/* 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:gstaudioencoder

 * @title: 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.

 *

 * ## Configuration

 *

 *   * Initially, GstAudioEncoder calls @start when the encoder element

 *     is activated, which allows subclass to perform any global setup.

 *

 *   * 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.

 *

 *   * GstAudioEncoder calls @stop at end of all processing.

 *

 * 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.

 *

 * ## Data processing

 *

 *     * Base class gathers input sample data (as directed by the context's

 *       frame_samples and frame_max) and provides this to subclass' @handle_frame.

 *     * 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

 *       gst_audio_encoder_finish_frame() (with a NULL buffer and some number of

 *       dropped samples) to indicate dropped (non-encoded) samples.

 *     * Just prior to actually pushing a buffer downstream,

 *       it is passed to @pre_push.

 *     * 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.

 *

 * ## Shutdown phase

 *

 *   * GstAudioEncoder class calls @stop to inform the subclass that data

 *     parsing will be stopped.

 *

 * Subclass is responsible for providing pad template caps for

 * source and sink pads. The pads need to be named "sink" and "src". It also

 * needs to set the fixed caps on srcpad, when the format is ensured.  This

 * is typically when base class calls subclass' @set_format function, though

 * it might be delayed until calling @gst_audio_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:

 *

 *   * Provide pad templates

 *   * Set source pad caps when appropriate

 *   * Inform base class of buffer processing needs using context's

 *      frame_samples and frame_bytes.

 *   * Set user-configurable properties to sane defaults for format and

 *      implementing codec at hand, e.g. those controlling timestamp behaviour

 *      and discontinuity processing.

 *   * Accept data in @handle_frame and provide encoded results to

 *      gst_audio_encoder_finish_frame().

 *

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include "gstaudioencoder.h"

#include "gstaudioutilsprivate.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 */

  /* last negotiated input caps */

  GstCaps *input_caps;

  /* last negotiated input info */

  GstAudioInfo info;

  /* output */

  GstCaps *caps;

  GstCaps *allocation_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;

  /* upstream stream tags (global tags are passed through as-is) */

  GstTagList *upstream_tags;

  /* subclass tags */

  GstTagList *tags;

  GstTagMergeMode tags_merge_mode;

  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 gboolean gst_audio_encoder_negotiate_unlocked (GstAudioEncoder * enc);

static gboolean gst_audio_encoder_transform_meta_default (GstAudioEncoder *

    encoder, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf);

static gboolean gst_audio_encoder_sink_query_default (GstAudioEncoder * encoder,

    GstQuery * query);

static gboolean gst_audio_encoder_src_query_default (GstAudioEncoder * encoder,

    GstQuery * query);

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->sink_query = gst_audio_encoder_sink_query_default;

  klass->src_query = gst_audio_encoder_src_query_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;

  klass->transform_meta = gst_audio_encoder_transform_meta_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 */

  enc->priv->ctx.min_latency = 0;

  enc->priv->ctx.max_latency = 0;

  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;

    GST_OBJECT_LOCK (enc);

    enc->priv->samples_in = 0;

    enc->priv->bytes_out = 0;

    GST_OBJECT_UNLOCK (enc);

    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;

    if (enc->priv->ctx.allocator)

      gst_object_unref (enc->priv->ctx.allocator);

    enc->priv->ctx.allocator = NULL;

    GST_OBJECT_LOCK (enc);

    gst_caps_replace (&enc->priv->ctx.input_caps, NULL);

    gst_caps_replace (&enc->priv->ctx.caps, NULL);

    gst_caps_replace (&enc->priv->ctx.allocation_caps, NULL);

    memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));

    gst_audio_info_init (&enc->priv->ctx.info);

    GST_OBJECT_UNLOCK (enc);

    if (enc->priv->upstream_tags) {

      gst_tag_list_unref (enc->priv->upstream_tags);

      enc->priv->upstream_tags = NULL;

    }

    if (enc->priv->tags)

      gst_tag_list_unref (enc->priv->tags);

    enc->priv->tags = NULL;

    enc->priv->tags_merge_mode = GST_TAG_MERGE_APPEND;

    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;

  }

  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);

}

static inline void

gst_audio_encoder_push_pending_events (GstAudioEncoder * enc)

{

  GstAudioEncoderPrivate *priv = enc->priv;

  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);

  }

}

static GstEvent *

gst_audio_encoder_create_merged_tags_event (GstAudioEncoder * enc)

{

  GstTagList *merged_tags;

  GST_LOG_OBJECT (enc, "upstream : %" GST_PTR_FORMAT, enc->priv->upstream_tags);

  GST_LOG_OBJECT (enc, "encoder  : %" GST_PTR_FORMAT, enc->priv->tags);

  GST_LOG_OBJECT (enc, "mode     : %d", enc->priv->tags_merge_mode);

  merged_tags =

      gst_tag_list_merge (enc->priv->upstream_tags, enc->priv->tags,

      enc->priv->tags_merge_mode);

  GST_DEBUG_OBJECT (enc, "merged   : %" GST_PTR_FORMAT, merged_tags);

  if (merged_tags == NULL)

    return NULL;

  if (gst_tag_list_is_empty (merged_tags)) {

    gst_tag_list_unref (merged_tags);

    return NULL;

  }

  /* add codec info to pending tags */

#if 0

  caps = gst_pad_get_current_caps (enc->srcpad);

  gst_pb_utils_add_codec_description_to_tag_list (merged_tags,

      GST_TAG_AUDIO_CODEC, caps);

#endif

  return gst_event_new_tag (merged_tags);

}

static void

gst_audio_encoder_check_and_push_pending_tags (GstAudioEncoder * enc)

{

  if (enc->priv->tags_changed) {

    GstEvent *tags_event;

    tags_event = gst_audio_encoder_create_merged_tags_event (enc);

    if (tags_event != NULL)

      gst_audio_encoder_push_event (enc, tags_event);

    enc->priv->tags_changed = FALSE;

  }

}

static gboolean

gst_audio_encoder_transform_meta_default (GstAudioEncoder *

    encoder, GstBuffer * outbuf, GstMeta * meta, GstBuffer * inbuf)

{

  const GstMetaInfo *info = meta->info;

  const gchar *const *tags;

  tags = gst_meta_api_type_get_tags (info->api);

  if (!tags || (g_strv_length ((gchar **) tags) == 1

          && gst_meta_api_type_has_tag (info->api,

              g_quark_from_string (GST_META_TAG_AUDIO_STR))))

    return TRUE;

  return FALSE;

}

typedef struct

{

  GstAudioEncoder *encoder;

  GstBuffer *outbuf;

} CopyMetaData;

static gboolean

foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)

{

  CopyMetaData *data = user_data;

  GstAudioEncoder *encoder = data->encoder;

  GstAudioEncoderClass *klass = GST_AUDIO_ENCODER_GET_CLASS (encoder);

  GstBuffer *outbuf = data->outbuf;

  const GstMetaInfo *info = (*meta)->info;

  gboolean do_copy = FALSE;

  if (gst_meta_api_type_has_tag (info->api, _gst_meta_tag_memory)) {

    /* never call the transform_meta with memory specific metadata */

    GST_DEBUG_OBJECT (encoder, "not copying memory specific metadata %s",

        g_type_name (info->api));

    do_copy = FALSE;

  } else if (klass->transform_meta) {

    do_copy = klass->transform_meta (encoder, outbuf, *meta, inbuf);

    GST_DEBUG_OBJECT (encoder, "transformed metadata %s: copy: %d",

        g_type_name (info->api), do_copy);

  }

  /* we only copy metadata when the subclass implemented a transform_meta

   * function and when it returns %TRUE */

  if (do_copy && info->transform_func) {

    GstMetaTransformCopy copy_data = { FALSE, 0, -1 };

    GST_DEBUG_OBJECT (encoder, "copy metadata %s", g_type_name (info->api));

    /* simply copy then */

    info->transform_func (outbuf, *meta, inbuf,

        _gst_meta_transform_copy, &copy_data);

  }

  return TRUE;

}

/**

 * gst_audio_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;

  gboolean needs_reconfigure = FALSE;

  GstBuffer *inbuf = NULL;

  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);

  needs_reconfigure = gst_pad_check_reconfigure (enc->srcpad);

  if (G_UNLIKELY (ctx->output_caps_changed || needs_reconfigure)) {

    if (!gst_audio_encoder_negotiate_unlocked (enc)) {

      gst_pad_mark_reconfigure (enc->srcpad);

      if (GST_PAD_IS_FLUSHING (enc->srcpad))

        ret = GST_FLOW_FLUSHING;

      else

        ret = GST_FLOW_NOT_NEGOTIATED;

      if (buf)

        gst_buffer_unref (buf);

      goto exit;

    }

  }

  /* mark subclass still alive and providing */

  if (G_LIKELY (buf))

    priv->got_data = TRUE;

  gst_audio_encoder_push_pending_events (enc);

  /* send after pending events, which likely includes segment event */

  gst_audio_encoder_check_and_push_pending_tags (enc);

  /* 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_pts (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)) {

      guint avail = gst_adapter_available (priv->adapter);

      if (G_LIKELY (!priv->force)) {

        /* we should have received EOS to enable force */

        goto overflow;

      } else {

        priv->offset = 0;

        if (avail > 0 && samples * ctx->info.bpf >= avail) {

          inbuf = gst_adapter_take_buffer_fast (priv->adapter, avail);

          gst_adapter_clear (priv->adapter);

        } else if (avail > 0) {

          inbuf =

              gst_adapter_take_buffer_fast (priv->adapter,

              samples * ctx->info.bpf);

        }

      }

    } else {

      guint avail = gst_adapter_available (priv->adapter);

      if (avail > 0) {

        inbuf =

            gst_adapter_take_buffer_fast (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);