACCEPT_CAPS:{
if (dec->priv->use_default_pad_acceptcaps) {
res =
gst_pad_query_default (GST_AUDIO_DECODER_SINK_PAD (dec),
GST_OBJECT_CAST (dec), query);
} else {
GstCaps *caps;
GstCaps *allowed_caps;
GstCaps *template_caps;
gboolean accept;
gst_query_parse_accept_caps (query, &caps);
template_caps = gst_pad_get_pad_template_caps (pad);
accept = gst_caps_is_subset (caps, template_caps);
gst_caps_unref (template_caps);
if (accept) {
allowed_caps = gst_pad_query_caps (GST_AUDIO_DECODER_SINK_PAD (dec),
caps);
accept = gst_caps_can_intersect (caps, allowed_caps);
gst_caps_unref (allowed_caps);
}
gst_query_set_accept_caps_result (query, accept);
res = TRUE;
}
break;
}
case GST_QUERY_SEEKING:
{
GstFormat format;
/* non-TIME segments are discarded, so we won't seek that way either */
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
if (format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (dec, "discarding non-TIME SEEKING query");
res = FALSE;
break;
}
/* fall-through */
}
default:
res = gst_pad_query_default (pad, GST_OBJECT_CAST (dec), query);
break;
}
error:
return res;
}
static gboolean
gst_audio_decoder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
GstAudioDecoderClass *dec_class;
GstAudioDecoder *dec;
gboolean ret = FALSE;
dec = GST_AUDIO_DECODER (parent);
dec_class = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (pad, "received query %" GST_PTR_FORMAT, query);
if (dec_class->sink_query)
ret = dec_class->sink_query (dec, query);
return ret;
}
/* FIXME ? are any of these queries (other than latency) a decoder's business ??
* also, the conversion stuff might seem to make sense, but seems to not mind
* segment stuff etc at all
* Supposedly that's backward compatibility ... */
static gboolean
gst_audio_decoder_src_query_default (GstAudioDecoder * dec, GstQuery * query)
{
GstPad *pad = GST_AUDIO_DECODER_SRC_PAD (dec);
gboolean res = FALSE;
GST_LOG_OBJECT (dec, "handling query: %" GST_PTR_FORMAT, query);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
{
GstFormat format;
/* upstream in any case */
if ((res = gst_pad_query_default (pad, GST_OBJECT_CAST (dec), query)))
break;
gst_query_parse_duration (query, &format, NULL);
/* try answering TIME by converting from BYTE if subclass allows */
if (format == GST_FORMAT_TIME && gst_audio_decoder_do_byte (dec)) {
gint64 value;
if (gst_pad_peer_query_duration (dec->sinkpad, GST_FORMAT_BYTES,
&value)) {
GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
GST_FORMAT_TIME, &value)) {
gst_query_set_duration (query, GST_FORMAT_TIME, value);
res = TRUE;
}
}
}
break;
}
case GST_QUERY_POSITION:
{
GstFormat format;
gint64 time, value;
if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
GST_LOG_OBJECT (dec, "returning peer response");
break;
}
/* Refuse BYTES format queries. If it made sense to
* answer them, upstream would have already */
gst_query_parse_position (query, &format, NULL);
if (format == GST_FORMAT_BYTES) {
GST_LOG_OBJECT (dec, "Ignoring BYTES position query");
break;
}
/* we start from the last seen time */
time = dec->output_segment.position;
/* correct for the segment values */
time =
gst_segment_to_stream_time (&dec->output_segment, GST_FORMAT_TIME,
time);
GST_LOG_OBJECT (dec,
"query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
/* and convert to the final format */
if (!(res = gst_pad_query_convert (pad, GST_FORMAT_TIME, time,
format, &value)))
break;
gst_query_set_position (query, format, value);
GST_LOG_OBJECT (dec,
"query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
format);
break;
}
case GST_QUERY_FORMATS:
{
gst_query_set_formats (query, 3,
GST_FORMAT_TIME, GST_FORMAT_BYTES, GST_FORMAT_DEFAULT);
res = TRUE;
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
GST_OBJECT_LOCK (dec);
res = gst_audio_info_convert (&dec->priv->ctx.info,
src_fmt, src_val, dest_fmt, &dest_val);
GST_OBJECT_UNLOCK (dec);
if (!res)
break;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
case GST_QUERY_LATENCY:
{
if ((res = gst_pad_peer_query (dec->sinkpad, query))) {
gboolean live;
GstClockTime min_latency, max_latency;
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
GST_DEBUG_OBJECT (dec, "Peer latency: live %d, min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT, live,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
GST_OBJECT_LOCK (dec);
/* add our latency */
min_latency += dec->priv->ctx.min_latency;
if (max_latency == -1 || dec->priv->ctx.max_latency == -1)
max_latency = -1;
else
max_latency += dec->priv->ctx.max_latency;
GST_OBJECT_UNLOCK (dec);
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query_default (pad, GST_OBJECT_CAST (dec), query);
break;
}
return res;
}
static gboolean
gst_audio_decoder_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAudioDecoder *dec;
GstAudioDecoderClass *dec_class;
gboolean ret = FALSE;
dec = GST_AUDIO_DECODER (parent);
dec_class = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (pad, "received query %" GST_PTR_FORMAT, query);
if (dec_class->src_query)
ret = dec_class->src_query (dec, query);
return ret;
}
static gboolean
gst_audio_decoder_stop (GstAudioDecoder * dec)
{
GstAudioDecoderClass *klass;
gboolean ret = TRUE;
GST_DEBUG_OBJECT (dec, "gst_audio_decoder_stop");
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
if (klass->stop) {
ret = klass->stop (dec);
}
/* clean up */
gst_audio_decoder_reset (dec, TRUE);
if (ret)
dec->priv->active = FALSE;
return ret;
}
static gboolean
gst_audio_decoder_start (GstAudioDecoder * dec)
{
GstAudioDecoderClass *klass;
gboolean ret = TRUE;
GST_DEBUG_OBJECT (dec, "gst_audio_decoder_start");
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
/* arrange clean state */
gst_audio_decoder_reset (dec, TRUE);
if (klass->start) {
ret = klass->start (dec);
}
if (ret)
dec->priv->active = TRUE;
return ret;
}
static void
gst_audio_decoder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioDecoder *dec;
dec = GST_AUDIO_DECODER (object);
switch (prop_id) {
case PROP_LATENCY:
g_value_set_int64 (value, dec->priv->latency);
break;
case PROP_TOLERANCE:
g_value_set_int64 (value, dec->priv->tolerance);
break;
case PROP_PLC:
g_value_set_boolean (value, dec->priv->plc);
break;
case PROP_MAX_ERRORS:
g_value_set_int (value, gst_audio_decoder_get_max_errors (dec));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_decoder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioDecoder *dec;
dec = GST_AUDIO_DECODER (object);
switch (prop_id) {
case PROP_LATENCY:
dec->priv->latency = g_value_get_int64 (value);
break;
case PROP_TOLERANCE:
dec->priv->tolerance = g_value_get_int64 (value);
break;
case PROP_PLC:
dec->priv->plc = g_value_get_boolean (value);
break;
case PROP_MAX_ERRORS:
gst_audio_decoder_set_max_errors (dec, g_value_get_int (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_audio_decoder_change_state (GstElement * element, GstStateChange transition)
{
GstAudioDecoder *codec;
GstAudioDecoderClass *klass;
GstStateChangeReturn ret;
codec = GST_AUDIO_DECODER (element);
klass = GST_AUDIO_DECODER_GET_CLASS (codec);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (klass->open) {
if (!klass->open (codec))
goto open_failed;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
if (!gst_audio_decoder_start (codec)) {
goto start_failed;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (!gst_audio_decoder_stop (codec)) {
goto stop_failed;
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (klass->close) {
if (!klass->close (codec))
goto close_failed;
}
break;
default:
break;
}
return ret;
start_failed:
{
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to start codec"));
return GST_STATE_CHANGE_FAILURE;
}
stop_failed:
{
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to stop codec"));
return GST_STATE_CHANGE_FAILURE;
}
open_failed:
{
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to open codec"));
return GST_STATE_CHANGE_FAILURE;
}
close_failed:
{
GST_ELEMENT_ERROR (codec, LIBRARY, INIT, (NULL), ("Failed to close codec"));
return GST_STATE_CHANGE_FAILURE;
}
}
GstFlowReturn
_gst_audio_decoder_error (GstAudioDecoder * dec, gint weight,
GQuark domain, gint code, gchar * txt, gchar * dbg, const gchar * file,
const gchar * function, gint line)
{
if (txt)
GST_WARNING_OBJECT (dec, "error: %s", txt);
if (dbg)
GST_WARNING_OBJECT (dec, "error: %s", dbg);
dec->priv->error_count += weight;
dec->priv->discont = TRUE;
if (dec->priv->max_errors >= 0
&& dec->priv->max_errors < dec->priv->error_count) {
gst_element_message_full (GST_ELEMENT (dec), GST_MESSAGE_ERROR, domain,
code, txt, dbg, file, function, line);
return GST_FLOW_ERROR;
} else {
g_free (txt);
g_free (dbg);
return GST_FLOW_OK;
}
}
/**
* gst_audio_decoder_get_audio_info:
* @dec: a #GstAudioDecoder
*
* Returns: (transfer none): a #GstAudioInfo describing the input audio format
*/
GstAudioInfo *
gst_audio_decoder_get_audio_info (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), NULL);
return &dec->priv->ctx.info;
}
/**
* gst_audio_decoder_set_plc_aware:
* @dec: a #GstAudioDecoder
* @plc: new plc state
*
* Indicates whether or not subclass handles packet loss concealment (plc).
*/
void
gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec, gboolean plc)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
dec->priv->ctx.do_plc = plc;
}
/**
* gst_audio_decoder_get_plc_aware:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured plc handling
*/
gint
gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->ctx.do_plc;
}
/**
* gst_audio_decoder_set_estimate_rate:
* @dec: a #GstAudioDecoder
* @enabled: whether to enable byte to time conversion
*
* Allows baseclass to perform byte to time estimated conversion.
*/
void
gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
dec->priv->ctx.do_estimate_rate = enabled;
}
/**
* gst_audio_decoder_get_estimate_rate:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured byte to time conversion setting
*/
gint
gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->ctx.do_estimate_rate;
}
/**
* gst_audio_decoder_get_delay:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured decoder delay
*/
gint
gst_audio_decoder_get_delay (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->ctx.delay;
}
/**
* gst_audio_decoder_set_max_errors:
* @dec: a #GstAudioDecoder
* @num: max tolerated errors
*
* Sets numbers of tolerated decoder errors, where a tolerated one is then only
* warned about, but more than tolerated will lead to fatal error. You can set
* -1 for never returning fatal errors. Default is set to
* GST_AUDIO_DECODER_MAX_ERRORS.
*/
void
gst_audio_decoder_set_max_errors (GstAudioDecoder * dec, gint num)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
dec->priv->max_errors = num;
}
/**
* gst_audio_decoder_get_max_errors:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured decoder tolerated error count.
*/
gint
gst_audio_decoder_get_max_errors (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->max_errors;
}
/**
* gst_audio_decoder_set_latency:
* @dec: a #GstAudioDecoder
* @min: minimum latency
* @max: maximum latency
*
* Sets decoder latency.
*/
void
gst_audio_decoder_set_latency (GstAudioDecoder * dec,
GstClockTime min, GstClockTime max)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (min));
g_return_if_fail (min <= max);
GST_OBJECT_LOCK (dec);
dec->priv->ctx.min_latency = min;
dec->priv->ctx.max_latency = max;
GST_OBJECT_UNLOCK (dec);
/* post latency message on the bus */
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_latency (GST_OBJECT (dec)));
}
/**
* gst_audio_decoder_get_latency:
* @dec: a #GstAudioDecoder
* @min: (out) (allow-none): a pointer to storage to hold minimum latency
* @max: (out) (allow-none): a pointer to storage to hold maximum latency
*
* Sets the variables pointed to by @min and @max to the currently configured
* latency.
*/
void
gst_audio_decoder_get_latency (GstAudioDecoder * dec,
GstClockTime * min, GstClockTime * max)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
GST_OBJECT_LOCK (dec);
if (min)
*min = dec->priv->ctx.min_latency;
if (max)
*max = dec->priv->ctx.max_latency;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_parse_state:
* @dec: a #GstAudioDecoder
* @sync: (out) (optional): a pointer to a variable to hold the current sync state
* @eos: (out) (optional): a pointer to a variable to hold the current eos state
*
* Return current parsing (sync and eos) state.
*/
void
gst_audio_decoder_get_parse_state (GstAudioDecoder * dec,
gboolean * sync, gboolean * eos)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
if (sync)
*sync = dec->priv->ctx.sync;
if (eos)
*eos = dec->priv->ctx.eos;
}
/**
* gst_audio_decoder_set_allocation_caps:
* @dec: a #GstAudioDecoder
* @allocation_caps: (allow-none): a #GstCaps or %NULL
*
* Sets a caps in allocation query which are different from the set
* pad's caps. Use this function before calling
* gst_audio_decoder_negotiate(). Setting to %NULL the allocation
* query will use the caps from the pad.
*
* Since: 1.10
*/
void
gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec,
GstCaps * allocation_caps)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
gst_caps_replace (&dec->priv->ctx.allocation_caps, allocation_caps);
}
/**
* gst_audio_decoder_set_plc:
* @dec: a #GstAudioDecoder
* @enabled: new state
*
* Enable or disable decoder packet loss concealment, provided subclass
* and codec are capable and allow handling plc.
*
* MT safe.
*/
void
gst_audio_decoder_set_plc (GstAudioDecoder * dec, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
GST_LOG_OBJECT (dec, "enabled: %d", enabled);
GST_OBJECT_LOCK (dec);
dec->priv->plc = enabled;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_plc:
* @dec: a #GstAudioDecoder
*
* Queries decoder packet loss concealment handling.
*
* Returns: TRUE if packet loss concealment is enabled.
*
* MT safe.
*/
gboolean
gst_audio_decoder_get_plc (GstAudioDecoder * dec)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
GST_OBJECT_LOCK (dec);
result = dec->priv->plc;
GST_OBJECT_UNLOCK (dec);
return result;
}
/**
* gst_audio_decoder_set_min_latency:
* @dec: a #GstAudioDecoder
* @num: new minimum latency
*
* Sets decoder minimum aggregation latency.
*
* MT safe.
*/
void
gst_audio_decoder_set_min_latency (GstAudioDecoder * dec, GstClockTime num)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (num));
GST_OBJECT_LOCK (dec);
dec->priv->latency = num;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_min_latency:
* @dec: a #GstAudioDecoder
*
* Queries decoder's latency aggregation.
*
* Returns: aggregation latency.
*
* MT safe.
*/
GstClockTime
gst_audio_decoder_get_min_latency (GstAudioDecoder * dec)
{
GstClockTime result;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
GST_OBJECT_LOCK (dec);
result = dec->priv->latency;
GST_OBJECT_UNLOCK (dec);
return result;
}
/**
* gst_audio_decoder_set_tolerance:
* @dec: a #GstAudioDecoder
* @tolerance: new tolerance
*
* Configures decoder audio jitter tolerance threshold.
*
* MT safe.
*/
void
gst_audio_decoder_set_tolerance (GstAudioDecoder * dec, GstClockTime tolerance)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (tolerance));
GST_OBJECT_LOCK (dec);
dec->priv->tolerance = tolerance;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_tolerance:
* @dec: a #GstAudioDecoder
*
* Queries current audio jitter tolerance threshold.
*
* Returns: decoder audio jitter tolerance threshold.
*
* MT safe.
*/
GstClockTime
gst_audio_decoder_get_tolerance (GstAudioDecoder * dec)
{
GstClockTime result;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
GST_OBJECT_LOCK (dec);
result = dec->priv->tolerance;
GST_OBJECT_UNLOCK (dec);
return result;
}
/**
* gst_audio_decoder_set_drainable:
* @dec: a #GstAudioDecoder
* @enabled: new state
*
* Configures decoder drain handling. If drainable, subclass might
* be handed a NULL buffer to have it return any leftover decoded data.
* Otherwise, it is not considered so capable and will only ever be passed
* real data.
*
* MT safe.
*/
void
gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
GST_OBJECT_LOCK (dec);
dec->priv->drainable = enabled;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_drainable:
* @dec: a #GstAudioDecoder
*
* Queries decoder drain handling.
*
* Returns: TRUE if drainable handling is enabled.
*
* MT safe.
*/
gboolean
gst_audio_decoder_get_drainable (GstAudioDecoder * dec)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
GST_OBJECT_LOCK (dec);
result = dec->priv->drainable;
GST_OBJECT_UNLOCK (dec);
return result;
}
/**
* gst_audio_decoder_set_needs_format:
* @dec: a #GstAudioDecoder
* @enabled: new state
*
* Configures decoder format needs. If enabled, subclass needs to be
* negotiated with format caps before it can process any data. It will then
* never be handed any data before it has been configured.
* Otherwise, it might be handed data without having been configured and
* is then expected being able to do so either by default
* or based on the input data.
*
* MT safe.
*/
void
gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
GST_OBJECT_LOCK (dec);
dec->priv->needs_format = enabled;
GST_OBJECT_UNLOCK (dec);
}
/**
* gst_audio_decoder_get_needs_format:
* @dec: a #GstAudioDecoder
*
* Queries decoder required format handling.
*
* Returns: TRUE if required format handling is enabled.
*
* MT safe.
*/
gboolean
gst_audio_decoder_get_needs_format (GstAudioDecoder * dec)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), FALSE);
GST_OBJECT_LOCK (dec);
result = dec->priv->needs_format;
GST_OBJECT_UNLOCK (dec);
return result;
}
/**
* gst_audio_decoder_merge_tags:
* @dec: a #GstAudioDecoder
* @tags: (allow-none): a #GstTagList to merge, or NULL
* @mode: the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*
* Sets the audio decoder tags and how they should be merged with any
* upstream stream tags. This will override any tags previously-set
* with gst_audio_decoder_merge_tags().
*
* Note that this is provided for convenience, and the subclass is
* not required to use this and can still do tag handling on its own.
*/
void
gst_audio_decoder_merge_tags (GstAudioDecoder * dec,
const GstTagList * tags, GstTagMergeMode mode)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
g_return_if_fail (mode != GST_TAG_MERGE_UNDEFINED);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (dec->priv->taglist != tags) {
if (dec->priv->taglist) {
gst_tag_list_unref (dec->priv->taglist);
dec->priv->taglist = NULL;
dec->priv->decoder_tags_merge_mode = GST_TAG_MERGE_KEEP_ALL;
}
if (tags) {
dec->priv->taglist = gst_tag_list_ref ((GstTagList *) tags);
dec->priv->decoder_tags_merge_mode = mode;
}
GST_DEBUG_OBJECT (dec, "setting decoder tags to %" GST_PTR_FORMAT, tags);
dec->priv->taglist_changed = TRUE;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
}
/**
* gst_audio_decoder_allocate_output_buffer:
* @dec: a #GstAudioDecoder
* @size: size of the buffer
*
* Helper function that allocates a buffer to hold an audio frame
* for @dec's current output format.
*
* Returns: (transfer full): allocated buffer
*/
GstBuffer *
gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec, gsize size)
{
GstBuffer *buffer = NULL;
gboolean needs_reconfigure = FALSE;
g_return_val_if_fail (size > 0, NULL);
GST_DEBUG ("alloc src buffer");
GST_AUDIO_DECODER_STREAM_LOCK (dec);
needs_reconfigure = gst_pad_check_reconfigure (dec->srcpad);
if (G_UNLIKELY (dec->priv->ctx.output_format_changed ||
(GST_AUDIO_INFO_IS_VALID (&dec->priv->ctx.info)
&& needs_reconfigure))) {
if (!gst_audio_decoder_negotiate_unlocked (dec)) {
GST_INFO_OBJECT (dec, "Failed to negotiate, fallback allocation");
gst_pad_mark_reconfigure (dec->srcpad);
goto fallback;
}
}
buffer =
gst_buffer_new_allocate (dec->priv->ctx.allocator, size,
&dec->priv->ctx.params);
if (!buffer) {
GST_INFO_OBJECT (dec, "couldn't allocate output buffer");
goto fallback;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return buffer;
fallback:
buffer = gst_buffer_new_allocate (NULL, size, NULL);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return buffer;
}
/**
* gst_audio_decoder_get_allocator:
* @dec: a #GstAudioDecoder
* @allocator: (out) (allow-none) (transfer full): the #GstAllocator
* used
* @params: (out) (allow-none) (transfer full): the
* #GstAllocationParams of @allocator
*
* Lets #GstAudioDecoder sub-classes to know the memory @allocator
* used by the base class and its @params.
*
* Unref the @allocator after use it.
*/
void
gst_audio_decoder_get_allocator (GstAudioDecoder * dec,
GstAllocator ** allocator, GstAllocationParams * params)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
if (allocator)
*allocator = dec->priv->ctx.allocator ?
gst_object_ref (dec->priv->ctx.allocator) : NULL;
if (params)
*params = dec->priv->ctx.params;
}
/**
* gst_audio_decoder_set_use_default_pad_acceptcaps:
* @decoder: a #GstAudioDecoder
* @use: if the default pad accept-caps query handling should be used
*
* Lets #GstAudioDecoder sub-classes decide if they want the sink pad
* to use the default pad query handler to reply to accept-caps queries.
*
* By setting this to true it is possible to further customize the default
* handler with %GST_PAD_SET_ACCEPT_INTERSECT and
* %GST_PAD_SET_ACCEPT_TEMPLATE
*
* Since: 1.6
*/
void
gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder,
gboolean use)
{
decoder->priv->use_default_pad_acceptcaps = use;
}
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