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gstreamer/gst-libs/gst/audio/gstaudiodecoder.c
Mark Nauwelaerts c41f3cbef0 audiodecoder: set a non-zero default maximum tolerated errors 
Whereas the previous default 0 was backwards compatible in that it lead
to erroring out immediately upon any error, elements that are really
ported and using the base class error macro can be assumed to intend to
improve behaviour rather than maintaining the old one.  So, make it easy
on those and any future one and tolerate some errors by default, as intended.

Fixes #666579.
2011-12-20 12:50:18 +01:00

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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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstaudiodecoder
* @short_description: Base class for audio decoders
* @see_also: #GstBaseTransform
* @since: 0.10.36
*
* This base class is for audio decoders turning encoded data into
* raw audio samples.
*
* GstAudioDecoder and subclass should cooperate as follows.
* <orderedlist>
* <listitem>
* <itemizedlist><title>Configuration</title>
* <listitem><para>
* 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).
* </para></listitem>
* <listitem><para>
* 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.
* </para></listitem>
* <listitem><para>
* GstAudioDecoder calls @stop at end of all processing.
* </para></listitem>
* </itemizedlist>
* </listitem>
* 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.
* <listitem>
* <itemizedlist>
* <title>Data processing</title>
* <listitem><para>
* 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.
* </para></listitem>
* <listitem><para>
* Input frame is provided to subclass' @handle_frame.
* </para></listitem>
* <listitem><para>
* If codec processing results in decoded data, subclass should call
* @gst_audio_decoder_finish_frame to have decoded data pushed
* downstream.
* </para></listitem>
* <listitem><para>
* 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.
* </para></listitem>
* <listitem><para>
* 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.
* </para></listitem>
* </itemizedlist>
* </listitem>
* <listitem>
* <itemizedlist><title>Shutdown phase</title>
* <listitem><para>
* GstAudioDecoder class calls @stop to inform the subclass that data
* parsing will be stopped.
* </para></listitem>
* </itemizedlist>
* </listitem>
* </orderedlist>
*
* 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.
*
* 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:
* <itemizedlist>
* <listitem><para>Provide pad templates</para></listitem>
* <listitem><para>
* Set source pad caps when appropriate
* </para></listitem>
* <listitem><para>
* Set user-configurable properties to sane defaults for format and
* implementing codec at hand, and convey some subclass capabilities and
* expectations in context.
* </para></listitem>
* <listitem><para>
* 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.
* </para></listitem>
* </itemizedlist>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include "gstaudiodecoder.h"
#include <gst/pbutils/descriptions.h>
#include <string.h>
GST_DEBUG_CATEGORY (audiodecoder_debug);
#define GST_CAT_DEFAULT audiodecoder_debug
#define GST_AUDIO_DECODER_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_DECODER, \
GstAudioDecoderPrivate))
enum
{
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LATENCY,
PROP_TOLERANCE,
PROP_PLC
};
#define DEFAULT_LATENCY 0
#define DEFAULT_TOLERANCE 0
#define DEFAULT_PLC FALSE
typedef struct _GstAudioDecoderContext
{
/* input */
/* (output) audio format */
GstAudioInfo info;
/* parsing state */
gboolean eos;
gboolean sync;
/* misc */
gint delay;
/* output */
gboolean do_plc;
gboolean do_byte_time;
gint max_errors;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
} 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;
/* 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 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;
/* codec id tag */
GstTagList *taglist;
/* 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;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
};
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_event (GstPad * pad, GstEvent * event);
static gboolean gst_audio_decoder_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_audio_decoder_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_audio_decoder_src_query (GstPad * pad, GstQuery * query);
static gboolean gst_audio_decoder_sink_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_audio_decoder_get_query_types (GstPad * pad);
static void gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full);
GST_BOILERPLATE (GstAudioDecoder, gst_audio_decoder, GstElement,
GST_TYPE_ELEMENT);
static void
gst_audio_decoder_base_init (gpointer g_class)
{
}
static void
gst_audio_decoder_class_init (GstAudioDecoderClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
gobject_class = G_OBJECT_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
g_type_class_add_private (klass, sizeof (GstAudioDecoderPrivate));
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_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));
}
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_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_setcaps_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_sink_setcaps));
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_setcaps_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_src_setcaps));
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_pad_set_query_type_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_decoder_get_query_types));
gst_pad_use_fixed_caps (dec->srcpad);
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_static_rec_mutex_init (&dec->stream_lock);
/* property default */
dec->priv->latency = DEFAULT_LATENCY;
dec->priv->tolerance = DEFAULT_TOLERANCE;
dec->priv->plc = DEFAULT_PLC;
/* init state */
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;
dec->priv->bytes_in = 0;
dec->priv->samples_out = 0;
dec->priv->agg = -1;
dec->priv->error_count = 0;
gst_audio_decoder_clear_queues (dec);
gst_audio_info_clear (&dec->priv->ctx.info);
memset (&dec->priv->ctx, 0, sizeof (dec->priv->ctx));
dec->priv->ctx.max_errors = GST_AUDIO_DECODER_MAX_ERRORS;
if (dec->priv->taglist) {
gst_tag_list_free (dec->priv->taglist);
dec->priv->taglist = NULL;
}
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (dec->priv->pending_events);
dec->priv->pending_events = NULL;
}
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->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_static_rec_mutex_free (&dec->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* automagically perform sanity checking of src caps;
* also extracts output data format */
static gboolean
gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioDecoder *dec;
gboolean res = TRUE;
guint old_rate;
dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* parse caps here to check subclass;
* also makes us aware of output format */
if (!gst_caps_is_fixed (caps))
goto refuse_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;
}
if (!gst_audio_info_from_caps (&dec->priv->ctx.info, caps))
goto refuse_caps;
done:
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
gst_object_unref (dec);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
res = FALSE;
goto done;
}
}
static gboolean
gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioDecoder *dec;
GstAudioDecoderClass *klass;
gboolean res = TRUE;
dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* NOTE pbutils only needed here */
/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
if (dec->priv->taglist)
gst_tag_list_free (dec->priv->taglist);
dec->priv->taglist = gst_tag_list_new ();
gst_pb_utils_add_codec_description_to_tag_list (dec->priv->taglist,
GST_TAG_AUDIO_CODEC, caps);
if (klass->set_format)
res = klass->set_format (dec, caps);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
g_object_unref (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;
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;
}
GST_LOG_OBJECT (dec, "clipping buffer of size %d with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* clip buffer */
buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
ctx->info.bpf);
if (G_UNLIKELY (!buf)) {
GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
goto exit;
}
/* decorate */
gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
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_TIMESTAMP_IS_VALID (buf))) {
/* duration should always be valid for raw audio */
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
dec->segment.last_stop =
GST_BUFFER_TIMESTAMP (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 %d with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (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 %d with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
again:
inbuf = NULL;
if (priv->agg && dec->priv->latency > 0) {
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_TIMESTAMP_IS_VALID (buf) ||
!GST_CLOCK_TIME_IS_VALID (priv->out_ts) ||
((diff = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (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_TIMESTAMP (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_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_TIMESTAMP (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->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;
}
/**
* gst_audio_decoder_finish_frame:
* @dec: a #GstAudioDecoder
* @buf: 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 gst_audio_decoder_handle_frame() may be
* invalidated by a call to this function.
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*
* Since: 0.10.36
*/
GstFlowReturn
gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
gint frames)
{
GstAudioDecoderPrivate *priv;
GstAudioDecoderContext *ctx;
gint samples = 0;
GstClockTime ts, next_ts;
GstFlowReturn ret = GST_FLOW_OK;
/* subclass should know what it is producing by now */
g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
GST_FLOW_ERROR);
/* subclass should not hand us no data */
g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
GST_FLOW_ERROR);
/* no dummy calls please */
g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
priv = dec->priv;
ctx = &dec->priv->ctx;
/* 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 %d bytes == %d samples for %d frames",
buf ? GST_BUFFER_SIZE (buf) : -1,
buf ? GST_BUFFER_SIZE (buf) / ctx->info.bpf : -1, frames);
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (priv->pending_events) {
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_pad_push_event (dec->srcpad, l->data);
g_list_free (pending_events);
}
/* output shoud be whole number of sample frames */
if (G_LIKELY (buf && ctx->info.bpf)) {
if (GST_BUFFER_SIZE (buf) % ctx->info.bpf)
goto wrong_buffer;
/* per channel least */
samples = GST_BUFFER_SIZE (buf) / ctx->info.bpf;
}
/* frame and ts book-keeping */
if (G_UNLIKELY (frames < 0)) {
if (G_UNLIKELY (-frames - 1 > priv->frames.length))
goto overflow;
frames = priv->frames.length + frames + 1;
} else if (G_UNLIKELY (frames > priv->frames.length)) {
if (G_LIKELY (!priv->force)) {
/* no way we can let this pass */
g_assert_not_reached ();
/* really no way */
goto overflow;
}
}
if (G_LIKELY (priv->frames.length))
ts = GST_BUFFER_TIMESTAMP (priv->frames.head->data);
else
ts = GST_CLOCK_TIME_NONE;
GST_DEBUG_OBJECT (dec, "leading frame ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (ts));
while (priv->frames.length && frames) {
gst_buffer_unref (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));
}
/* slightly convoluted approach caters for perfect ts if subclass desires */
if (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) {
GST_DEBUG_OBJECT (dec, "codec tag %" GST_PTR_FORMAT, priv->taglist);
if (gst_tag_list_is_empty (priv->taglist)) {
gst_tag_list_free (priv->taglist);
} else {
gst_element_found_tags (GST_ELEMENT (dec), priv->taglist);
}
priv->taglist = NULL;
}
buf = gst_buffer_make_metadata_writable (buf);
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
GST_BUFFER_TIMESTAMP (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_TIMESTAMP (buf);
} else {
GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buf) =
GST_FRAMES_TO_CLOCK_TIME (samples, ctx->info.rate);
}
priv->samples += samples;
priv->samples_out += samples;
/* we got data, so note things are looking up */
if (G_UNLIKELY (dec->priv->error_count))
dec->priv->error_count--;
exit:
ret = gst_audio_decoder_output (dec, buf);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
return ret;
/* ERRORS */
wrong_buffer:
{
GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
ctx->info.bpf));
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
overflow:
{
GST_ELEMENT_ERROR (dec, STREAM, ENCODE,
("received more decoded frames %d than provided %d", frames,
priv->frames.length), (NULL));
if (buf)
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
}
static GstFlowReturn
gst_audio_decoder_handle_frame (GstAudioDecoder * dec,
GstAudioDecoderClass * klass, GstBuffer * buffer)
{
if (G_LIKELY (buffer)) {
/* keep around for admin */
GST_LOG_OBJECT (dec, "tracking frame size %d, ts %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (buffer),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
g_queue_push_tail (&dec->priv->frames, buffer);
dec->priv->ctx.delay = dec->priv->frames.length;
dec->priv->bytes_in += GST_BUFFER_SIZE (buffer);
} 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;
/* 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, "setting DISCONT");
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_UNEXPECTED) {
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 (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_timestamp (priv->adapter, NULL);
if (ts == priv->prev_ts) {
GST_LOG_OBJECT (dec, "ts == prev_ts; discarding");
ts = GST_CLOCK_TIME_NONE;
} else {
priv->prev_ts = ts;
}
buffer = gst_adapter_take_buffer (priv->adapter, len);
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) = ts;
flush += len;
} else {
if (!force)
break;
buffer = NULL;
}
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;
else {
/* dispatch reverse pending buffers */
/* chain eventually calls upon drain as well, but by that time
* gather list should be clear, so ok ... */
if (dec->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);
/* ensure all output sent */
ret = gst_audio_decoder_output (dec, NULL);
/* 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->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_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 incomming 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_TIMESTAMP (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);
/* duration should always be valid for raw audio */
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
/* interpolate (backward) if needed */
if (G_LIKELY (timestamp != -1))
timestamp -= GST_BUFFER_DURATION (buf);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
GST_LOG_OBJECT (dec, "applying reverse interpolated ts %"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp));
GST_BUFFER_TIMESTAMP (buf) = timestamp;
} else {
/* track otherwise */
timestamp = GST_BUFFER_TIMESTAMP (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 %u, "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* should be already, but let's be sure */
buf = gst_buffer_make_metadata_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 %u, "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (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, GstBuffer * buffer)
{
GstAudioDecoder *dec;
GstFlowReturn ret;
dec = GST_AUDIO_DECODER (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (dec,
"received buffer of size %d with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (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->segment.rate > 0.0 && !GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
GST_DEBUG_OBJECT (dec, "... but restoring previous ts tracking");
dec->priv->base_ts = ts;
dec->priv->samples = samples;
}
}
if (dec->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;
}
/* 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)
{
return dec->priv->ctx.do_byte_time && dec->priv->ctx.info.bpf &&
dec->priv->ctx.info.rate <= dec->priv->samples_out;
}
static gboolean
gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
{
gboolean handled = FALSE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
if (format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (dec, "received TIME NEW_SEGMENT %" GST_TIME_FORMAT
" -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT
", rate %g, applied_rate %g",
GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
rate, arate);
} else {
GstFormat dformat = GST_FORMAT_TIME;
GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
" -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
", rate %g, applied_rate %g", start, stop, time, rate, arate);
/* 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_byte_time &&
gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, start,
&dformat, &start)) {
/* 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 (start));
format = GST_FORMAT_TIME;
time = start;
stop = GST_CLOCK_TIME_NONE;
/* replace event */
gst_event_unref (event);
event = gst_event_new_new_segment_full (update, rate, arate,
GST_FORMAT_TIME, start, stop, time);
} else {
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
}
/* finish current segment */
gst_audio_decoder_drain (dec);
if (update) {
/* time progressed without data, see if we can fill the gap with
* some concealment data */
GST_DEBUG_OBJECT (dec,
"segment update: plc %d, do_plc %d, last_stop %" GST_TIME_FORMAT,
dec->priv->plc, dec->priv->ctx.do_plc,
GST_TIME_ARGS (dec->segment.last_stop));
if (dec->priv->plc && dec->priv->ctx.do_plc &&
dec->segment.rate > 0.0 && dec->segment.last_stop < start) {
GstAudioDecoderClass *klass;
GstBuffer *buf;
klass = GST_AUDIO_DECODER_GET_CLASS (dec);
/* hand subclass empty frame with duration that needs covering */
buf = gst_buffer_new ();
GST_BUFFER_DURATION (buf) = start - dec->segment.last_stop;
/* best effort, not much error handling */
gst_audio_decoder_handle_frame (dec, klass, buf);
}
} else {
/* prepare for next one */
gst_audio_decoder_flush (dec, FALSE);
/* and that's where we time from,
* in case upstream does not come up with anything better
* (e.g. upstream BYTE) */
if (format != GST_FORMAT_TIME) {
dec->priv->base_ts = start;
dec->priv->samples = 0;
}
}
/* and follow along with segment */
gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
format, start, stop, time);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
handled = TRUE;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* prepare for fresh start */
gst_audio_decoder_flush (dec, 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;
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
case GST_EVENT_EOS:
GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_audio_decoder_drain (dec);
GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
default:
break;
}
return handled;
}
static gboolean
gst_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
{
GstAudioDecoder *dec;
GstAudioDecoderClass *klass;
gboolean handled = FALSE;
gboolean ret = TRUE;
dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
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->event)
handled = klass->event (dec, event);
if (!handled)
handled = gst_audio_decoder_sink_eventfunc (dec, event);
if (!handled) {
/* Forward non-serialized events and EOS/FLUSH_STOP immediately.
* For EOS this is required because no buffer or serialized event
* will come after EOS and nothing could trigger another
* _finish_frame() call.
*
* For FLUSH_STOP this is required because it is expected
* to be forwarded immediately and no buffers are queued anyway.
*/
if (!GST_EVENT_IS_SERIALIZED (event)
|| GST_EVENT_TYPE (event) == GST_EVENT_EOS
|| GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
ret = gst_pad_event_default (pad, event);
} else {
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;
}
}
GST_DEBUG_OBJECT (dec, "event handled");
gst_object_unref (dec);
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_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->segment, sizeof (seek_segment));
gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
start_time, end_type, end_time, NULL);
start_time = seek_segment.last_stop;
format = GST_FORMAT_BYTES;
if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
&format, &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_event (GstPad * pad, GstEvent * event)
{
GstAudioDecoder *dec;
gboolean res = FALSE;
dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
if (G_UNLIKELY (dec == NULL)) {
gst_event_unref (event);
return FALSE;
}
GST_DEBUG_OBJECT (dec, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
GstFormat format, tformat;
gdouble rate;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 tcur, tstop;
guint32 seqnum;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
&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 */
tformat = GST_FORMAT_TIME;
if (!(res = gst_pad_query_convert (pad, format, cur, &tformat, &tcur)))
goto convert_error;
if (!(res = gst_pad_query_convert (pad, format, stop, &tformat, &tstop)))
goto convert_error;
/* then seek with time on the peer */
event = gst_event_new_seek (rate, GST_FORMAT_TIME,
flags, cur_type, tcur, stop_type, tstop);
gst_event_set_seqnum (event, seqnum);
res = gst_pad_push_event (dec->sinkpad, event);
break;
}
default:
res = gst_pad_push_event (dec->sinkpad, event);
break;
}
done:
gst_object_unref (dec);
return res;
/* ERRORS */
convert_error:
{
GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
goto done;
}
}
/*
* gst_audio_encoded_audio_convert:
* @fmt: audio format of the encoded audio
* @bytes: number of encoded bytes
* @samples: number of encoded samples
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE and TIME format by using estimated bitrate based on
* @samples and @bytes (and @fmt).
*/
/* FIXME: make gst_audio_encoded_audio_convert() public? */
static gboolean
gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
if (samples == 0 || bytes == 0 || fmt->rate == 0) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
bytes *= fmt->rate;
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value,
GST_SECOND * samples, bytes);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = gst_util_uint64_scale (src_value, bytes,
samples * GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}
static gboolean
gst_audio_decoder_sink_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstAudioDecoder *dec;
dec = GST_AUDIO_DECODER (gst_pad_get_parent (pad));
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);
if (!(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)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
error:
gst_object_unref (dec);
return res;
}
static const GstQueryType *
gst_audio_decoder_get_query_types (GstPad * pad)
{
static const GstQueryType gst_audio_decoder_src_query_types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
GST_QUERY_LATENCY,
0
};
return gst_audio_decoder_src_query_types;
}
/* 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 (GstPad * pad, GstQuery * query)
{
GstAudioDecoder *dec;
GstPad *peerpad;
gboolean res = FALSE;
dec = GST_AUDIO_DECODER (GST_PAD_PARENT (pad));
if (G_UNLIKELY (dec == NULL))
return FALSE;
peerpad = gst_pad_get_peer (GST_PAD (dec->sinkpad));
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, 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;
format = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (dec->sinkpad, &format, &value)) {
GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
format = GST_FORMAT_TIME;
if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
&format, &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;
}
/* we start from the last seen time */
time = dec->segment.last_stop;
/* correct for the segment values */
time = gst_segment_to_stream_time (&dec->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 */
gst_query_parse_position (query, &format, NULL);
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);
if (!(res = gst_audio_info_convert (&dec->priv->ctx.info,
src_fmt, src_val, dest_fmt, &dest_val)))
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 */
if (min_latency != -1)
min_latency += dec->priv->ctx.min_latency;
if (max_latency != -1)
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, query);
break;
}
gst_object_unref (peerpad);
return res;
}
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 TRUE;
}
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 TRUE;
}
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;
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;
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;
GstStateChangeReturn ret;
codec = GST_AUDIO_DECODER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
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 = 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:
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;
}
}
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->ctx.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 {
return GST_FLOW_OK;
}
}
/**
* gst_audio_decoder_get_audio_info:
* @dec: a #GstAudioDecoder
*
* Returns: a #GstAudioInfo describing the input audio format
*
* Since: 0.10.36
*/
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).
*
* Since: 0.10.36
*/
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
*
* Since: 0.10.36
*/
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_byte_time:
* @dec: a #GstAudioDecoder
* @enabled: whether to enable byte to time conversion
*
* Allows baseclass to perform byte to time estimated conversion.
*
* Since: 0.10.36
*/
void
gst_audio_decoder_set_byte_time (GstAudioDecoder * dec, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
dec->priv->ctx.do_byte_time = enabled;
}
/**
* gst_audio_decoder_get_byte_time:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured byte to time conversion setting
*
* Since: 0.10.36
*/
gint
gst_audio_decoder_get_byte_time (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->ctx.do_byte_time;
}
/**
* gst_audio_decoder_get_delay:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured decoder delay
*
* Since: 0.10.36
*/
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.
*
* Since: 0.10.36
*/
void
gst_audio_decoder_set_max_errors (GstAudioDecoder * dec, gint num)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
dec->priv->ctx.max_errors = num;
}
/**
* gst_audio_decoder_get_max_errors:
* @dec: a #GstAudioDecoder
*
* Returns: currently configured decoder tolerated error count.
*
* Since: 0.10.36
*/
gint
gst_audio_decoder_get_max_errors (GstAudioDecoder * dec)
{
g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
return dec->priv->ctx.max_errors;
}
/**
* gst_audio_decoder_set_latency:
* @dec: a #GstAudioDecoder
* @min: minimum latency
* @max: maximum latency
*
* Sets decoder latency.
*
* Since: 0.10.36
*/
void
gst_audio_decoder_set_latency (GstAudioDecoder * dec,
GstClockTime min, GstClockTime max)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
GST_OBJECT_LOCK (dec);
dec->priv->ctx.min_latency = min;
dec->priv->ctx.max_latency = max;
GST_OBJECT_UNLOCK (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.
*
* Since: 0.10.36
*/
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: a pointer to a variable to hold the current sync state
* @eos: a pointer to a variable to hold the current eos state
*
* Return current parsing (sync and eos) state.
*
* Since: 0.10.36
*/
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_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.
*
* Since: 0.10.36
*/
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.
*
* Since: 0.10.36
*/
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.
*
* Since: 0.10.36
*/
void
gst_audio_decoder_set_min_latency (GstAudioDecoder * dec, gint64 num)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
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.
*
* Since: 0.10.36
*/
gint64
gst_audio_decoder_get_min_latency (GstAudioDecoder * dec)
{
gint64 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.
*
* Since: 0.10.36
*/
void
gst_audio_decoder_set_tolerance (GstAudioDecoder * dec, gint64 tolerance)
{
g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
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.
*
* Since: 0.10.36
*/
gint64
gst_audio_decoder_get_tolerance (GstAudioDecoder * dec)
{
gint64 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;
}
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