mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-13 04:41:16 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/omx/gstbaseaudioencoder.c
Sebastian Dröge d42390efd9 baseaudioencoder: Add support for requesting a minimum and maximum number of samples per frame 
This extends the special case of a fixed number of samples per frame
that was supported before already.
2011-08-17 14:28:44 +02:00

1507 lines
48 KiB
C
Raw Blame History

/* 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:gstbaseaudioencoder
* @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.
*
* GstBaseAudioEncoder and subclass should cooperate as follows.
* <orderedlist>
* <listitem>
* <itemizedlist><title>Configuration</title>
* <listitem><para>
* Initially, GstBaseAudioEncoder calls @start when the encoder element
* is activated, which allows subclass to perform any global setup.
* </para></listitem>
* <listitem><para>
* GstBaseAudioEncoder 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 context 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>
* GstBaseAudioEncoder calls @stop at end of all processing.
* </para></listitem>
* </itemizedlist>
* </listitem>
* As of configuration stage, and throughout processing, GstBaseAudioEncoder
* provides a GstBaseAudioEncoderContext that provides required context,
* e.g. describing the format of input audio data.
* 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 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_base_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 GstBaseAudioEncoderClass 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>
* GstBaseAudioEncoder 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_base_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 #GstBaseAudioEncoder:perfect-ts.
* 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 #GstBaseAudioEncoder: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-ts 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-ts case, this one optionally
* (see #GstBaseAudioEncoder: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_base_audio_encoder_finish_frame.
* </para></listitem>
* </itemizedlist>
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstbaseaudioencoder.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
#include <stdlib.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_encoder_debug);
#define GST_CAT_DEFAULT gst_base_audio_encoder_debug
#define GST_BASE_AUDIO_ENCODER_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_ENCODER, \
GstBaseAudioEncoderPrivate))
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
struct _GstBaseAudioEncoderPrivate
{
/* 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;
/* output bps estimatation */
/* global in samples seen */
guint64 samples_in;
/* global bytes sent out */
guint64 bytes_out;
/* context storage */
GstBaseAudioEncoderContext ctx;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
};
static void
do_init (GType gtype)
{
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
g_type_add_interface_static (gtype, GST_TYPE_PRESET, &preset_interface_info);
}
GST_BOILERPLATE_FULL (GstBaseAudioEncoder, gst_base_audio_encoder, GstElement,
GST_TYPE_ELEMENT, do_init);
static void gst_base_audio_encoder_finalize (GObject * object);
static void gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc,
gboolean full);
static void gst_base_audio_encoder_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_base_audio_encoder_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_base_audio_encoder_sink_activate_push (GstPad * pad,
gboolean active);
static gboolean gst_base_audio_encoder_sink_event (GstPad * pad,
GstEvent * event);
static gboolean gst_base_audio_encoder_sink_setcaps (GstPad * pad,
GstCaps * caps);
static GstFlowReturn gst_base_audio_encoder_chain (GstPad * pad,
GstBuffer * buffer);
static gboolean gst_base_audio_encoder_src_query (GstPad * pad,
GstQuery * query);
static gboolean gst_base_audio_encoder_sink_query (GstPad * pad,
GstQuery * query);
static const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad *
pad);
static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad);
static void
gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_base_audio_encoder_debug, "baseaudioencoder", 0,
"baseaudioencoder element");
g_type_class_add_private (klass, sizeof (GstBaseAudioEncoderPrivate));
gobject_class->set_property = gst_base_audio_encoder_set_property;
gobject_class->get_property = gst_base_audio_encoder_get_property;
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_audio_encoder_finalize);
/* properties */
g_object_class_install_property (gobject_class, PROP_PERFECT_TS,
g_param_spec_boolean ("perfect-ts", "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 ("granule", "Granule Marking",
"Apply granule semantics to buffer metadata (implies perfect-ts)",
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));
}
static void
gst_base_audio_encoder_base_init (gpointer g_class)
{
}
static void
gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
GstBaseAudioEncoderClass * bclass)
{
GstPadTemplate *pad_template;
GST_DEBUG_OBJECT (enc, "gst_base_audio_encoder_init");
enc->priv = GST_BASE_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_base_audio_encoder_sink_event));
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_setcaps));
gst_pad_set_getcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_getcaps));
gst_pad_set_query_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_query));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_chain));
gst_pad_set_activatepush_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_sink_activate_push));
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_query_function (enc->srcpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_src_query));
gst_pad_set_query_type_function (enc->srcpad,
GST_DEBUG_FUNCPTR (gst_base_audio_encoder_get_query_types));
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 ();
enc->ctx = &enc->priv->ctx;
g_static_rec_mutex_init (&enc->stream_lock);
/* property default */
enc->perfect_ts = DEFAULT_PERFECT_TS;
enc->hard_resync = DEFAULT_HARD_RESYNC;
enc->tolerance = DEFAULT_TOLERANCE;
/* init state */
gst_base_audio_encoder_reset (enc, TRUE);
GST_DEBUG_OBJECT (enc, "init ok");
}
static void
gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
{
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
if (full) {
enc->priv->active = FALSE;
enc->priv->samples_in = 0;
enc->priv->bytes_out = 0;
g_free (enc->ctx->state.channel_pos);
memset (enc->ctx, 0, sizeof (enc->ctx));
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_segment_init (&enc->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_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
static void
gst_base_audio_encoder_finalize (GObject * object)
{
GstBaseAudioEncoder *enc = GST_BASE_AUDIO_ENCODER (object);
g_object_unref (enc->priv->adapter);
g_static_rec_mutex_free (&enc->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* gst_base_audio_encoder_finish_frame:
* @enc: a #GstBaseAudioEncoder
* @buffer: encoded data
* @samples: number of samples (per channel) represented by encoded data
*
* Collects encoded data and/or pushes encoded data downstream.
* Source pad caps must be set when this is called. Depending on the nature
* of the (framing of) the format, subclass can decide whether to push
* encoded data directly or to collect various "frames" in a single buffer.
* Note that the latter behaviour is recommended whenever the format is allowed,
* as it incurs no additional latency and avoids otherwise generating a
* a multitude of (small) output buffers. If not explicitly pushed,
* any available encoded data is pushed at the end of each processing cycle,
* i.e. which encodes as much data as available input data allows.
*
* 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 @buf == NULL => push == TRUE).
*
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
*/
GstFlowReturn
gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
gint samples)
{
GstBaseAudioEncoderClass *klass;
GstBaseAudioEncoderPrivate *priv;
GstBaseAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
priv = enc->priv;
ctx = enc->ctx;
/* subclass should know what it is producing by now */
g_return_val_if_fail (GST_PAD_CAPS (enc->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);
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
buf ? GST_BUFFER_SIZE (buf) : -1, samples);
/* mark subclass still alive and providing */
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 = priv->pending_events; l; l = l->next)
gst_pad_push_event (enc->srcpad, l->data);
g_list_free (pending_events);
}
/* remove corresponding samples from input */
if (samples < 0)
samples = (enc->priv->offset / ctx->state.bpf);
if (G_LIKELY (samples)) {
/* track upstream ts if so configured */
if (!enc->perfect_ts) {
guint64 ts, distance;
ts = gst_adapter_prev_timestamp (priv->adapter, &distance);
g_assert (distance % ctx->state.bpf == 0);
distance /= ctx->state.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->state.rate);
old_ts = priv->base_ts +
gst_util_uint64_scale (priv->samples, GST_SECOND, ctx->state.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->tolerance || diff > enc->tolerance)) {
GST_DEBUG_OBJECT (enc, "marked discont");
priv->discont = TRUE;
}
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;
}
}
/* advance sample view */
if (G_UNLIKELY (samples * ctx->state.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->state.bpf >= gst_adapter_available (priv->adapter))
gst_adapter_clear (priv->adapter);
else
gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
}
} else {
gst_adapter_flush (priv->adapter, samples * ctx->state.bpf);
priv->offset -= samples * ctx->state.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)) {
GST_LOG_OBJECT (enc, "taking %d bytes for output", GST_BUFFER_SIZE (buf));
buf = gst_buffer_make_metadata_writable (buf);
/* decorate */
gst_buffer_set_caps (buf, GST_PAD_CAPS (enc->srcpad));
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->state.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->state.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->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->state.rate);
} else {
GST_BUFFER_OFFSET (buf) = priv->bytes_out;
GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
}
}
priv->bytes_out += GST_BUFFER_SIZE (buf);
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 %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 (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_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
overflow:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
("received more encoded samples %d than provided %d",
samples, priv->offset / ctx->state.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_base_audio_encoder_push_buffers (GstBaseAudioEncoder * enc, gboolean force)
{
GstBaseAudioEncoderClass *klass;
GstBaseAudioEncoderPrivate *priv;
GstBaseAudioEncoderContext *ctx;
gint av, need;
GstBuffer *buf;
GstFlowReturn ret = GST_FLOW_OK;
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
g_return_val_if_fail (klass->handle_frame != NULL, GST_FLOW_ERROR);
priv = enc->priv;
ctx = enc->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->state.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->state.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);
}
}
if (need) {
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = (guint8 *)
gst_adapter_peek (priv->adapter, priv->offset + need) + priv->offset;
GST_BUFFER_SIZE (buf) = need;
}
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->state.bpf;
priv->got_data = FALSE;
ret = klass->handle_frame (enc, buf);
if (G_LIKELY (buf))
gst_buffer_unref (buf);
/* 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_base_audio_encoder_drain (GstBaseAudioEncoder * enc)
{
if (enc->priv->drained)
return GST_FLOW_OK;
else
return gst_base_audio_encoder_push_buffers (enc, TRUE);
}
static void
gst_base_audio_encoder_set_base_gp (GstBaseAudioEncoder * enc)
{
GstClockTime ts;
if (!enc->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->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->ctx->state.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_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
{
GstBaseAudioEncoder *enc;
GstBaseAudioEncoderPrivate *priv;
GstBaseAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
gboolean discont;
enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
priv = enc->priv;
ctx = enc->ctx;
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
/* should know what is coming by now */
if (!ctx->state.bpf)
goto not_negotiated;
GST_LOG_OBJECT (enc,
"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)));
/* input shoud be whole number of sample frames */
if (GST_BUFFER_SIZE (buffer) % ctx->state.bpf)
goto wrong_buffer;
#ifndef GST_DISABLE_GST_DEBUG
{
GstClockTime duration;
GstClockTimeDiff diff;
/* verify buffer duration */
duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
ctx->state.rate * ctx->state.bpf);
diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
(diff > GST_SECOND / ctx->state.rate / 2 ||
diff < -GST_SECOND / ctx->state.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->segment, ctx->state.rate,
ctx->state.bpf);
if (G_UNLIKELY (!buffer)) {
GST_DEBUG_OBJECT (buffer, "no data after clipping to segment");
goto done;
}
GST_LOG_OBJECT (enc,
"buffer after segment clipping has 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)));
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_base_audio_encoder_set_base_gp (enc);
}
/* check for continuity;
* checked elsewhere in non-perfect case */
if (enc->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->state.bpf;
next_ts = priv->base_ts +
gst_util_uint64_scale (samples, GST_SECOND, ctx->state.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->tolerance || diff > enc->tolerance)) {
GST_DEBUG_OBJECT (enc, "marked discont");
discont = TRUE;
}
}
/* do some fancy tweaking in hard resync case */
if (discont && enc->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->state.rate) * ctx->state.bpf;
if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
gst_buffer_unref (buffer);
goto done;
}
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_DATA (buffer) += diff_bytes;
GST_BUFFER_SIZE (buffer) -= diff_bytes;
GST_BUFFER_TIMESTAMP (buffer) += diff;
/* care even less about duration after this */
} else {
/* drain stuff prior to resync */
gst_base_audio_encoder_drain (enc);
}
}
/* now re-sync ts */
priv->base_ts += diff;
gst_base_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_base_audio_encoder_push_buffers (enc, FALSE);
done:
GST_LOG_OBJECT (enc, "chain leaving");
GST_BASE_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 %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
ctx->state.bpf));
gst_buffer_unref (buffer);
ret = GST_FLOW_ERROR;
goto done;
}
}
static gboolean
gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseAudioEncoder *enc;
GstBaseAudioEncoderClass *klass;
GstBaseAudioEncoderContext *ctx;
GstAudioState *state;
gboolean res = TRUE, changed = FALSE;
enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
/* subclass must do something here ... */
g_return_val_if_fail (klass->set_format != NULL, FALSE);
ctx = enc->ctx;
state = &ctx->state;
GST_BASE_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 */
if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && state->rate) {
enc->priv->base_ts +=
GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, state->rate);
enc->priv->samples = 0;
}
if (!gst_base_audio_parse_caps (caps, state, &changed))
goto refuse_caps;
if (changed) {
GstClockTime old_min_latency;
GstClockTime old_max_latency;
/* drain any pending old data stuff */
gst_base_audio_encoder_drain (enc);
/* context defaults */
enc->ctx->frame_samples_min = 0;
enc->ctx->frame_samples_max = 0;
enc->ctx->frame_max = 0;
enc->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);
/* 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_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
goto exit;
}
}
/**
* gst_base_audio_encoder_proxy_getcaps:
* @enc: a #GstBaseAudioEncoder
* @caps: initial
*
* 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_base_audio_encoder_proxy_getcaps (GstBaseAudioEncoder * enc, GstCaps * caps)
{
const GstCaps *templ_caps;
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 ? 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 = gst_caps_copy (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_id_empty_new (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);
gst_caps_merge_structure (filter_caps, s);
}
}
fcaps = gst_caps_intersect (filter_caps, templ_caps);
gst_caps_unref (filter_caps);
done:
gst_caps_replace (&allowed, NULL);
GST_LOG_OBJECT (enc, "proxy caps %" GST_PTR_FORMAT, fcaps);
return fcaps;
}
static GstCaps *
gst_base_audio_encoder_sink_getcaps (GstPad * pad)
{
GstBaseAudioEncoder *enc;
GstBaseAudioEncoderClass *klass;
GstCaps *caps;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
g_assert (pad == enc->sinkpad);
if (klass->getcaps)
caps = klass->getcaps (enc);
else
caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL);
gst_object_unref (enc);
GST_LOG_OBJECT (enc, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
GstEvent * event)
{
GstBaseAudioEncoderClass *klass;
gboolean handled = FALSE;
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
if (format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (enc, "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 {
GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
" -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
", rate %g, applied_rate %g", start, stop, time, rate, arate);
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
break;
}
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
/* finish current segment */
gst_base_audio_encoder_drain (enc);
/* reset partially for new segment */
gst_base_audio_encoder_reset (enc, FALSE);
/* and follow along with segment */
gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
format, start, stop, time);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
GST_BASE_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_base_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_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
case GST_EVENT_EOS:
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_base_audio_encoder_drain (enc);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
default:
break;
}
return handled;
}
static gboolean
gst_base_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseAudioEncoder *enc;
GstBaseAudioEncoderClass *klass;
gboolean handled = FALSE;
gboolean ret = TRUE;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
GST_DEBUG_OBJECT (enc, "received event %d, %s", GST_EVENT_TYPE (event),
GST_EVENT_TYPE_NAME (event));
if (klass->event)
handled = klass->event (enc, event);
if (!handled)
handled = gst_base_audio_encoder_sink_eventfunc (enc, 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_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
ret = TRUE;
}
}
GST_DEBUG_OBJECT (enc, "event handled");
gst_object_unref (enc);
return ret;
}
static gboolean
gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
{
gboolean res = TRUE;
GstBaseAudioEncoder *enc;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
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_base_audio_raw_audio_convert (&enc->ctx->state,
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 (enc);
return res;
}
static const GstQueryType *
gst_base_audio_encoder_get_query_types (GstPad * pad)
{
static const GstQueryType gst_base_audio_encoder_src_query_types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
GST_QUERY_LATENCY,
0
};
return gst_base_audio_encoder_src_query_types;
}
/* 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_base_audio_encoder_src_query (GstPad * pad, GstQuery * query)
{
GstBaseAudioEncoder *enc;
GstPad *peerpad;
gboolean res = FALSE;
enc = GST_BASE_AUDIO_ENCODER (GST_PAD_PARENT (pad));
peerpad = gst_pad_get_peer (GST_PAD (enc->sinkpad));
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;
}
if (!peerpad) {
GST_LOG_OBJECT (enc, "no peer");
break;
}
gst_query_parse_position (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
break;
if ((res = gst_pad_query_convert (peerpad, 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;
}
if (!peerpad) {
GST_LOG_OBJECT (enc, "no peer");
break;
}
gst_query_parse_duration (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
break;
if ((res = gst_pad_query_convert (peerpad, 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_base_audio_encoded_audio_convert (&enc->ctx->state,
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->ctx->min_latency;
if (max_latency != -1)
max_latency += enc->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, query);
break;
}
gst_object_unref (peerpad);
return res;
}
static void
gst_base_audio_encoder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioEncoder *enc;
enc = GST_BASE_AUDIO_ENCODER (object);
switch (prop_id) {
case PROP_PERFECT_TS:
if (enc->granule && !g_value_get_boolean (value))
GST_WARNING_OBJECT (enc, "perfect-ts can not be set FALSE");
else
enc->perfect_ts = g_value_get_boolean (value);
break;
case PROP_HARD_RESYNC:
enc->hard_resync = g_value_get_boolean (value);
break;
case PROP_TOLERANCE:
enc->tolerance = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_audio_encoder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioEncoder *enc;
enc = GST_BASE_AUDIO_ENCODER (object);
switch (prop_id) {
case PROP_PERFECT_TS:
g_value_set_boolean (value, enc->perfect_ts);
break;
case PROP_GRANULE:
g_value_set_boolean (value, enc->granule);
break;
case PROP_HARD_RESYNC:
g_value_set_boolean (value, enc->hard_resync);
break;
case PROP_TOLERANCE:
g_value_set_int64 (value, enc->tolerance);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_base_audio_encoder_activate (GstBaseAudioEncoder * enc, gboolean active)
{
GstBaseAudioEncoderClass *klass;
gboolean result = FALSE;
klass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
g_return_val_if_fail (!enc->granule || enc->perfect_ts, FALSE);
GST_DEBUG_OBJECT (enc, "activate %d", active);
if (active) {
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_base_audio_encoder_reset (enc, TRUE);
}
GST_DEBUG_OBJECT (enc, "activate return: %d", result);
return result;
}
static gboolean
gst_base_audio_encoder_sink_activate_push (GstPad * pad, gboolean active)
{
gboolean result = TRUE;
GstBaseAudioEncoder *enc;
enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (enc, "sink activate push %d", active);
result = gst_base_audio_encoder_activate (enc, active);
if (result)
enc->priv->active = active;
GST_DEBUG_OBJECT (enc, "sink activate push return: %d", result);
gst_object_unref (enc);
return result;
}
Reference in a new issue View git blame Copy permalink