# include "config.h"
#endif
-#define GST_USE_UNSTABLE_API
#include "gstaudioencoder.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
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);
gboolean active);
static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event);
-static gboolean gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps);
+static gboolean gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc,
+ GstCaps * caps);
static GstFlowReturn gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_audio_encoder_src_query (GstPad * pad, GstQuery * query);
static gboolean gst_audio_encoder_sink_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_audio_encoder_get_query_types (GstPad * pad);
-static GstCaps *gst_audio_encoder_sink_getcaps (GstPad * pad);
-
-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 (GstAudioEncoder, gst_audio_encoder, GstElement,
- GST_TYPE_ELEMENT, do_init);
+static GstCaps *gst_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter);
static void
gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
+ parent_class = g_type_class_peek_parent (klass);
GST_DEBUG_CATEGORY_INIT (gst_audio_encoder_debug, "audioencoder", 0,
"audio encoder base class");
}
static void
-gst_audio_encoder_base_init (gpointer g_class)
-{
-}
-
-static void
gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
{
GstPadTemplate *pad_template;
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_setcaps_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_setcaps));
gst_pad_set_getcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_audio_encoder_sink_getcaps));
gst_pad_set_query_function (enc->sinkpad,
enc->priv->active = FALSE;
enc->priv->samples_in = 0;
enc->priv->bytes_out = 0;
- gst_audio_info_clear (&enc->priv->ctx.info);
+ gst_audio_info_init (&enc->priv->ctx.info);
memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
if (enc->priv->tags)
ctx = &enc->priv->ctx;
/* subclass should know what it is producing by now */
- g_return_val_if_fail (GST_PAD_CAPS (enc->srcpad) != NULL, GST_FLOW_ERROR);
+ g_return_val_if_fail (gst_pad_has_current_caps (enc->srcpad), GST_FLOW_ERROR);
/* subclass should not hand us no data */
- g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
+ g_return_val_if_fail (buf == NULL || gst_buffer_get_size (buf) > 0,
GST_FLOW_ERROR);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
}
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
- buf ? GST_BUFFER_SIZE (buf) : -1, samples);
+ buf ? gst_buffer_get_size (buf) : -1, samples);
/* mark subclass still alive and providing */
priv->got_data = TRUE;
/* 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);
+ gsize size;
+
+ size = gst_buffer_get_size (buf);
+
+ GST_LOG_OBJECT (enc, "taking %d bytes for output", size);
+ buf = gst_buffer_make_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) */
ctx->info.rate);
} else {
GST_BUFFER_OFFSET (buf) = priv->bytes_out;
- GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + GST_BUFFER_SIZE (buf);
+ GST_BUFFER_OFFSET_END (buf) = priv->bytes_out + size;
}
}
- priv->bytes_out += GST_BUFFER_SIZE (buf);
+ priv->bytes_out += size;
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (enc, "marking discont");
}
GST_LOG_OBJECT (enc, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ ", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
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;
+ const guint8 *data;
+
+ data = gst_adapter_map (priv->adapter, priv->offset + need);
+ buf =
+ gst_buffer_new_wrapped_full ((gpointer) data, NULL, priv->offset,
+ need);
}
GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
priv->got_data = FALSE;
ret = klass->handle_frame (enc, buf);
- if (G_LIKELY (buf))
+ if (G_LIKELY (buf)) {
gst_buffer_unref (buf);
+ gst_adapter_unmap (priv->adapter, 0);
+ }
/* no data to feed, no leftover provided, then bail out */
if (G_UNLIKELY (!buf && !priv->got_data)) {
GstAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
gboolean discont;
+ gsize size;
enc = GST_AUDIO_ENCODER (GST_OBJECT_PARENT (pad));
if (!ctx->info.bpf)
goto not_negotiated;
+ size = gst_buffer_get_size (buffer);
+
GST_LOG_OBJECT (enc,
"received buffer of size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ ", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
/* input shoud be whole number of sample frames */
- if (GST_BUFFER_SIZE (buffer) % ctx->info.bpf)
+ if (size % ctx->info.bpf)
goto wrong_buffer;
#ifndef GST_DISABLE_GST_DEBUG
GstClockTimeDiff diff;
/* verify buffer duration */
- duration = gst_util_uint64_scale (GST_BUFFER_SIZE (buffer), GST_SECOND,
+ duration = gst_util_uint64_scale (size, GST_SECOND,
ctx->info.rate * ctx->info.bpf);
diff = GST_CLOCK_DIFF (duration, GST_BUFFER_DURATION (buffer));
if (GST_BUFFER_DURATION (buffer) != GST_CLOCK_TIME_NONE &&
goto done;
}
+ size = gst_buffer_get_size (buffer);
+
GST_LOG_OBJECT (enc,
"buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ ", duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
diff_bytes =
GST_CLOCK_TIME_TO_FRAMES (-diff, ctx->info.rate) * ctx->info.bpf;
- if (diff_bytes >= GST_BUFFER_SIZE (buffer)) {
+ if (diff_bytes >= size) {
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;
+ buffer = gst_buffer_make_writable (buffer);
+ gst_buffer_resize (buffer, diff_bytes, size - diff_bytes);
GST_BUFFER_TIMESTAMP (buffer) += diff;
/* care even less about duration after this */
wrong_buffer:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
- ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
+ ("buffer size %d not a multiple of %d", gst_buffer_get_size (buffer),
ctx->info.bpf));
gst_buffer_unref (buffer);
ret = GST_FLOW_ERROR;
}
static gboolean
-gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
{
- GstAudioEncoder *enc;
GstAudioEncoderClass *klass;
GstAudioEncoderContext *ctx;
- GstAudioInfo *state, *old_state;
+ GstAudioInfo state;
gboolean res = TRUE, changed = FALSE;
guint old_rate;
- enc = GST_AUDIO_ENCODER (GST_PAD_PARENT (pad));
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
/* subclass must do something here ... */
g_return_val_if_fail (klass->set_format != NULL, FALSE);
ctx = &enc->priv->ctx;
- state = &ctx->info;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
goto refuse_caps;
/* adjust ts tracking to new sample rate */
- old_rate = GST_AUDIO_INFO_RATE (state);
+ old_rate = GST_AUDIO_INFO_RATE (&ctx->info);
if (GST_CLOCK_TIME_IS_VALID (enc->priv->base_ts) && old_rate) {
enc->priv->base_ts +=
GST_FRAMES_TO_CLOCK_TIME (enc->priv->samples, old_rate);
enc->priv->samples = 0;
}
- old_state = gst_audio_info_copy (state);
- if (!gst_audio_info_from_caps (state, caps))
+ if (!gst_audio_info_from_caps (&state, caps))
goto refuse_caps;
- changed = !audio_info_is_equal (state, old_state);
- gst_audio_info_free (old_state);
+ changed = !audio_info_is_equal (&state, &ctx->info);
if (changed) {
GstClockTime old_min_latency;
GST_OBJECT_UNLOCK (enc);
if (klass->set_format)
- res = klass->set_format (enc, state);
+ res = klass->set_format (enc, &state);
/* notify if new latency */
GST_OBJECT_LOCK (enc);
}
static GstCaps *
-gst_audio_encoder_sink_getcaps (GstPad * pad)
+gst_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstAudioEncoder *enc;
GstAudioEncoderClass *klass;
g_assert (pad == enc->sinkpad);
if (klass->getcaps)
- caps = klass->getcaps (enc);
+ caps = klass->getcaps (enc, filter);
else
caps = gst_audio_encoder_proxy_getcaps (enc, NULL);
gst_object_unref (enc);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:
+ case GST_EVENT_SEGMENT:
{
- 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);
+ GstSegment seg;
+
+ gst_event_copy_segment (event, &seg);
+
+ if (seg.format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (enc, "received TIME SEGMENT %" GST_PTR_FORMAT, &seg);
} 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, "received SEGMENT %" GST_PTR_FORMAT, &seg);
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
break;
}
/* reset partially for new segment */
gst_audio_encoder_reset (enc, FALSE);
/* and follow along with segment */
- gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
- format, start, stop, time);
+ enc->segment = seg;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
break;
}
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ gst_event_parse_caps (event, &caps);
+ gst_audio_encoder_sink_setcaps (enc, caps);
+ gst_event_unref (event);
+ handled = TRUE;
+ break;
+ }
+
default:
break;
}
gst_query_parse_position (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
- if (!(res = gst_pad_query_position (peerpad, &fmt, &pos)))
+ if (!(res = gst_pad_query_position (peerpad, fmt, &pos)))
break;
- if ((res = gst_pad_query_convert (peerpad, fmt, pos, &req_fmt, &val))) {
+ if ((res = gst_pad_query_convert (peerpad, fmt, pos, req_fmt, &val))) {
gst_query_set_position (query, req_fmt, val);
}
break;
gst_query_parse_duration (query, &req_fmt, NULL);
fmt = GST_FORMAT_TIME;
- if (!(res = gst_pad_query_duration (peerpad, &fmt, &dur)))
+ if (!(res = gst_pad_query_duration (peerpad, fmt, &dur)))
break;
- if ((res = gst_pad_query_convert (peerpad, fmt, dur, &req_fmt, &val))) {
+ if ((res = gst_pad_query_convert (peerpad, fmt, dur, req_fmt, &val))) {
gst_query_set_duration (query, req_fmt, val);
}
break;
* Sets number of samples (per channel) subclass needs to be handed,
* at least or will be handed all available if 0.
*
+ * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
+ * must be called with the same number.
+ *
* Since: 0.10.36
*/
void
* Sets number of samples (per channel) subclass needs to be handed,
* at most or will be handed all available if 0.
*
+ * If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
+ * must be called with the same number.
+ *
* Since: 0.10.36
*/
void
projects / platform / upstream / gstreamer.git / blobdiff