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gstreamer/gst-libs/gst/audio/gstaudioencoder.c
Tim-Philipp Müller cdb22274e6 audioencoder: make stop() vfunc also optional 
Just change default value, since we also don't want to fail
if we want to deactivate and aren't active or want to activate
and are already active.

https://bugzilla.gnome.org/show_bug.cgi?id=685490
2012-10-04 13:40:32 +01:00

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/* GStreamer
* 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:gstaudioencoder
* @short_description: Base class for audio encoders
* @see_also: #GstBaseTransform
*
* This base class is for audio encoders turning raw audio samples into
* encoded audio data.
*
* GstAudioEncoder and subclass should cooperate as follows.
* <orderedlist>
* <listitem>
* <itemizedlist><title>Configuration</title>
* <listitem><para>
* Initially, GstAudioEncoder calls @start when the encoder element
* is activated, which allows subclass to perform any global setup.
* </para></listitem>
* <listitem><para>
* GstAudioEncoder calls @set_format to inform subclass of the format
* of input audio data that it is about to receive. Subclass should
* setup for encoding and configure various base class parameters
* appropriately, notably those directing desired input data handling.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
* </para></listitem>
* <listitem><para>
* GstAudioEncoder calls @stop at end of all processing.
* </para></listitem>
* </itemizedlist>
* </listitem>
* As of configuration stage, and throughout processing, GstAudioEncoder
* maintains various parameters that provide required context,
* e.g. describing the format of input audio data.
* Conversely, subclass can and should configure these context parameters
* to inform base class of its expectation w.r.t. buffer handling.
* <listitem>
* <itemizedlist>
* <title>Data processing</title>
* <listitem><para>
* Base class gathers input sample data (as directed by the context's
* frame_samples and frame_max) and provides this to subclass' @handle_frame.
* </para></listitem>
* <listitem><para>
* If codec processing results in encoded data, subclass should call
* @gst_audio_encoder_finish_frame to have encoded data pushed
* downstream. Alternatively, it might also call to indicate dropped
* (non-encoded) samples.
* </para></listitem>
* <listitem><para>
* Just prior to actually pushing a buffer downstream,
* it is passed to @pre_push.
* </para></listitem>
* <listitem><para>
* During the parsing process GstAudioEncoderClass 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>
* GstAudioEncoder 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_encoder_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 either favor tracking upstream timestamps
* (at the possible expense of jitter) or aim to arrange for a perfect stream of
* output timestamps, depending on #GstAudioEncoder:perfect-timestamp.
* However, in the latter case, the input may not be so perfect or ideal, which
* is handled as follows. An input timestamp is compared with the expected
* timestamp as dictated by input sample stream and if the deviation is less
* than #GstAudioEncoder:tolerance, the deviation is discarded.
* Otherwise, it is considered a discontuinity and subsequent output timestamp
* is resynced to the new position after performing configured discontinuity
* processing. In the non-perfect-timestamp case, an upstream variation
* exceeding tolerance only leads to marking DISCONT on subsequent outgoing
* (while timestamps are adjusted to upstream regardless of variation).
* While DISCONT is also marked in the perfect-timestamp case, this one
* optionally (see #GstAudioEncoder:hard-resync)
* performs some additional steps, such as clipping of (early) input samples
* or draining all currently remaining input data, depending on the direction
* of the discontuinity.
*
* If perfect timestamps are arranged, it is also possible to request baseclass
* (usually set by subclass) to provide additional buffer metadata (in OFFSET
* and OFFSET_END) fields according to granule defined semantics currently
* needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
* including buffer) and OFFSET_END to corresponding timestamp (as determined
* by same sample count and sample rate).
*
* 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>
* Inform base class of buffer processing needs using context's
* frame_samples and frame_bytes.
* </para></listitem>
* <listitem><para>
* Set user-configurable properties to sane defaults for format and
* implementing codec at hand, e.g. those controlling timestamp behaviour
* and discontinuity processing.
* </para></listitem>
* <listitem><para>
* Accept data in @handle_frame and provide encoded results to
* @gst_audio_encoder_finish_frame.
* </para></listitem>
* </itemizedlist>
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstaudioencoder.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
#include <gst/pbutils/descriptions.h>
#include <stdlib.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_audio_encoder_debug);
#define GST_CAT_DEFAULT gst_audio_encoder_debug
#define GST_AUDIO_ENCODER_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_ENCODER, \
GstAudioEncoderPrivate))
enum
{
PROP_0,
PROP_PERFECT_TS,
PROP_GRANULE,
PROP_HARD_RESYNC,
PROP_TOLERANCE
};
#define DEFAULT_PERFECT_TS FALSE
#define DEFAULT_GRANULE FALSE
#define DEFAULT_HARD_RESYNC FALSE
#define DEFAULT_TOLERANCE 40000000
#define DEFAULT_HARD_MIN FALSE
#define DEFAULT_DRAINABLE TRUE
typedef struct _GstAudioEncoderContext
{
/* input */
GstAudioInfo info;
/* output */
GstCaps *caps;
gboolean output_caps_changed;
gint frame_samples_min, frame_samples_max;
gint frame_max;
gint lookahead;
/* MT-protected (with LOCK) */
GstClockTime min_latency;
GstClockTime max_latency;
GList *headers;
gboolean new_headers;
GstAllocator *allocator;
GstAllocationParams params;
} GstAudioEncoderContext;
struct _GstAudioEncoderPrivate
{
/* activation status */
gboolean active;
/* input base/first ts as basis for output ts;
* kept nearly constant for perfect_ts,
* otherwise resyncs to upstream ts */
GstClockTime base_ts;
/* corresponding base granulepos */
gint64 base_gp;
/* input samples processed and sent downstream so far (w.r.t. base_ts) */
guint64 samples;
/* currently collected sample data */
GstAdapter *adapter;
/* offset in adapter up to which already supplied to encoder */
gint offset;
/* mark outgoing discont */
gboolean discont;
/* to guess duration of drained data */
GstClockTime last_duration;
/* subclass provided data in processing round */
gboolean got_data;
/* subclass gave all it could already */
gboolean drained;
/* subclass currently being forcibly drained */
gboolean force;
/* need to handle changed input caps */
gboolean do_caps;
/* output bps estimatation */
/* global in samples seen */
guint64 samples_in;
/* global bytes sent out */
guint64 bytes_out;
/* context storage */
GstAudioEncoderContext ctx;
/* properties */
gint64 tolerance;
gboolean perfect_ts;
gboolean hard_resync;
gboolean granule;
gboolean hard_min;
gboolean drainable;
/* pending tags */
GstTagList *tags;
gboolean tags_changed;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
};
static GstElementClass *parent_class = NULL;
static void gst_audio_encoder_class_init (GstAudioEncoderClass * klass);
static void gst_audio_encoder_init (GstAudioEncoder * parse,
GstAudioEncoderClass * klass);
GType
gst_audio_encoder_get_type (void)
{
static GType audio_encoder_type = 0;
if (!audio_encoder_type) {
static const GTypeInfo audio_encoder_info = {
sizeof (GstAudioEncoderClass),
(GBaseInitFunc) NULL,
(GBaseFinalizeFunc) NULL,
(GClassInitFunc) gst_audio_encoder_class_init,
NULL,
NULL,
sizeof (GstAudioEncoder),
0,
(GInstanceInitFunc) gst_audio_encoder_init,
};
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAudioEncoder", &audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
g_type_add_interface_static (audio_encoder_type, GST_TYPE_PRESET,
&preset_interface_info);
}
return audio_encoder_type;
}
static void gst_audio_encoder_finalize (GObject * object);
static void gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full);
static void gst_audio_encoder_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_encoder_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_encoder_sink_activate_mode (GstPad * pad,
GstObject * parent, GstPadMode mode, gboolean active);
static GstCaps *gst_audio_encoder_getcaps_default (GstAudioEncoder * enc,
GstCaps * filter);
static gboolean gst_audio_encoder_sink_event_default (GstAudioEncoder * enc,
GstEvent * event);
static gboolean gst_audio_encoder_src_event_default (GstAudioEncoder * enc,
GstEvent * event);
static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_encoder_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc,
GstCaps * caps);
static GstFlowReturn gst_audio_encoder_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static gboolean gst_audio_encoder_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_audio_encoder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstStateChangeReturn gst_audio_encoder_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_audio_encoder_decide_allocation_default (GstAudioEncoder *
enc, GstQuery * query);
static gboolean gst_audio_encoder_propose_allocation_default (GstAudioEncoder *
enc, GstQuery * query);
static gboolean gst_audio_encoder_negotiate_default (GstAudioEncoder * enc);
static void
gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
"audio encoder base class");
g_type_class_add_private (klass, sizeof (GstAudioEncoderPrivate));
gobject_class->set_property = gst_audio_encoder_set_property;
gobject_class->get_property = gst_audio_encoder_get_property;
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_encoder_finalize);
/* properties */
g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
g_param_spec_boolean ("perfect-timestamp", "Perfect Timestamps",
"Favour perfect timestamps over tracking upstream timestamps",
DEFAULT_PERFECT_TS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_GRANULE,
g_param_spec_boolean ("mark-granule", "Granule Marking",
"Apply granule semantics to buffer metadata (implies perfect-timestamp)",
DEFAULT_GRANULE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_HARD_RESYNC,
g_param_spec_boolean ("hard-resync", "Hard Resync",
"Perform clipping and sample flushing upon discontinuity",
DEFAULT_HARD_RESYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TOLERANCE,
g_param_spec_int64 ("tolerance", "Tolerance",
"Consider discontinuity if timestamp jitter/imperfection exceeds tolerance (ns)",
0, G_MAXINT64, DEFAULT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_audio_encoder_change_state);
klass->getcaps = gst_audio_encoder_getcaps_default;
klass->sink_event = gst_audio_encoder_sink_event_default;
klass->src_event = gst_audio_encoder_src_event_default;
klass->propose_allocation = gst_audio_encoder_propose_allocation_default;
klass->decide_allocation = gst_audio_encoder_decide_allocation_default;
klass->negotiate = gst_audio_encoder_negotiate_default;
}
static void
gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
{
GstPadTemplate *pad_template;
GST_DEBUG_OBJECT (enc, "gst_audio_encoder_init");
enc->priv = GST_AUDIO_ENCODER_GET_PRIVATE (enc);
/* only push mode supported */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
g_return_if_fail (pad_template != NULL);
enc->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_event_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_event));
gst_pad_set_query_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_query));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_chain));
gst_pad_set_activatemode_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_activate_mode));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
GST_DEBUG_OBJECT (enc, "sinkpad created");
/* and we don't mind upstream traveling stuff that much ... */
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
g_return_if_fail (pad_template != NULL);
enc->srcpad = gst_pad_new_from_template (pad_template, "src");
gst_pad_set_event_function (enc->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_src_event));
gst_pad_set_query_function (enc->srcpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_src_query));
gst_pad_use_fixed_caps (enc->srcpad);
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
GST_DEBUG_OBJECT (enc, "src created");
enc->priv->adapter = gst_adapter_new ();
g_rec_mutex_init (&enc->stream_lock);
/* property default */
enc->priv->granule = DEFAULT_GRANULE;
enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
enc->priv->tolerance = DEFAULT_TOLERANCE;
enc->priv->hard_min = DEFAULT_HARD_MIN;
enc->priv->drainable = DEFAULT_DRAINABLE;
/* init state */
gst_audio_encoder_reset (enc, TRUE);
GST_DEBUG_OBJECT (enc, "init ok");
}
static void
gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
{
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_LOG_OBJECT (enc, "reset full %d", full);
if (full) {
enc->priv->active = FALSE;
enc->priv->samples_in = 0;
enc->priv->bytes_out = 0;
g_list_foreach (enc->priv->ctx.headers, (GFunc) gst_buffer_unref, NULL);
g_list_free (enc->priv->ctx.headers);
enc->priv->ctx.headers = NULL;
enc->priv->ctx.new_headers = FALSE;
gst_caps_replace (&enc->priv->ctx.caps, NULL);
memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
gst_audio_info_init (&enc->priv->ctx.info);
if (enc->priv->tags)
gst_tag_list_unref (enc->priv->tags);
enc->priv->tags = NULL;
enc->priv->tags_changed = FALSE;
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (enc->priv->pending_events);
enc->priv->pending_events = NULL;
if (enc->priv->ctx.allocator)
gst_object_unref (enc->priv->ctx.allocator);
enc->priv->ctx.allocator = NULL;
}
gst_segment_init (&enc->input_segment, GST_FORMAT_TIME);
gst_segment_init (&enc->output_segment, GST_FORMAT_TIME);
gst_adapter_clear (enc->priv->adapter);
enc->priv->got_data = FALSE;
enc->priv->drained = TRUE;
enc->priv->offset = 0;
enc->priv->base_ts = GST_CLOCK_TIME_NONE;
enc->priv->base_gp = -1;
enc->priv->samples = 0;
enc->priv->discont = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
static void
gst_audio_encoder_finalize (GObject * object)
{
GstAudioEncoder *enc = GST_AUDIO_ENCODER (object);
g_object_unref (enc->priv->adapter);
g_rec_mutex_clear (&enc->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_audio_encoder_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstAudioEncoder *enc = GST_AUDIO_ENCODER (element);
GstAudioEncoderClass *klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (klass->open) {
if (!klass->open (enc))
goto open_failed;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
if (klass->close) {
if (!klass->close (enc))
goto close_failed;
}
default:
break;
}
return ret;
open_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("Failed to open codec"));
return GST_STATE_CHANGE_FAILURE;
}
close_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("Failed to close codec"));
return GST_STATE_CHANGE_FAILURE;
}
}
static gboolean
gst_audio_encoder_push_event (GstAudioEncoder * enc, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:{
GstSegment seg;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_event_copy_segment (event, &seg);
GST_DEBUG_OBJECT (enc, "starting segment %" GST_SEGMENT_FORMAT, &seg);
enc->output_segment = seg;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
default:
break;
}
return gst_pad_push_event (enc->srcpad, event);
}
/**
* gst_audio_encoder_finish_frame:
* @enc: a #GstAudioEncoder
* @buffer: encoded data
* @samples: number of samples (per channel) represented by encoded data
*
* Collects encoded data and pushes encoded data downstream.
* Source pad caps must be set when this is called.
*
* If @samples < 0, then best estimate is all samples provided to encoder
* (subclass) so far. @buf may be NULL, in which case next number of @samples
* are considered discarded, e.g. as a result of discontinuous transmission,
* and a discontinuity is marked.
*
* Note that samples received in gst_audio_encoder_handle_frame()
* may be invalidated by a call to this function.
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*/
GstFlowReturn
gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
gint samples)
{
GstAudioEncoderClass *klass;
GstAudioEncoderPrivate *priv;
GstAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
priv = enc->priv;
ctx = &enc->priv->ctx;
/* subclass should not hand us no data */
g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
GST_FLOW_ERROR);
/* subclass should know what it is producing by now */
if (!ctx->caps)
goto no_caps;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_LOG_OBJECT (enc,
"accepting %" G_GSIZE_FORMAT " bytes encoded data as %d samples",
buf ? gst_buffer_get_size (buf) : -1, samples);
if (G_UNLIKELY (ctx->output_caps_changed
|| gst_pad_check_reconfigure (enc->srcpad))) {
if (!gst_audio_encoder_negotiate (enc)) {
ret = GST_FLOW_NOT_NEGOTIATED;
goto exit;
}
}
/* mark subclass still alive and providing */
if (G_LIKELY (buf))
priv->got_data = TRUE;
if (priv->pending_events) {
GList *pending_events, *l;
pending_events = priv->pending_events;
priv->pending_events = NULL;
GST_DEBUG_OBJECT (enc, "Pushing pending events");
for (l = pending_events; l; l = l->next)
gst_audio_encoder_push_event (enc, l->data);
g_list_free (pending_events);
}
/* send after pending events, which likely includes newsegment event */
if (G_UNLIKELY (enc->priv->tags && enc->priv->tags_changed)) {
#if 0
GstCaps *caps;
#endif
/* add codec info to pending tags */
#if 0
if (!enc->priv->tags)
enc->priv->tags = gst_tag_list_new ();
enc->priv->tags = gst_tag_list_make_writable (enc->priv->tags);
caps = gst_pad_get_current_caps (enc->srcpad);
gst_pb_utils_add_codec_description_to_tag_list (enc->priv->tags,
GST_TAG_CODEC, caps);
gst_pb_utils_add_codec_description_to_tag_list (enc->priv->tags,
GST_TAG_AUDIO_CODEC, caps);
#endif
GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, enc->priv->tags);
gst_audio_encoder_push_event (enc,
gst_event_new_tag (gst_tag_list_ref (enc->priv->tags)));
enc->priv->tags_changed = FALSE;
}
/* remove corresponding samples from input */
if (samples < 0)
samples = (enc->priv->offset / ctx->info.bpf);
if (G_LIKELY (samples)) {
/* track upstream ts if so configured */
if (!enc->priv->perfect_ts) {
guint64 ts, distance;
ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
g_assert (distance % ctx->info.bpf == 0);
distance /= ctx->info.bpf;
GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past prev_ts %"
GST_TIME_FORMAT, distance, GST_TIME_ARGS (ts));
GST_LOG_OBJECT (enc, "%" G_GUINT64_FORMAT " samples past base_ts %"
GST_TIME_FORMAT, priv->samples, GST_TIME_ARGS (priv->base_ts));
/* when draining adapter might be empty and no ts to offer */
if (GST_CLOCK_TIME_IS_VALID (ts) && ts != priv->base_ts) {
GstClockTimeDiff diff;
GstClockTime old_ts, next_ts;
/* passed into another buffer;
* mild check for discontinuity and only mark if so */
next_ts = ts +
gst_util_uint64_scale (distance, GST_SECOND, ctx->info.rate);
old_ts = priv->base_ts +
gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->info.rate);
diff = GST_CLOCK_DIFF (next_ts, old_ts);
GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
/* only mark discontinuity if beyond tolerance */
if (G_UNLIKELY (diff < -enc->priv->tolerance ||
diff > enc->priv->tolerance)) {
GST_DEBUG_OBJECT (enc, "marked discont");
priv->discont = TRUE;
}
if (diff > GST_SECOND / ctx->info.rate / 2 ||
diff < -GST_SECOND / ctx->info.rate / 2) {
GST_LOG_OBJECT (enc, "new upstream ts %" GST_TIME_FORMAT
" at distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (ts), distance);
/* re-sync to upstream ts */
priv->base_ts = ts;
priv->samples = distance;
} else {
GST_LOG_OBJECT (enc, "new upstream ts only introduces jitter");
}
}
}
/* advance sample view */
if (G_UNLIKELY (samples * ctx->info.bpf > priv->offset)) {
if (G_LIKELY (!priv->force)) {
/* no way we can let this pass */
g_assert_not_reached ();
/* really no way */
goto overflow;
} else {
priv->offset = 0;
if (samples * ctx->info.bpf >= gst_adapter_available (priv->adapter))
gst_adapter_clear (priv->adapter);
else
gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
}
} else {
gst_adapter_flush (priv->adapter, samples * ctx->info.bpf);
priv->offset -= samples * ctx->info.bpf;
/* avoid subsequent stray prev_ts */
if (G_UNLIKELY (gst_adapter_available (priv->adapter) == 0))
gst_adapter_clear (priv->adapter);
}
/* sample count advanced below after buffer handling */
}
/* collect output */
if (G_LIKELY (buf)) {
gsize size;
/* Pushing headers first */
if (G_UNLIKELY (priv->ctx.new_headers)) {
GList *tmp;
GST_DEBUG_OBJECT (enc, "Sending headers");
for (tmp = priv->ctx.headers; tmp; tmp = tmp->next) {
GstBuffer *tmpbuf = gst_buffer_ref (tmp->data);
tmpbuf = gst_buffer_make_writable (tmpbuf);
size = gst_buffer_get_size (tmpbuf);
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (enc, "marking discont");
GST_BUFFER_FLAG_SET (tmpbuf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* Ogg codecs like Vorbis use offset/offset-end in a special
* way and both should be 0 for these codecs */
if (priv->base_gp >= 0) {
GST_BUFFER_OFFSET (tmpbuf) = 0;
GST_BUFFER_OFFSET_END (tmpbuf) = 0;
} else {
GST_BUFFER_OFFSET (tmpbuf) = priv->bytes_out;
GST_BUFFER_OFFSET_END (tmpbuf) = priv->bytes_out + size;
}
priv->bytes_out += size;
gst_pad_push (enc->srcpad, tmpbuf);
}
priv->ctx.new_headers = FALSE;
}
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (enc, "taking %" G_GSIZE_FORMAT " bytes for output", size);
buf = gst_buffer_make_writable (buf);
/* decorate */
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
/* FIXME ? lookahead could lead to weird ts and duration ?
* (particularly if not in perfect mode) */
/* mind sample rounding and produce perfect output */
GST_BUFFER_TIMESTAMP (buf) = priv->base_ts +
gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
ctx->info.rate);
GST_DEBUG_OBJECT (enc, "out samples %d", samples);
if (G_LIKELY (samples > 0)) {
priv->samples += samples;
GST_BUFFER_DURATION (buf) = priv->base_ts +
gst_util_uint64_scale (priv->samples - ctx->lookahead, GST_SECOND,
ctx->info.rate) - GST_BUFFER_TIMESTAMP (buf);
priv->last_duration = GST_BUFFER_DURATION (buf);
} else {
/* duration forecast in case of handling remainder;
* the last one is probably like the previous one ... */
GST_BUFFER_DURATION (buf) = priv->last_duration;
}
if (priv->base_gp >= 0) {
/* pamper oggmux */
/* FIXME: in longer run, muxer should take care of this ... */
/* offset_end = granulepos for ogg muxer */
GST_BUFFER_OFFSET_END (buf) = priv->base_gp + priv->samples -
enc->priv->ctx.lookahead;
/* offset = timestamp corresponding to granulepos for ogg muxer */
GST_BUFFER_OFFSET (buf) =
GST_FRAMES_TO_CLOCK_TIME (GST_BUFFER_OFFSET_END (buf),
ctx->info.rate);
} else {
GST_BUFFER_OFFSET (buf) = priv->bytes_out;
GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + size;
}
}
priv->bytes_out += size;
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (enc, "marking discont");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
if (klass->pre_push) {
/* last chance for subclass to do some dirty stuff */
ret = klass->pre_push (enc, &buf);
if (ret != GST_FLOW_OK || !buf) {
GST_DEBUG_OBJECT (enc, "subclass returned %s, buf %p",
gst_flow_get_name (ret), buf);
if (buf)
gst_buffer_unref (buf);
goto exit;
}
}
GST_LOG_OBJECT (enc,
"pushing buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
ret = gst_pad_push (enc->srcpad, buf);
GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
} else {
/* merely advance samples, most work for that already done above */
priv->samples += samples;
}
exit:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
no_caps:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, ("no caps set"), (NULL));
if (buf)
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
overflow:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
("received more encoded samples %d than provided %d",
samples, priv->offset / ctx->info.bpf), (NULL));
if (buf)
gst_buffer_unref (buf);
ret = GST_FLOW_ERROR;
goto exit;
}
}
/* adapter tracking idea:
* - start of adapter corresponds with what has already been encoded
* (i.e. really returned by encoder subclass)
* - start + offset is what needs to be fed to subclass next */
static GstFlowReturn
gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
{
GstAudioEncoderClass *klass;
GstAudioEncoderPrivate *priv;
GstAudioEncoderContext *ctx;
gint av, need;
GstBuffer *buf;
GstFlowReturn ret = GST_FLOW_OK;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
priv = enc->priv;
ctx = &enc->priv->ctx;
while (ret == GST_FLOW_OK) {
buf = NULL;
av = gst_adapter_available (priv->adapter);
g_assert (priv->offset <= av);
av -= priv->offset;
need =
ctx->frame_samples_min >
0 ? ctx->frame_samples_min * ctx->info.bpf : av;
GST_LOG_OBJECT (enc, "available: %d, needed: %d, force: %d", av, need,
force);
if ((need > av) || !av) {
if (G_UNLIKELY (force)) {
priv->force = TRUE;
need = av;
} else {
break;
}
} else {
priv->force = FALSE;
}
if (ctx->frame_samples_max > 0)
need = MIN (av, ctx->frame_samples_max * ctx->info.bpf);
if (ctx->frame_samples_min == ctx->frame_samples_max) {
/* if we have some extra metadata,
* provide for integer multiple of frames to allow for better granularity
* of processing */
if (ctx->frame_samples_min > 0 && need) {
if (ctx->frame_max > 1)
need = need * MIN ((av / need), ctx->frame_max);
else if (ctx->frame_max == 0)
need = need * (av / need);
}
}
priv->got_data = FALSE;
if (G_LIKELY (need)) {
const guint8 *data;
data = gst_adapter_map (priv->adapter, priv->offset + need);
buf =
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
(gpointer) data, priv->offset + need, priv->offset, need, NULL, NULL);
} else if (!priv->drainable) {
GST_DEBUG_OBJECT (enc, "non-drainable and no more data");
goto finish;
}
GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
need, priv->offset);
/* mark this already as consumed,
* which it should be when subclass gives us data in exchange for samples */
priv->offset += need;
priv->samples_in += need / ctx->info.bpf;
/* subclass might not want to be bothered with leftover data,
* so take care of that here if so, otherwise pass along */
if (G_UNLIKELY (priv->force && priv->hard_min && buf)) {
GST_DEBUG_OBJECT (enc, "bypassing subclass with leftover");
ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
} else {
ret = klass->handle_frame (enc, buf);
}
if (G_LIKELY (buf)) {
gst_buffer_unref (buf);
gst_adapter_unmap (priv->adapter);
}
finish:
/* no data to feed, no leftover provided, then bail out */
if (G_UNLIKELY (!buf && !priv->got_data)) {
priv->drained = TRUE;
GST_LOG_OBJECT (enc, "no more data drained from subclass");
break;
}
}
return ret;
}
static GstFlowReturn
gst_audio_encoder_drain (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (enc, "draining");
if (enc->priv->drained)
return GST_FLOW_OK;
else {
GST_DEBUG_OBJECT (enc, "... really");
return gst_audio_encoder_push_buffers (enc, TRUE);
}
}
static void
gst_audio_encoder_set_base_gp (GstAudioEncoder * enc)
{
GstClockTime ts;
if (!enc->priv->granule)
return;
/* use running time for granule */
/* incoming data is clipped, so a valid input should yield a valid output */
ts = gst_segment_to_running_time (&enc->input_segment, GST_FORMAT_TIME,
enc->priv->base_ts);
if (GST_CLOCK_TIME_IS_VALID (ts)) {
enc->priv->base_gp =
GST_CLOCK_TIME_TO_FRAMES (enc->priv->base_ts, enc->priv->ctx.info.rate);
GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT, enc->priv->base_gp);
} else {
/* should reasonably have a valid base,
* otherwise start at 0 if we did not already start there earlier */
if (enc->priv->base_gp < 0) {
enc->priv->base_gp = 0;
GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
enc->priv->base_gp);
}
}
}
static GstFlowReturn
gst_audio_encoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstAudioEncoder *enc;
GstAudioEncoderPrivate *priv;
GstAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
gboolean discont;
gsize size;
enc = GST_AUDIO_ENCODER (parent);
priv = enc->priv;
ctx = &enc->priv->ctx;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (G_UNLIKELY (priv->do_caps)) {
GstCaps *caps = gst_pad_get_current_caps (enc->sinkpad);
if (!caps)
goto not_negotiated;
if (!gst_audio_encoder_sink_setcaps (enc, caps)) {
gst_caps_unref (caps);
goto not_negotiated;
}
gst_caps_unref (caps);
priv->do_caps = FALSE;
}
/* should know what is coming by now */
if (!ctx->info.bpf)
goto not_negotiated;
size = gst_buffer_get_size (buffer);
GST_LOG_OBJECT (enc,
"received buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
/* input shoud be whole number of sample frames */
if (size % ctx->info.bpf)
goto wrong_buffer;
#ifndef GST_DISABLE_GST_DEBUG
{
GstClockTime duration;
GstClockTimeDiff diff;
/* verify buffer duration */
duration = gst_util_uint64_scale (size, GST_SECOND,
ctx->info.rate * ctx->info.bpf);
diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
(diff > GST_SECOND / ctx->info.rate / 2 ||
diff < -GST_SECOND / ctx->info.rate / 2)) {
GST_DEBUG_OBJECT (enc, "incoming buffer had incorrect duration %"
GST_TIME_FORMAT ", expected duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
GST_TIME_ARGS (duration));
}
}
#endif
discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
if (G_UNLIKELY (discont)) {
GST_LOG_OBJECT (buffer, "marked discont");
enc->priv->discont = discont;
}
/* clip to segment */
/* NOTE: slightly painful linking -laudio only for this one ... */
buffer = gst_audio_buffer_clip (buffer, &enc->input_segment, ctx->info.rate,
ctx->info.bpf);
if (G_UNLIKELY (!buffer)) {
GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
goto done;
}
size = gst_buffer_get_size (buffer);
GST_LOG_OBJECT (enc,
"buffer after segment clipping has size %" G_GSIZE_FORMAT " with ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
if (!GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
priv->base_ts = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (enc, "new base ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->base_ts));
gst_audio_encoder_set_base_gp (enc);
}
/* check for continuity;
* checked elsewhere in non-perfect case */
if (enc->priv->perfect_ts) {
GstClockTimeDiff diff = 0;
GstClockTime next_ts = 0;
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer) &&
GST_CLOCK_TIME_IS_VALID (priv->base_ts)) {
guint64 samples;
samples = priv->samples +
gst_adapter_available (priv->adapter) / ctx->info.bpf;
next_ts = priv->base_ts +
gst_util_uint64_scale (samples, GST_SECOND, ctx->info.rate);
GST_LOG_OBJECT (enc, "buffer is %" G_GUINT64_FORMAT
" samples past base_ts %" GST_TIME_FORMAT
", expected ts %" GST_TIME_FORMAT, samples,
GST_TIME_ARGS (priv->base_ts), GST_TIME_ARGS (next_ts));
diff = GST_CLOCK_DIFF (next_ts, GST_BUFFER_TIMESTAMP (buffer));
GST_LOG_OBJECT (enc, "ts diff %d ms", (gint) (diff / GST_MSECOND));
/* if within tolerance,
* discard buffer ts and carry on producing perfect stream,
* otherwise clip or resync to ts */
if (G_UNLIKELY (diff < -enc->priv->tolerance ||
diff > enc->priv->tolerance)) {
GST_DEBUG_OBJECT (enc, "marked discont");
discont = TRUE;
}
}
/* do some fancy tweaking in hard resync case */
if (discont && enc->priv->hard_resync) {
if (diff < 0) {
guint64 diff_bytes;
GST_WARNING_OBJECT (enc, "Buffer is older than expected ts %"
GST_TIME_FORMAT ". Clipping buffer", GST_TIME_ARGS (next_ts));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
if (diff_bytes >= size) {
gst_buffer_unref (buffer);
goto done;
}
buffer = gst_buffer_make_writable (buffer);
gst_buffer_resize (buffer, diff_bytes, size - diff_bytes);
GST_BUFFER_TIMESTAMP (buffer) += diff;
/* care even less about duration after this */
} else {
/* drain stuff prior to resync */
gst_audio_encoder_drain (enc);
}
}
if (discont) {
/* now re-sync ts */
priv->base_ts += diff;
gst_audio_encoder_set_base_gp (enc);
priv->discont |= discont;
}
}
gst_adapter_push (enc->priv->adapter, buffer);
/* new stuff, so we can push subclass again */
enc->priv->drained = FALSE;
ret = gst_audio_encoder_push_buffers (enc, FALSE);
done:
GST_LOG_OBJECT (enc, "chain leaving");
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
("encoder not initialized"));
gst_buffer_unref (buffer);
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
wrong_buffer:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("buffer size %" G_GSIZE_FORMAT " not a multiple of %d",
gst_buffer_get_size (buffer), ctx->info.bpf));
gst_buffer_unref (buffer);
ret = GST_FLOW_ERROR;
goto done;
}
}
static gboolean
audio_info_is_equal (GstAudioInfo * from, GstAudioInfo * to)
{
if (from == to)
return TRUE;
if (from->finfo == NULL || to->finfo == NULL)
return FALSE;
if (GST_AUDIO_INFO_FORMAT (from) != GST_AUDIO_INFO_FORMAT (to))
return FALSE;
if (GST_AUDIO_INFO_RATE (from) != GST_AUDIO_INFO_RATE (to))
return FALSE;
if (GST_AUDIO_INFO_CHANNELS (from) != GST_AUDIO_INFO_CHANNELS (to))
return FALSE;
if (GST_AUDIO_INFO_CHANNELS (from) > 64)
return TRUE;
return (memcmp (from->position, to->position,
GST_AUDIO_INFO_CHANNELS (from) * sizeof (to->position[0])) == 0);
}
static gboolean
gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
{
GstAudioEncoderClass *klass;
GstAudioEncoderContext *ctx;
GstAudioInfo state;
gboolean res = TRUE, changed = FALSE;
guint old_rate;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
/* subclass must do something here ... */
g_return_val_if_fail (klass->set_format != NULL, FALSE);
ctx = &enc->priv->ctx;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
if (!gst_caps_is_fixed (caps))
goto refuse_caps;
/* adjust ts tracking to new sample rate */
old_rate = GST_AUDIO_INFO_RATE (&ctx->info);
if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
enc->priv->base_ts +=
GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
enc->priv->samples = 0;
}
if (!gst_audio_info_from_caps (&state, caps))
goto refuse_caps;
changed = !audio_info_is_equal (&state, &ctx->info);
if (changed) {
GstClockTime old_min_latency;
GstClockTime old_max_latency;
/* drain any pending old data stuff */
gst_audio_encoder_drain (enc);
/* context defaults */
enc->priv->ctx.frame_samples_min = 0;
enc->priv->ctx.frame_samples_max = 0;
enc->priv->ctx.frame_max = 0;
enc->priv->ctx.lookahead = 0;
/* element might report latency */
GST_OBJECT_LOCK (enc);
old_min_latency = ctx->min_latency;
old_max_latency = ctx->max_latency;
GST_OBJECT_UNLOCK (enc);
if (klass->set_format)
res = klass->set_format (enc, &state);
if (res)
ctx->info = state;
/* invalidate state to ensure no casual carrying on */
if (!res) {
GST_DEBUG_OBJECT (enc, "subclass did not accept format");
gst_audio_info_init (&state);
goto exit;
}
/* notify if new latency */
GST_OBJECT_LOCK (enc);
if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
(ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
GST_OBJECT_UNLOCK (enc);
/* post latency message on the bus */
gst_element_post_message (GST_ELEMENT (enc),
gst_message_new_latency (GST_OBJECT (enc)));
GST_OBJECT_LOCK (enc);
}
GST_OBJECT_UNLOCK (enc);
} else {
GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
}
exit:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
goto exit;
}
}
/**
* gst_audio_encoder_proxy_getcaps:
* @enc: a #GstAudioEncoder
* @caps: initial caps
* @filter: filter caps
*
* Returns caps that express @caps (or sink template caps if @caps == NULL)
* restricted to channel/rate combinations supported by downstream elements
* (e.g. muxers).
*
* Returns: a #GstCaps owned by caller
*/
GstCaps *
gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps,
GstCaps * filter)
{
GstCaps *templ_caps = NULL;
GstCaps *allowed = NULL;
GstCaps *fcaps, *filter_caps;
gint i, j;
/* we want to be able to communicate to upstream elements like audioconvert
* and audioresample any rate/channel restrictions downstream (e.g. muxer
* only accepting certain sample rates) */
templ_caps =
caps ? gst_caps_ref (caps) : gst_pad_get_pad_template_caps (enc->sinkpad);
allowed = gst_pad_get_allowed_caps (enc->srcpad);
if (!allowed || gst_caps_is_empty (allowed) || gst_caps_is_any (allowed)) {
fcaps = templ_caps;
goto done;
}
GST_LOG_OBJECT (enc, "template caps %" GST_PTR_FORMAT, templ_caps);
GST_LOG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed);
filter_caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (templ_caps); i++) {
GQuark q_name;
q_name = gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
/* pick rate + channel fields from allowed caps */
for (j = 0; j < gst_caps_get_size (allowed); j++) {
const GstStructure *allowed_s = gst_caps_get_structure (allowed, j);
const GValue *val;
GstStructure *s;
s = gst_structure_new_id_empty (q_name);
if ((val = gst_structure_get_value (allowed_s, "rate")))
gst_structure_set_value (s, "rate", val);
if ((val = gst_structure_get_value (allowed_s, "channels")))
gst_structure_set_value (s, "channels", val);
/* following might also make sense for some encoded formats,
* e.g. wavpack */
if ((val = gst_structure_get_value (allowed_s, "channel-mask")))
gst_structure_set_value (s, "channel-mask", val);
filter_caps = gst_caps_merge_structure (filter_caps, s);
}
}
fcaps = gst_caps_intersect (filter_caps, templ_caps);
gst_caps_unref (filter_caps);
gst_caps_unref (templ_caps);
if (filter) {
GST_LOG_OBJECT (enc, "intersecting with %" GST_PTR_FORMAT, filter);
filter_caps = gst_caps_intersect_full (filter, fcaps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (fcaps);
fcaps = filter_caps;
}
done:
gst_caps_replace (&allowed, NULL);
GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
return fcaps;
}
static GstCaps *
gst_audio_encoder_getcaps_default (GstAudioEncoder * enc, GstCaps * filter)
{
GstCaps *caps;
caps = gst_audio_encoder_proxy_getcaps (enc, NULL, filter);
GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
{
GstAudioEncoderClass *klass;
gboolean res;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
GstSegment seg;
gst_event_copy_segment (event, &seg);
if (seg.format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (enc, "received TIME SEGMENT %" GST_SEGMENT_FORMAT,
&seg);
} else {
GST_DEBUG_OBJECT (enc, "received SEGMENT %" GST_SEGMENT_FORMAT, &seg);
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
res = TRUE;
break;
}
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* finish current segment */
gst_audio_encoder_drain (enc);
/* reset partially for new segment */
gst_audio_encoder_reset (enc, FALSE);
/* and follow along with segment */
enc->input_segment = seg;
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = TRUE;
break;
}
case GST_EVENT_FLUSH_START:
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_FLUSH_STOP:
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* discard any pending stuff */
/* TODO route through drain ?? */
if (!enc->priv->drained && klass->flush)
klass->flush (enc);
/* and get (re)set for the sequel */
gst_audio_encoder_reset (enc, FALSE);
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (enc->priv->pending_events);
enc->priv->pending_events = NULL;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_EOS:
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_audio_encoder_drain (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
/* forward immediately because no buffer or serialized event
* will come after EOS and nothing could trigger another
* _finish_frame() call. */
res = gst_audio_encoder_push_event (enc, event);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
enc->priv->do_caps = TRUE;
res = TRUE;
gst_event_unref (event);
break;
}
case GST_EVENT_TAG:
{
GstTagList *tags;
gst_event_parse_tag (event, &tags);
if (gst_tag_list_get_scope (tags) == GST_TAG_SCOPE_STREAM) {
tags = gst_tag_list_copy (tags);
/* FIXME: make generic based on GST_TAG_FLAG_ENCODED */
gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_VIDEO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_SUBTITLE_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
gst_tag_list_remove_tag (tags, GST_TAG_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_NOMINAL_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MAXIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MINIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER_VERSION);
gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (tags);
gst_event_unref (event);
event = NULL;
res = TRUE;
break;
}
/* fall through */
}
default:
/* Forward non-serialized events immediately. */
if (!GST_EVENT_IS_SERIALIZED (event)) {
res =
gst_pad_event_default (enc->sinkpad, GST_OBJECT_CAST (enc), event);
} else {
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = TRUE;
}
break;
}
return res;
}
static gboolean
gst_audio_encoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstAudioEncoder *enc;
GstAudioEncoderClass *klass;
gboolean ret;
enc = GST_AUDIO_ENCODER (parent);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
if (klass->sink_event)
ret = klass->sink_event (enc, event);
else {
gst_event_unref (event);
ret = FALSE;
}
GST_DEBUG_OBJECT (enc, "event result %d", ret);
return ret;
}
static gboolean
gst_audio_encoder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
GstAudioEncoder *enc;
enc = GST_AUDIO_ENCODER (parent);
switch (GST_QUERY_TYPE (query)) {
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 (&enc->priv->ctx.info,
src_fmt, src_val, dest_fmt, &dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
res = TRUE;
break;
}
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
GstAudioEncoderClass *klass;
gst_query_parse_caps (query, &filter);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
if (klass->getcaps) {
caps = klass->getcaps (enc, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
}
break;
}
case GST_QUERY_ALLOCATION:
{
GstAudioEncoderClass *klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
if (klass->propose_allocation)
res = klass->propose_allocation (enc, query);
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
error:
return res;
}
static gboolean
gst_audio_encoder_src_event_default (GstAudioEncoder * enc, GstEvent * event)
{
gboolean res;
switch (GST_EVENT_TYPE (event)) {
default:
res = gst_pad_event_default (enc->srcpad, GST_OBJECT_CAST (enc), event);
break;
}
return res;
}
static gboolean
gst_audio_encoder_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAudioEncoder *enc;
GstAudioEncoderClass *klass;
gboolean ret;
enc = GST_AUDIO_ENCODER (parent);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
if (klass->src_event)
ret = klass->src_event (enc, event);
else {
gst_event_unref (event);
ret = FALSE;
}
return ret;
}
static gboolean
gst_audio_encoder_decide_allocation_default (GstAudioEncoder * enc,
GstQuery * query)
{
GstAllocator *allocator = NULL;
GstAllocationParams params;
gboolean update_allocator;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
/* try the allocator */
gst_query_parse_nth_allocation_param (query, 0, &allocator, &params);
update_allocator = TRUE;
} else {
allocator = NULL;
gst_allocation_params_init (&params);
update_allocator = FALSE;
}
if (update_allocator)
gst_query_set_nth_allocation_param (query, 0, allocator, &params);
else
gst_query_add_allocation_param (query, allocator, &params);
if (allocator)
gst_object_unref (allocator);
return TRUE;
}
static gboolean
gst_audio_encoder_propose_allocation_default (GstAudioEncoder * enc,
GstQuery * query)
{
return TRUE;
}
/*
* 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;
}
/* FIXME ? are any of these queries (other than latency) an encoder'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_encoder_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstAudioEncoder *enc;
gboolean res = FALSE;
enc = GST_AUDIO_ENCODER (parent);
GST_LOG_OBJECT (enc, "handling query: %" GST_PTR_FORMAT, query);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat fmt, req_fmt;
gint64 pos, val;
if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
GST_LOG_OBJECT (enc, "returning peer response");
break;
}
gst_query_parse_position (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_peer_query_position (enc->sinkpad, fmt, &pos)))
break;
if ((res =
gst_pad_peer_query_convert (enc->sinkpad, fmt, pos, req_fmt,
&val))) {
gst_query_set_position (query, req_fmt, val);
}
break;
}
case GST_QUERY_DURATION:
{
GstFormat fmt, req_fmt;
gint64 dur, val;
if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
GST_LOG_OBJECT (enc, "returning peer response");
break;
}
gst_query_parse_duration (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_peer_query_duration (enc->sinkpad, fmt, &dur)))
break;
if ((res =
gst_pad_peer_query_convert (enc->sinkpad, fmt, dur, req_fmt,
&val))) {
gst_query_set_duration (query, req_fmt, val);
}
break;
}
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 (&enc->priv->ctx.info,
enc->priv->bytes_out, enc->priv->samples_in, 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 (enc->sinkpad, query))) {
gboolean live;
GstClockTime min_latency, max_latency;
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
GST_DEBUG_OBJECT (enc, "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 (enc);
/* add our latency */
if (min_latency != -1)
min_latency += enc->priv->ctx.min_latency;
if (max_latency != -1)
max_latency += enc->priv->ctx.max_latency;
GST_OBJECT_UNLOCK (enc);
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static void
gst_audio_encoder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioEncoder *enc;
enc = GST_AUDIO_ENCODER (object);
switch (prop_id) {
case PROP_PERFECT_TS:
if (enc->priv->granule && !g_value_get_boolean (value))
GST_WARNING_OBJECT (enc, "perfect-timestamp can not be set FALSE "
"while granule handling is enabled");
else
enc->priv->perfect_ts = g_value_get_boolean (value);
break;
case PROP_HARD_RESYNC:
enc->priv->hard_resync = g_value_get_boolean (value);
break;
case PROP_TOLERANCE:
enc->priv->tolerance = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_encoder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioEncoder *enc;
enc = GST_AUDIO_ENCODER (object);
switch (prop_id) {
case PROP_PERFECT_TS:
g_value_set_boolean (value, enc->priv->perfect_ts);
break;
case PROP_GRANULE:
g_value_set_boolean (value, enc->priv->granule);
break;
case PROP_HARD_RESYNC:
g_value_set_boolean (value, enc->priv->hard_resync);
break;
case PROP_TOLERANCE:
g_value_set_int64 (value, enc->priv->tolerance);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_audio_encoder_activate (GstAudioEncoder * enc, gboolean active)
{
GstAudioEncoderClass *klass;
gboolean result = TRUE;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
g_return_val_if_fail (!enc->priv->granule || enc->priv->perfect_ts, FALSE);
GST_DEBUG_OBJECT (enc, "activate %d", active);
if (active) {
if (enc->priv->tags)
gst_tag_list_unref (enc->priv->tags);
enc->priv->tags = gst_tag_list_new_empty ();
enc->priv->tags_changed = FALSE;
if (!enc->priv->active && klass->start)
result = klass->start (enc);
} else {
/* We must make sure streaming has finished before resetting things
* and calling the ::stop vfunc */
GST_PAD_STREAM_LOCK (enc->sinkpad);
GST_PAD_STREAM_UNLOCK (enc->sinkpad);
if (enc->priv->active && klass->stop)
result = klass->stop (enc);
/* clean up */
gst_audio_encoder_reset (enc, TRUE);
}
GST_DEBUG_OBJECT (enc, "activate return: %d", result);
return result;
}
static gboolean
gst_audio_encoder_sink_activate_mode (GstPad * pad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean result = TRUE;
GstAudioEncoder *enc;
enc = GST_AUDIO_ENCODER (parent);
GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
result = gst_audio_encoder_activate (enc, active);
if (result)
enc->priv->active = active;
GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
return result;
}
/**
* gst_audio_encoder_get_audio_info:
* @enc: a #GstAudioEncoder
*
* Returns: a #GstAudioInfo describing the input audio format
*/
GstAudioInfo *
gst_audio_encoder_get_audio_info (GstAudioEncoder * enc)
{
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), NULL);
return &enc->priv->ctx.info;
}
/**
* gst_audio_encoder_set_frame_samples_min:
* @enc: a #GstAudioEncoder
* @num: number of samples per frame
*
* Sets number of samples (per channel) subclass needs to be handed,
* at least or will be handed all available if 0.
*
* If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
* must be called with the same number.
*/
void
gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
enc->priv->ctx.frame_samples_min = num;
}
/**
* gst_audio_encoder_get_frame_samples_min:
* @enc: a #GstAudioEncoder
*
* Returns: currently minimum requested samples per frame
*/
gint
gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc)
{
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
return enc->priv->ctx.frame_samples_min;
}
/**
* gst_audio_encoder_set_frame_samples_max:
* @enc: a #GstAudioEncoder
* @num: number of samples per frame
*
* Sets number of samples (per channel) subclass needs to be handed,
* at most or will be handed all available if 0.
*
* If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
* must be called with the same number.
*/
void
gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
enc->priv->ctx.frame_samples_max = num;
}
/**
* gst_audio_encoder_get_frame_samples_max:
* @enc: a #GstAudioEncoder
*
* Returns: currently maximum requested samples per frame
*/
gint
gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc)
{
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
return enc->priv->ctx.frame_samples_max;
}
/**
* gst_audio_encoder_set_frame_max:
* @enc: a #GstAudioEncoder
* @num: number of frames
*
* Sets max number of frames accepted at once (assumed minimally 1).
* Requires @frame_samples_min and @frame_samples_max to be the equal.
*/
void
gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
enc->priv->ctx.frame_max = num;
}
/**
* gst_audio_encoder_get_frame_max:
* @enc: a #GstAudioEncoder
*
* Returns: currently configured maximum handled frames
*/
gint
gst_audio_encoder_get_frame_max (GstAudioEncoder * enc)
{
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
return enc->priv->ctx.frame_max;
}
/**
* gst_audio_encoder_set_lookahead:
* @enc: a #GstAudioEncoder
* @num: lookahead
*
* Sets encoder lookahead (in units of input rate samples)
*/
void
gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
enc->priv->ctx.lookahead = num;
}
/**
* gst_audio_encoder_get_lookahead:
* @enc: a #GstAudioEncoder
*
* Returns: currently configured encoder lookahead
*/
gint
gst_audio_encoder_get_lookahead (GstAudioEncoder * enc)
{
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
return enc->priv->ctx.lookahead;
}
/**
* gst_audio_encoder_set_latency:
* @enc: a #GstAudioEncoder
* @min: minimum latency
* @max: maximum latency
*
* Sets encoder latency.
*/
void
gst_audio_encoder_set_latency (GstAudioEncoder * enc,
GstClockTime min, GstClockTime max)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_OBJECT_LOCK (enc);
enc->priv->ctx.min_latency = min;
enc->priv->ctx.max_latency = max;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_latency:
* @enc: a #GstAudioEncoder
* @min: (out) (allow-none): a pointer to storage to hold minimum latency
* @max: (out) (allow-none): a pointer to storage to hold maximum latency
*
* Sets the variables pointed to by @min and @max to the currently configured
* latency.
*/
void
gst_audio_encoder_get_latency (GstAudioEncoder * enc,
GstClockTime * min, GstClockTime * max)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_OBJECT_LOCK (enc);
if (min)
*min = enc->priv->ctx.min_latency;
if (max)
*max = enc->priv->ctx.max_latency;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_set_headers:
* @enc: a #GstAudioEncoder
* @headers: (transfer full) (element-type Gst.Buffer): a list of
* #GstBuffer containing the codec header
*
* Set the codec headers to be sent downstream whenever requested.
*/
void
gst_audio_encoder_set_headers (GstAudioEncoder * enc, GList * headers)
{
GST_DEBUG_OBJECT (enc, "new headers %p", headers);
if (enc->priv->ctx.headers) {
g_list_foreach (enc->priv->ctx.headers, (GFunc) gst_buffer_unref, NULL);
g_list_free (enc->priv->ctx.headers);
}
enc->priv->ctx.headers = headers;
enc->priv->ctx.new_headers = TRUE;
}
/**
* gst_audio_encoder_set_mark_granule:
* @enc: a #GstAudioEncoder
* @enabled: new state
*
* Enable or disable encoder granule handling.
*
* MT safe.
*/
void
gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_LOG_OBJECT (enc, "enabled: %d", enabled);
GST_OBJECT_LOCK (enc);
enc->priv->granule = enabled;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_mark_granule:
* @enc: a #GstAudioEncoder
*
* Queries if the encoder will handle granule marking.
*
* Returns: TRUE if granule marking is enabled.
*
* MT safe.
*/
gboolean
gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
GST_OBJECT_LOCK (enc);
result = enc->priv->granule;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_set_perfect_timestamp:
* @enc: a #GstAudioEncoder
* @enabled: new state
*
* Enable or disable encoder perfect output timestamp preference.
*
* MT safe.
*/
void
gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_LOG_OBJECT (enc, "enabled: %d", enabled);
GST_OBJECT_LOCK (enc);
enc->priv->perfect_ts = enabled;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_perfect_timestamp:
* @enc: a #GstAudioEncoder
*
* Queries encoder perfect timestamp behaviour.
*
* Returns: TRUE if perfect timestamp setting enabled.
*
* MT safe.
*/
gboolean
gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
GST_OBJECT_LOCK (enc);
result = enc->priv->perfect_ts;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_set_hard_sync:
* @enc: a #GstAudioEncoder
* @enabled: new state
*
* Sets encoder hard resync handling.
*
* MT safe.
*/
void
gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_LOG_OBJECT (enc, "enabled: %d", enabled);
GST_OBJECT_LOCK (enc);
enc->priv->hard_resync = enabled;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_hard_sync:
* @enc: a #GstAudioEncoder
*
* Queries encoder's hard resync setting.
*
* Returns: TRUE if hard resync is enabled.
*
* MT safe.
*/
gboolean
gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
GST_OBJECT_LOCK (enc);
result = enc->priv->hard_resync;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_set_tolerance:
* @enc: a #GstAudioEncoder
* @tolerance: new tolerance
*
* Configures encoder audio jitter tolerance threshold.
*
* MT safe.
*/
void
gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, GstClockTime tolerance)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_OBJECT_LOCK (enc);
enc->priv->tolerance = tolerance;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_tolerance:
* @enc: a #GstAudioEncoder
*
* Queries current audio jitter tolerance threshold.
*
* Returns: encoder audio jitter tolerance threshold.
*
* MT safe.
*/
GstClockTime
gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
{
GstClockTime result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
GST_OBJECT_LOCK (enc);
result = enc->priv->tolerance;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_set_hard_min:
* @enc: a #GstAudioEncoder
* @enabled: new state
*
* Configures encoder hard minimum handling. If enabled, subclass
* will never be handed less samples than it configured, which otherwise
* might occur near end-of-data handling. Instead, the leftover samples
* will simply be discarded.
*
* MT safe.
*/
void
gst_audio_encoder_set_hard_min (GstAudioEncoder * enc, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_OBJECT_LOCK (enc);
enc->priv->hard_min = enabled;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_hard_min:
* @enc: a #GstAudioEncoder
*
* Queries encoder hard minimum handling.
*
* Returns: TRUE if hard minimum handling is enabled.
*
* MT safe.
*/
gboolean
gst_audio_encoder_get_hard_min (GstAudioEncoder * enc)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
GST_OBJECT_LOCK (enc);
result = enc->priv->hard_min;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_set_drainable:
* @enc: a #GstAudioEncoder
* @enabled: new state
*
* Configures encoder drain handling. If drainable, subclass might
* be handed a NULL buffer to have it return any leftover encoded data.
* Otherwise, it is not considered so capable and will only ever be passed
* real data.
*
* MT safe.
*/
void
gst_audio_encoder_set_drainable (GstAudioEncoder * enc, gboolean enabled)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
GST_OBJECT_LOCK (enc);
enc->priv->drainable = enabled;
GST_OBJECT_UNLOCK (enc);
}
/**
* gst_audio_encoder_get_drainable:
* @enc: a #GstAudioEncoder
*
* Queries encoder drain handling.
*
* Returns: TRUE if drainable handling is enabled.
*
* MT safe.
*/
gboolean
gst_audio_encoder_get_drainable (GstAudioEncoder * enc)
{
gboolean result;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
GST_OBJECT_LOCK (enc);
result = enc->priv->drainable;
GST_OBJECT_UNLOCK (enc);
return result;
}
/**
* gst_audio_encoder_merge_tags:
* @enc: a #GstAudioEncoder
* @tags: a #GstTagList to merge
* @mode: the #GstTagMergeMode to use
*
* Adds tags to so-called pending tags, which will be processed
* before pushing out data downstream.
*
* Note that this is provided for convenience, and the subclass is
* not required to use this and can still do tag handling on its own,
* although it should be aware that baseclass already takes care
* of the usual CODEC/AUDIO_CODEC tags.
*
* MT safe.
*/
void
gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
const GstTagList * tags, GstTagMergeMode mode)
{
GstTagList *otags;
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (tags)
GST_DEBUG_OBJECT (enc, "merging tags %" GST_PTR_FORMAT, tags);
otags = enc->priv->tags;
enc->priv->tags = gst_tag_list_merge (enc->priv->tags, tags, mode);
if (otags)
gst_tag_list_unref (otags);
enc->priv->tags_changed = TRUE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
static gboolean
gst_audio_encoder_negotiate_default (GstAudioEncoder * enc)
{
GstAudioEncoderClass *klass;
gboolean res = FALSE;
GstQuery *query = NULL;
GstAllocator *allocator;
GstAllocationParams params;
GstCaps *caps;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
g_return_val_if_fail (GST_IS_CAPS (enc->priv->ctx.caps), FALSE);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
caps = enc->priv->ctx.caps;
GST_DEBUG_OBJECT (enc, "Setting srcpad caps %" GST_PTR_FORMAT, caps);
res = gst_pad_set_caps (enc->srcpad, caps);
if (!res)
goto done;
enc->priv->ctx.output_caps_changed = FALSE;
query = gst_query_new_allocation (caps, TRUE);
if (!gst_pad_peer_query (enc->srcpad, query)) {
GST_DEBUG_OBJECT (enc, "didn't get downstream ALLOCATION hints");
}
g_assert (klass->decide_allocation != NULL);
res = klass->decide_allocation (enc, query);
GST_DEBUG_OBJECT (enc, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, res,
query);
if (!res)
goto no_decide_allocation;
/* we got configuration from our peer or the decide_allocation method,
* parse them */
if (gst_query_get_n_allocation_params (query) > 0) {
gst_query_parse_nth_allocation_param (query, 0, &allocator, &params);
} else {
allocator = NULL;
gst_allocation_params_init (&params);
}
if (enc->priv->ctx.allocator)
gst_object_unref (enc->priv->ctx.allocator);
enc->priv->ctx.allocator = allocator;
enc->priv->ctx.params = params;
done:
if (query)
gst_query_unref (query);
return res;
/* ERRORS */
no_decide_allocation:
{
GST_WARNING_OBJECT (enc, "Subclass failed to decide allocation");
goto done;
}
}
/**
* gst_audio_encoder_negotiate:
* @enc: a #GstAudioEncoder
*
* Negotiate with downstreame elements to currently configured #GstCaps.
*
* Returns: #TRUE if the negotiation succeeded, else #FALSE.
*/
gboolean
gst_audio_encoder_negotiate (GstAudioEncoder * enc)
{
GstAudioEncoderClass *klass;
gboolean ret = TRUE;
g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), FALSE);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (klass->negotiate)
ret = klass->negotiate (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
}
/*
* gst_audio_encoder_set_output_format:
* @enc: a #GstAudioEncoder
* @caps: #GstCaps
*
* Configure output caps on the srcpad of @enc.
*
* Returns: %TRUE on success.
**/
gboolean
gst_audio_encoder_set_output_format (GstAudioEncoder * enc, GstCaps * caps)
{
gboolean res = TRUE;
GstCaps *templ_caps;
GST_DEBUG_OBJECT (enc, "Setting srcpad caps %" GST_PTR_FORMAT, caps);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (!gst_caps_is_fixed (caps))
goto refuse_caps;
/* Only allow caps that are a subset of the template caps */
templ_caps = gst_pad_get_pad_template_caps (enc->srcpad);
if (!gst_caps_is_subset (caps, templ_caps)) {
gst_caps_unref (templ_caps);
goto refuse_caps;
}
gst_caps_unref (templ_caps);
gst_caps_replace (&enc->priv->ctx.caps, caps);
enc->priv->ctx.output_caps_changed = TRUE;
done:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "refused caps %" GST_PTR_FORMAT, caps);
res = FALSE;
goto done;
}
}
/**
* gst_audio_encoder_allocate_output_buffer:
* @enc: a #GstAudioEncoder
* @size: size of the buffer
*
* Helper function that allocates a buffer to hold an encoded audio frame
* for @enc's current output format.
*
* Returns: (transfer full): allocated buffer
*/
GstBuffer *
gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc, gsize size)
{
GstBuffer *buffer = NULL;
g_return_val_if_fail (size > 0, NULL);
GST_DEBUG ("alloc src buffer");
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (G_UNLIKELY (enc->priv->ctx.output_caps_changed || (enc->priv->ctx.caps
&& gst_pad_check_reconfigure (enc->srcpad)))) {
if (!gst_audio_encoder_negotiate (enc))
goto done;
}
buffer =
gst_buffer_new_allocate (enc->priv->ctx.allocator, size,
&enc->priv->ctx.params);
done:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return buffer;
}
/**
* gst_audio_encoder_get_allocator:
* @enc: a #GstAudioEncoder
* @allocator: (out) (allow-none) (transfer full): the #GstAllocator
* used
* @params: (out) (allow-none) (transfer full): the
* #GstAllocatorParams of @allocator
*
* Lets #GstAudioEncoder sub-classes to know the memory @allocator
* used by the base class and its @params.
*
* Unref the @allocator after use it.
*/
void
gst_audio_encoder_get_allocator (GstAudioEncoder * enc,
GstAllocator ** allocator, GstAllocationParams * params)
{
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
if (allocator)
*allocator = enc->priv->ctx.allocator ?
gst_object_ref (enc->priv->ctx.allocator) : NULL;
if (params)
*params = enc->priv->ctx.params;
}
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