#include "config.h"
#endif
-#define GST_USE_UNSTABLE_API
#include "gstbaseaudiodecoder.h"
#include <gst/pbutils/descriptions.h>
GstEvent * event);
static gboolean gst_base_audio_decoder_src_event (GstPad * pad,
GstEvent * event);
-static gboolean gst_base_audio_decoder_sink_setcaps (GstPad * pad,
- GstCaps * caps);
-static gboolean gst_base_audio_decoder_src_setcaps (GstPad * pad,
- GstCaps * caps);
static GstFlowReturn gst_base_audio_decoder_chain (GstPad * pad,
GstBuffer * buf);
static gboolean gst_base_audio_decoder_src_query (GstPad * pad,
static void gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec,
gboolean full);
-
-GST_BOILERPLATE (GstBaseAudioDecoder, gst_base_audio_decoder, GstElement,
- GST_TYPE_ELEMENT);
-
-static void
-gst_base_audio_decoder_base_init (gpointer g_class)
-{
-}
+#define gst_base_audio_decoder_parent_class parent_class
+G_DEFINE_TYPE (GstBaseAudioDecoder, gst_base_audio_decoder, GST_TYPE_ELEMENT);
static void
gst_base_audio_decoder_class_init (GstBaseAudioDecoderClass * klass)
}
static void
-gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
- GstBaseAudioDecoderClass * klass)
+gst_base_audio_decoder_init (GstBaseAudioDecoder * dec)
{
+ GstBaseAudioDecoderClass *klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
GstPadTemplate *pad_template;
GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_init");
dec->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_event));
- gst_pad_set_setcaps_function (dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_base_audio_decoder_sink_setcaps));
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_audio_decoder_chain));
gst_pad_set_query_function (dec->sinkpad,
g_return_if_fail (pad_template != NULL);
dec->srcpad = gst_pad_new_from_template (pad_template, "src");
- gst_pad_set_setcaps_function (dec->srcpad,
- GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_setcaps));
gst_pad_set_event_function (dec->srcpad,
GST_DEBUG_FUNCPTR (gst_base_audio_decoder_src_event));
gst_pad_set_query_function (dec->srcpad,
dec->priv->error_count = 0;
gst_base_audio_decoder_clear_queues (dec);
- gst_audio_info_clear (&dec->priv->ctx.info);
+ gst_audio_info_init (&dec->priv->ctx.info);
memset (&dec->priv->ctx, 0, sizeof (dec->priv->ctx));
if (dec->priv->taglist) {
/* automagically perform sanity checking of src caps;
* also extracts output data format */
-static gboolean
-gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
+gboolean
+gst_base_audio_decoder_src_setcaps (GstBaseAudioDecoder * dec, GstCaps * caps)
{
- GstBaseAudioDecoder *dec;
gboolean res = TRUE;
guint old_rate;
- dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
-
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
/* parse caps here to check subclass;
}
static gboolean
-gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_base_audio_decoder_sink_setcaps (GstBaseAudioDecoder * dec, GstCaps * caps)
{
- GstBaseAudioDecoder *dec;
GstBaseAudioDecoderClass *klass;
gboolean res = TRUE;
- dec = GST_BASE_AUDIO_DECODER (gst_pad_get_parent (pad));
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
if (klass->set_format)
res = klass->set_format (dec, caps);
- g_object_unref (dec);
return res;
}
if (G_LIKELY (buf)) {
g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
- GST_LOG_OBJECT (dec, "output buffer of size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_LOG_OBJECT (dec,
+ "output buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
} else {
GST_LOG_OBJECT (dec,
- "buffer after segment clipping has size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ "buffer after segment clipping has size %" G_GSIZE_FORMAT " with ts %"
+ GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
+ gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
} else {
}
gst_adapter_push (priv->adapter_out, buf);
priv->out_dur += GST_BUFFER_DURATION (buf);
- av += GST_BUFFER_SIZE (buf);
+ av += gst_buffer_get_size (buf);
buf = NULL;
}
if (priv->out_dur > dec->priv->latency)
if (G_LIKELY (buf)) {
- /* decorate */
- gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
-
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (dec, "marking discont");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
/* duration should always be valid for raw audio */
g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
- dec->segment.last_stop =
+ dec->segment.position =
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
}
}
GST_LOG_OBJECT (dec, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+ ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
GstBaseAudioDecoderContext *ctx;
gint samples = 0;
GstClockTime ts, next_ts;
+ gsize size;
/* subclass should know what it is producing by now */
- g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
+ g_return_val_if_fail (buf == NULL || gst_pad_has_current_caps (dec->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);
/* no dummy calls please */
g_return_val_if_fail (frames != 0, GST_FLOW_ERROR);
priv = dec->priv;
ctx = &dec->priv->ctx;
+ size = buf ? gst_buffer_get_size (buf) : 0;
GST_LOG_OBJECT (dec, "accepting %d bytes == %d samples for %d frames",
- buf ? GST_BUFFER_SIZE (buf) : -1,
- buf ? GST_BUFFER_SIZE (buf) / ctx->info.bpf : -1, frames);
+ buf ? size : -1, buf ? size / ctx->info.bpf : -1, frames);
/* output shoud be whole number of sample frames */
if (G_LIKELY (buf && ctx->info.bpf)) {
- if (GST_BUFFER_SIZE (buf) % ctx->info.bpf)
+ if (size % ctx->info.bpf)
goto wrong_buffer;
/* per channel least */
- samples = GST_BUFFER_SIZE (buf) / ctx->info.bpf;
+ samples = size / ctx->info.bpf;
}
/* frame and ts book-keeping */
priv->taglist = NULL;
}
- buf = gst_buffer_make_metadata_writable (buf);
+ buf = gst_buffer_make_writable (buf);
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
GST_BUFFER_TIMESTAMP (buf) =
priv->base_ts +
wrong_buffer:
{
GST_ELEMENT_ERROR (dec, STREAM, ENCODE, (NULL),
- ("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
- ctx->info.bpf));
+ ("buffer size %d not a multiple of %d", size, ctx->info.bpf));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
GstBaseAudioDecoderClass * klass, GstBuffer * buffer)
{
if (G_LIKELY (buffer)) {
+ gsize size = gst_buffer_get_size (buffer);
/* keep around for admin */
GST_LOG_OBJECT (dec, "tracking frame size %d, ts %" GST_TIME_FORMAT,
- GST_BUFFER_SIZE (buffer),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+ size, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
g_queue_push_tail (&dec->priv->frames, buffer);
dec->priv->ctx.delay = dec->priv->frames.length;
- dec->priv->bytes_in += GST_BUFFER_SIZE (buffer);
+ dec->priv->bytes_in += size;
} else {
GST_LOG_OBJECT (dec, "providing subclass with NULL frame");
}
priv->prev_ts = ts;
}
buffer = gst_adapter_take_buffer (priv->adapter, len);
- buffer = gst_buffer_make_metadata_writable (buffer);
+ buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) = ts;
flush += len;
} else {
if (G_LIKELY (res == GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "pushing buffer %p of size %u, "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
- GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* should be already, but let's be sure */
- buf = gst_buffer_make_metadata_writable (buf);
+ buf = gst_buffer_make_writable (buf);
/* avoid stray DISCONT from forward processing,
* which have no meaning in reverse pushing */
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
if (G_LIKELY (buf)) {
GST_DEBUG_OBJECT (dec, "gathering buffer %p of size %u, "
"time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf,
- GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
/* add buffer to gather queue */
GST_LOG_OBJECT (dec,
"received buffer of size %d with ts %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buffer),
+ ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buffer),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
gboolean handled = FALSE;
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:
+ case GST_EVENT_SEGMENT:
{
- GstFormat format;
- gdouble rate, arate;
- gint64 start, stop, time;
- gboolean update;
+ GstSegment seg;
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
- &start, &stop, &time);
+ gst_event_copy_segment (event, &seg);
- if (format == GST_FORMAT_TIME) {
- GST_DEBUG_OBJECT (dec, "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);
+ if (seg.format == GST_FORMAT_TIME) {
+ GST_DEBUG_OBJECT (dec, "received TIME SEGMENT %" GST_PTR_FORMAT, &seg);
} else {
- GstFormat dformat = GST_FORMAT_TIME;
-
- GST_DEBUG_OBJECT (dec, "received NEW_SEGMENT %" G_GINT64_FORMAT
- " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT
- ", rate %g, applied_rate %g", start, stop, time, rate, arate);
+ gint64 nstart;
+ GST_DEBUG_OBJECT (dec, "received SEGMENT %" GST_PTR_FORMAT, &seg);
/* handle newsegment resulting from legacy simple seeking */
/* note that we need to convert this whether or not enough data
* to handle initial newsegment */
if (dec->priv->ctx.do_byte_time &&
- gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, start,
- &dformat, &start)) {
+ gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, seg.start,
+ GST_FORMAT_TIME, &nstart)) {
/* best attempt convert */
/* as these are only estimates, stop is kept open-ended to avoid
* premature cutting */
GST_DEBUG_OBJECT (dec, "converted to TIME start %" GST_TIME_FORMAT,
- GST_TIME_ARGS (start));
- format = GST_FORMAT_TIME;
- time = start;
- stop = GST_CLOCK_TIME_NONE;
+ GST_TIME_ARGS (nstart));
+ seg.format = GST_FORMAT_TIME;
+ seg.start = nstart;
+ seg.time = nstart;
+ seg.stop = GST_CLOCK_TIME_NONE;
/* replace event */
gst_event_unref (event);
- event = gst_event_new_new_segment_full (update, rate, arate,
- GST_FORMAT_TIME, start, stop, time);
+ event = gst_event_new_segment (&seg);
} else {
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
break;
/* finish current segment */
gst_base_audio_decoder_drain (dec);
+#if 0
if (update) {
/* time progressed without data, see if we can fill the gap with
* some concealment data */
GST_DEBUG_OBJECT (dec,
- "segment update: plc %d, do_plc %d, last_stop %" GST_TIME_FORMAT,
+ "segment update: plc %d, do_plc %d, position %" GST_TIME_FORMAT,
dec->priv->plc, dec->priv->ctx.do_plc,
- GST_TIME_ARGS (dec->segment.last_stop));
+ GST_TIME_ARGS (dec->segment.position));
if (dec->priv->plc && dec->priv->ctx.do_plc &&
- dec->segment.rate > 0.0 && dec->segment.last_stop < start) {
+ dec->segment.rate > 0.0 && dec->segment.position < start) {
GstBaseAudioDecoderClass *klass;
GstBuffer *buf;
klass = GST_BASE_AUDIO_DECODER_GET_CLASS (dec);
/* hand subclass empty frame with duration that needs covering */
buf = gst_buffer_new ();
- GST_BUFFER_DURATION (buf) = start - dec->segment.last_stop;
+ GST_BUFFER_DURATION (buf) = start - dec->segment.position;
/* best effort, not much error handling */
gst_base_audio_decoder_handle_frame (dec, klass, buf);
}
- } else {
+ } else
+#endif
+ {
/* prepare for next one */
gst_base_audio_decoder_flush (dec, FALSE);
/* and that's where we time from,
* in case upstream does not come up with anything better
* (e.g. upstream BYTE) */
- if (format != GST_FORMAT_TIME) {
- dec->priv->base_ts = start;
+ if (seg.format != GST_FORMAT_TIME) {
+ dec->priv->base_ts = seg.start;
dec->priv->samples = 0;
}
}
/* and follow along with segment */
- gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
- format, start, stop, time);
-
+ dec->segment = seg;
gst_pad_push_event (dec->srcpad, event);
handled = TRUE;
break;
gst_base_audio_decoder_drain (dec);
break;
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ gst_event_parse_caps (event, &caps);
+ gst_base_audio_decoder_sink_setcaps (dec, caps);
+ gst_event_unref (event);
+ handled = TRUE;
+ break;
+ }
default:
break;
}
}
memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
- gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
+ gst_segment_do_seek (&seek_segment, rate, format, flags, start_type,
start_time, end_type, end_time, NULL);
- start_time = seek_segment.last_stop;
+ start_time = seek_segment.position;
- format = GST_FORMAT_BYTES;
if (!gst_pad_query_convert (dec->sinkpad, GST_FORMAT_TIME, start_time,
- &format, &start)) {
+ GST_FORMAT_BYTES, &start)) {
GST_DEBUG_OBJECT (dec, "conversion failed");
return FALSE;
}
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
- GstFormat format, tformat;
+ GstFormat format;
gdouble rate;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
/* ... though a non-time seek can be aided as well */
/* First bring the requested format to time */
- tformat = GST_FORMAT_TIME;
- if (!(res = gst_pad_query_convert (pad, format, cur, &tformat, &tcur)))
+ if (!(res =
+ gst_pad_query_convert (pad, format, cur, GST_FORMAT_TIME, &tcur)))
goto convert_error;
- if (!(res = gst_pad_query_convert (pad, format, stop, &tformat, &tstop)))
+ if (!(res =
+ gst_pad_query_convert (pad, format, stop, GST_FORMAT_TIME,
+ &tstop)))
goto convert_error;
/* then seek with time on the peer */
if (format == GST_FORMAT_TIME && gst_base_audio_decoder_do_byte (dec)) {
gint64 value;
- format = GST_FORMAT_BYTES;
- if (gst_pad_query_peer_duration (dec->sinkpad, &format, &value)) {
+ if (gst_pad_query_peer_duration (dec->sinkpad, GST_FORMAT_BYTES,
+ &value)) {
GST_LOG_OBJECT (dec, "upstream size %" G_GINT64_FORMAT, value);
- format = GST_FORMAT_TIME;
if (gst_pad_query_convert (dec->sinkpad, GST_FORMAT_BYTES, value,
- &format, &value)) {
+ GST_FORMAT_TIME, &value)) {
gst_query_set_duration (query, GST_FORMAT_TIME, value);
res = TRUE;
}
}
/* we start from the last seen time */
- time = dec->segment.last_stop;
+ time = dec->segment.position;
/* correct for the segment values */
time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
/* and convert to the final format */
gst_query_parse_position (query, &format, NULL);
if (!(res = gst_pad_query_convert (pad, GST_FORMAT_TIME, time,
- &format, &value)))
+ format, &value)))
break;
gst_query_set_position (query, format, value);
break;
}
- ret = parent_class->change_state (element, transition);
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
# include "config.h"
#endif
-#define GST_USE_UNSTABLE_API
#include "gstbaseaudioencoder.h"
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
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 const GstQueryType *gst_base_audio_encoder_get_query_types (GstPad *
pad);
-static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad);
+static GstCaps *gst_base_audio_encoder_sink_getcaps (GstPad * pad,
+ GstCaps * filter);
static void
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,
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));
}
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_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)) {
GstBaseAudioEncoderContext *ctx;
GstFlowReturn ret = GST_FLOW_OK;
gboolean discont;
+ gsize size;
enc = GST_BASE_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);
return GST_FLOW_ERROR;
}
static gboolean
-gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_base_audio_encoder_sink_setcaps (GstBaseAudioEncoder * enc, GstCaps * caps)
{
- GstBaseAudioEncoder *enc;
GstBaseAudioEncoderClass *klass;
GstBaseAudioEncoderContext *ctx;
- GstAudioInfo *state, *old_state;
+ GstAudioInfo state;
gboolean res = TRUE, changed = FALSE;
guint old_rate;
- 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->priv->ctx;
- state = &ctx->info;
GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
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_base_audio_encoder_sink_getcaps (GstPad * pad)
+gst_base_audio_encoder_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstBaseAudioEncoder *enc;
GstBaseAudioEncoderClass *klass;
g_assert (pad == enc->sinkpad);
if (klass->getcaps)
- caps = klass->getcaps (enc);
+ caps = klass->getcaps (enc, filter);
else
caps = gst_base_audio_encoder_proxy_getcaps (enc, NULL);
gst_object_unref (enc);
klass = GST_BASE_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_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);
+ enc->segment = seg;
break;
}
gst_base_audio_encoder_drain (enc);
break;
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ gst_event_parse_caps (event, &caps);
+ gst_base_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;
projects / platform / upstream / gstreamer.git / commitdiff