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gstreamer/gst-libs/gst/audio/gstaudiodecoder.c
Mark Nauwelaerts 9f57d91137 audiodecoder: add some documentation
2011-11-28 11:37:27 +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
#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));
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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