--- /dev/null
+/* 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
+ * <link linkend="GstBaseAudioEncoder--perfect-ts">perfect-ts</link>.
+ * 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 <link linkend="GstBaseAudioEncoder--tolerance">tolerance</link>,
+ * 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 <link linkend="GstBaseAudioEncoder--hard-resync">hard-resync</link>)
+ * 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.
+ *
+ * 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 <stdlib.h>
+#include <string.h>
+
+#include "gstbaseaudioencoder.h"
+
+#include <gst/audio/audio.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;
+ /* collected encoded data */
+ GstAdapter *adapter_out;
+ /* (estimated) samples (w.r.t. input rate) represented in adapter_out */
+ gint samples_out;
+ /* 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;
+
+ /* MT safe latency; taken from ctx */
+ GstClockTime min_latency;
+ GstClockTime max_latency;
+ /* output bps estimatation */
+ /* global in samples seen */
+ guint64 samples_in;
+ /* global bytes sent out */
+ guint64 bytes_out;
+
+ /* context storage */
+ GstBaseAudioEncoderContext ctx;
+};
+
+
+static GstElementClass *parent_class = NULL;
+
+static void gst_base_audio_encoder_class_init (GstBaseAudioEncoderClass *
+ klass);
+static void gst_base_audio_encoder_init (GstBaseAudioEncoder * parse,
+ GstBaseAudioEncoderClass * klass);
+
+GType
+gst_base_audio_encoder_get_type (void)
+{
+ static GType base_audio_encoder_type = 0;
+
+ if (!base_audio_encoder_type) {
+ static const GTypeInfo base_audio_encoder_info = {
+ sizeof (GstBaseAudioEncoderClass),
+ (GBaseInitFunc) NULL,
+ (GBaseFinalizeFunc) NULL,
+ (GClassInitFunc) gst_base_audio_encoder_class_init,
+ NULL,
+ NULL,
+ sizeof (GstBaseAudioEncoder),
+ 0,
+ (GInstanceInitFunc) gst_base_audio_encoder_init,
+ };
+ const GInterfaceInfo preset_interface_info = {
+ NULL, /* interface_init */
+ NULL, /* interface_finalize */
+ NULL /* interface_data */
+ };
+
+ base_audio_encoder_type = g_type_register_static (GST_TYPE_ELEMENT,
+ "GstBaseAudioEncoder", &base_audio_encoder_info, G_TYPE_FLAG_ABSTRACT);
+
+ g_type_add_interface_static (base_audio_encoder_type, GST_TYPE_PRESET,
+ &preset_interface_info);
+ }
+ return base_audio_encoder_type;
+}
+
+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);
+ parent_class = g_type_class_peek_parent (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-ts",
+ "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",
+ "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",
+ 0, G_MAXINT64, DEFAULT_TOLERANCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+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->priv->adapter_out = gst_adapter_new ();
+ enc->ctx = &enc->priv->ctx;
+
+ /* 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_OBJECT_LOCK (enc);
+
+ if (full) {
+ enc->priv->active = FALSE;
+ enc->priv->samples_in = 0;
+ enc->priv->bytes_out = 0;
+ memset (enc->ctx, 0, sizeof (enc->ctx));
+ enc->ctx->state.bpf = 0;
+ enc->ctx->state.rate = 0;
+ enc->ctx->min_latency = 0;
+ enc->ctx->max_latency = 0;
+ g_free (enc->ctx->state.channel_pos);
+ enc->ctx->state.channel_pos = NULL;
+ }
+
+ gst_segment_init (&enc->segment, GST_FORMAT_TIME);
+
+ gst_adapter_clear (enc->priv->adapter);
+ gst_adapter_clear (enc->priv->adapter_out);
+ 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->samples_out = 0;
+ enc->priv->discont = FALSE;
+
+ GST_OBJECT_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_object_unref (enc->priv->adapter_out);
+
+ 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;
+ gint av;
+
+ 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_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;
+
+ /* remove corresponding samples from input */
+ if (samples < 0)
+ samples = (enc->priv->offset / ctx->state.bpf);
+
+ if (G_LIKELY (samples)) {
+ /* track upstream ts at output collection start if so configured */
+ if (!enc->perfect_ts && !priv->samples_out) {
+ 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);
+ }
+ if (G_LIKELY (buf)) {
+ priv->samples_out += samples;
+ samples = 0;
+ }
+ /* otherwise retain count of to-be-discarded samples */
+ }
+
+ /* collect output */
+ if (G_LIKELY (buf))
+ gst_adapter_push (enc->priv->adapter_out, buf);
+
+ av = gst_adapter_available (priv->adapter_out);
+ if (av) {
+ GST_LOG_OBJECT (enc, "collecting all %d bytes for output", av);
+ buf = gst_adapter_take_buffer (priv->adapter_out, av);
+
+ /* 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", (gint) priv->samples_out);
+ if (G_LIKELY (priv->samples_out > 0)) {
+ priv->samples += priv->samples_out;
+ priv->samples_out = 0;
+ 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);
+ }
+ }
+
+ if (G_UNLIKELY (priv->discont)) {
+ GST_LOG_OBJECT (enc, "marking discont");
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+ priv->discont = FALSE;
+ }
+ // TODO return value ?
+ if (klass->pre_push)
+ klass->pre_push (enc, buf);
+
+ 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)));
+
+ priv->bytes_out += GST_BUFFER_SIZE (buf);
+
+ ret = gst_pad_push (enc->srcpad, buf);
+ GST_LOG_OBJECT (enc, "buffer pushed: %s", gst_flow_get_name (ret));
+ }
+
+ /* account for discarded */
+ priv->samples += samples;
+
+ 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);
+ return GST_FLOW_ERROR;
+ }
+}
+
+ /* 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;
+
+ /* ensure clear start */
+ gst_adapter_clear (priv->adapter_out);
+ priv->samples_out = 0;
+
+ 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 > 0 ? ctx->frame_samples * 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 we have some extra metadata,
+ * provide for integer multiple of frames to allow for better granularity
+ * of processing */
+ if (ctx->frame_samples > 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;
+ }
+ }
+
+ if (gst_adapter_available (priv->adapter_out))
+ gst_base_audio_encoder_finish_frame (enc, NULL, 0);
+
+ 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 GstFlowReturn
+gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
+{
+ GstBaseAudioEncoderClass *bclass;
+ GstBaseAudioEncoder *enc;
+ GstBaseAudioEncoderPrivate *priv;
+ GstBaseAudioEncoderContext *ctx;
+ GstFlowReturn ret = GST_FLOW_OK;
+ gboolean discont;
+
+ enc = GST_BASE_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
+ bclass = GST_BASE_AUDIO_ENCODER_GET_CLASS (enc);
+
+ priv = enc->priv;
+ ctx = enc->ctx;
+
+ /* 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));
+ if (enc->granule) {
+ priv->base_gp =
+ GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate);
+ GST_DEBUG_OBJECT (enc, "new base gp %" G_GINT64_FORMAT,
+ GST_TIME_ARGS (priv->base_gp));
+ }
+ }
+
+ /* 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;
+ if (priv->base_gp >= 0)
+ priv->base_gp =
+ GST_CLOCK_TIME_TO_FRAMES (priv->base_ts, enc->ctx->state.rate);
+ 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");
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
+ ("encoder not initialized"));
+ gst_buffer_unref (buffer);
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+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);
+ return GST_FLOW_ERROR;
+ }
+}
+
+#define CHECK_VALUE(res, var, val) \
+ if (!res) \
+ goto refuse_caps; \
+ if (var != val) \
+ changed = TRUE; \
+ var = val;
+
+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;
+ GstStructure *s;
+ gboolean vb;
+ gint vi;
+
+ 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_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
+
+ if (!gst_caps_is_fixed (caps))
+ goto refuse_caps;
+
+ s = gst_caps_get_structure (caps, 0);
+ /* parse caps here to save subclass the trouble */
+ if (gst_structure_has_name (s, "audio/x-raw-int"))
+ state->xint = TRUE;
+ else if (gst_structure_has_name (s, "audio/x-raw-float"))
+ state->xint = FALSE;
+ else
+ goto refuse_caps;
+
+ res = gst_structure_get_int (s, "rate", &vi);
+ CHECK_VALUE (res, state->rate, vi);
+ res &= gst_structure_get_int (s, "channels", &vi);
+ CHECK_VALUE (res, state->channels, vi);
+ res &= gst_structure_get_int (s, "width", &vi);
+ CHECK_VALUE (res, state->width, vi);
+ res &= (!state->xint || gst_structure_get_int (s, "depth", &vi));
+ CHECK_VALUE (res, state->depth, vi);
+ res &= gst_structure_get_int (s, "endianness", &vi);
+ CHECK_VALUE (res, state->endian, vi);
+ res &= (!state->xint || gst_structure_get_boolean (s, "signed", &vb));
+ CHECK_VALUE (res, state->sign, vb);
+
+ state->bpf = (state->width / 8) * state->channels;
+ GST_LOG_OBJECT (enc, "bpf: %d", state->bpf);
+ if (!state->bpf)
+ goto refuse_caps;
+
+ g_free (state->channel_pos);
+ state->channel_pos = gst_audio_get_channel_positions (s);
+
+ 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 = 0;
+ enc->ctx->frame_max = 0;
+ enc->ctx->lookahead = 0;
+
+ /* element might report latency */
+ old_min_latency = ctx->min_latency;
+ old_max_latency = ctx->max_latency;
+
+ if (klass->set_format)
+ res = klass->set_format (enc, state);
+
+ /* notify if new latency */
+ if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||
+ (ctx->max_latency > 0 && ctx->max_latency != old_max_latency)) {
+ /* post latency message on the bus */
+ gst_element_post_message (GST_ELEMENT (enc),
+ gst_message_new_latency (GST_OBJECT (enc)));
+ GST_OBJECT_LOCK (enc);
+ enc->priv->min_latency = ctx->min_latency;
+ enc->priv->max_latency = ctx->max_latency;
+ GST_OBJECT_UNLOCK (enc);
+ }
+ } else {
+ GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
+ }
+
+ return res;
+
+ /* ERRORS */
+refuse_caps:
+ {
+ GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
+ return res;
+ }
+}
+
+
+/** 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
+ ", accum %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (enc->segment.start),
+ GST_TIME_ARGS (enc->segment.stop),
+ GST_TIME_ARGS (enc->segment.time),
+ GST_TIME_ARGS (enc->segment.accum));
+ } else {
+ GST_DEBUG_OBJECT (enc, "received NEW_SEGMENT %" G_GINT64_FORMAT
+ " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
+ ", accum %" G_GINT64_FORMAT,
+ enc->segment.start, enc->segment.stop,
+ enc->segment.time, enc->segment.accum);
+ GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
+ break;
+ }
+
+ /* 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);
+ break;
+ }
+
+ case GST_EVENT_FLUSH_START:
+ break;
+
+ case GST_EVENT_FLUSH_STOP:
+ /* 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);
+ break;
+
+ case GST_EVENT_EOS:
+ gst_base_audio_encoder_drain (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)
+ ret = gst_pad_event_default (pad, event);
+
+ GST_DEBUG_OBJECT (enc, "event handled");
+
+ gst_object_unref (enc);
+ return ret;
+}
+
+static gboolean
+gst_base_audio_encoder_convert_sink (GstPad * pad, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+{
+ GstBaseAudioEncoder *enc;
+ gboolean res = FALSE;
+ guint scale = 1;
+ gint bytes_per_sample, rate, byterate;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+
+ bytes_per_sample = enc->ctx->state.bpf;
+ rate = enc->ctx->state.rate;
+ byterate = bytes_per_sample * rate;
+
+ if (G_UNLIKELY (bytes_per_sample == 0 || rate == 0)) {
+ GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert");
+ goto exit;
+ }
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_DEFAULT:
+ *dest_value = src_value / bytes_per_sample;
+ res = TRUE;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value =
+ gst_util_uint64_scale_int (src_value, GST_SECOND, byterate);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_DEFAULT:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = src_value * bytes_per_sample;
+ res = TRUE;
+ break;
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, rate);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ scale = bytes_per_sample;
+ /* fallthrough */
+ case GST_FORMAT_DEFAULT:
+ *dest_value = gst_util_uint64_scale_int (src_value,
+ scale * rate, GST_SECOND);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+
+exit:
+ gst_object_unref (enc);
+ return res;
+}
+
+static gboolean
+gst_base_audio_encoder_sink_query (GstPad * pad, GstQuery * query)
+{
+ gboolean res = TRUE;
+
+ 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_encoder_convert_sink (pad, 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:
+ return res;
+}
+
+static gboolean
+gst_base_audio_encoder_convert_src (GstPad * pad, GstFormat src_format,
+ gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
+{
+ GstBaseAudioEncoder *enc;
+ gboolean res = FALSE;
+ gint64 avg;
+
+ enc = GST_BASE_AUDIO_ENCODER (gst_pad_get_parent (pad));
+
+ if (enc->priv->samples_in == 0 ||
+ enc->priv->bytes_out == 0 || enc->ctx->state.rate == 0) {
+ GST_DEBUG_OBJECT (enc, "not enough metadata yet to convert");
+ goto exit;
+ }
+
+ avg = (enc->priv->bytes_out * enc->ctx->state.rate) / (enc->priv->samples_in);
+
+ switch (src_format) {
+ case GST_FORMAT_BYTES:
+ switch (*dest_format) {
+ case GST_FORMAT_TIME:
+ *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, avg);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ case GST_FORMAT_TIME:
+ switch (*dest_format) {
+ case GST_FORMAT_BYTES:
+ *dest_value = gst_util_uint64_scale_int (src_value, avg, GST_SECOND);
+ res = TRUE;
+ break;
+ default:
+ res = FALSE;
+ }
+ break;
+ default:
+ res = FALSE;
+ }
+
+exit:
+ 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 (pad, 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 (pad, 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_encoder_convert_src (pad, 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 (pad, 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->min_latency;
+ if (max_latency != -1)
+ max_latency += enc->priv->max_latency;
+ GST_OBJECT_UNLOCK (enc);
+
+ gst_query_set_latency (query, live, min_latency, max_latency);
+ }
+ }
+ 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff