*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
+ * gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
- * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
+ * as the Wavpack encoder only accepts input with 32 bit width.
* |[
- * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
+ * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossless encoding (the file output will be fairly large).
* |[
- * gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
+ * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossy encoding at a certain bitrate (the file will be fairly small).
* </refsect2>
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
-static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
-static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
+static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
+static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
+ GstEvent * event);
+
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
-static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
-static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
- GstStateChange transition);
+static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
+
static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 32, "
- "depth = (int) [ 1, 32], "
- "endianness = (int) BYTE_ORDER, "
- "channels = (int) [ 1, 8 ], "
- "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S32) ", "
+ "layout = (string) interleaved, "
+ "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
- "width = (int) [ 1, 32 ], "
- "channels = (int) [ 1, 2 ], "
+ "depth = (int) [ 1, 32 ], "
+ "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
{GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
{GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
{GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
-#ifndef WAVPACK_OLD_API
{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
-#endif
{0, NULL, NULL}
};
return qtype;
}
-static void
-_do_init (GType object_type)
-{
- const GInterfaceInfo preset_interface_info = {
- NULL, /* interface_init */
- NULL, /* interface_finalize */
- NULL /* interface_data */
- };
-
- g_type_add_interface_static (object_type, GST_TYPE_PRESET,
- &preset_interface_info);
-}
-
-GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement,
- GST_TYPE_ELEMENT, _do_init);
+#define gst_wavpack_enc_parent_class parent_class
+G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
static void
-gst_wavpack_enc_base_init (gpointer klass)
+gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *element_class = (GstElementClass *) (klass);
+ GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
/* add pad templates */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&wvcsrc_factory));
+ gst_element_class_add_static_pad_template (element_class, &sink_factory);
+ gst_element_class_add_static_pad_template (element_class, &src_factory);
+ gst_element_class_add_static_pad_template (element_class, &wvcsrc_factory);
/* set element details */
- gst_element_class_set_details_simple (element_class, "Wavpack audio encoder",
+ gst_element_class_set_static_metadata (element_class, "Wavpack audio encoder",
"Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>");
-}
-
-
-static void
-gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
- GstElementClass *gstelement_class = (GstElementClass *) klass;
-
- parent_class = g_type_class_peek_parent (klass);
-
- /* set state change handler */
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
/* set property handlers */
gobject_class->set_property = gst_wavpack_enc_set_property;
gobject_class->get_property = gst_wavpack_enc_get_property;
+ base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
+ base_class->sink_event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
+
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
g_checksum_free (enc->md5_context);
enc->md5_context = NULL;
}
+ if (enc->pending_segment)
+ gst_event_unref (enc->pending_segment);
+ enc->pending_segment = NULL;
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
}
static void
-gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
+gst_wavpack_enc_init (GstWavpackEnc * enc)
{
- enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_pad_set_setcaps_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
- gst_pad_set_chain_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
- gst_pad_set_event_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
- gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
-
- /* setup src pad */
- enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+ GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
/* initialize object attributes */
enc->wp_config = NULL;
enc->md5 = FALSE;
enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
+
+ /* require perfect ts */
+ gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
+}
+
+
+static gboolean
+gst_wavpack_enc_start (GstAudioEncoder * enc)
+{
+ GST_DEBUG_OBJECT (enc, "start");
+
+ return TRUE;
+}
+
+static gboolean
+gst_wavpack_enc_stop (GstAudioEncoder * enc)
+{
+ GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
+
+ GST_DEBUG_OBJECT (enc, "stop");
+ gst_wavpack_enc_reset (wpenc);
+
+ return TRUE;
}
static gboolean
-gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
+gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- GstStructure *structure = gst_caps_get_structure (caps, 0);
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos;
+ GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
+ GstCaps *caps;
+ guint64 mask = 0;
- if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
- !gst_structure_get_int (structure, "rate", &enc->samplerate) ||
- !gst_structure_get_int (structure, "depth", &enc->depth)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("got invalid caps: %" GST_PTR_FORMAT, caps));
- gst_object_unref (enc);
- return FALSE;
- }
+ /* we may be configured again, but that change should have cleanup context */
+ g_assert (enc->wp_context == NULL);
- pos = gst_audio_get_channel_positions (structure);
- /* If one channel is NONE they'll be all undefined */
- if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
- g_free (pos);
- pos = NULL;
- }
+ enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+ enc->depth = GST_AUDIO_INFO_DEPTH (info);
+ enc->samplerate = GST_AUDIO_INFO_RATE (info);
- if (pos == NULL) {
- GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
- ("input has no valid channel layout"));
+ pos = info->position;
+ g_assert (pos);
- gst_object_unref (enc);
- return FALSE;
+ /* If one channel is NONE they'll be all undefined */
+ if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
+ goto invalid_channels;
}
enc->channel_mask =
enc->need_channel_remap =
gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping);
- g_free (pos);
+
+ /* wavpack caps hold gst mask, not wavpack mask */
+ gst_audio_channel_positions_to_mask (opos, enc->channels, FALSE, &mask);
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
- "width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+ "depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
- if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
- GST_WARNING_OBJECT (enc, "setting channel layout failed");
+ if (mask)
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
- if (!gst_pad_set_caps (enc->srcpad, caps)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("setting caps failed: %" GST_PTR_FORMAT, caps));
- gst_caps_unref (caps);
- gst_object_unref (enc);
- return FALSE;
- }
- gst_pad_use_fixed_caps (enc->srcpad);
+ if (!gst_audio_encoder_set_output_format (benc, caps))
+ goto setting_src_caps_failed;
gst_caps_unref (caps);
- gst_object_unref (enc);
+
+ /* no special feedback to base class; should provide all available samples */
+
return TRUE;
+
+ /* ERRORS */
+setting_src_caps_failed:
+ {
+ GST_DEBUG_OBJECT (enc,
+ "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
+ gst_caps_unref (caps);
+ return FALSE;
+ }
+invalid_channels:
+ {
+ GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
+ return FALSE;
+ }
}
static void
case GST_WAVPACK_ENC_MODE_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
break;
-#ifndef WAVPACK_OLD_API
case GST_WAVPACK_ENC_MODE_VERY_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
break;
-#endif
}
/* Bitrate, enables lossy mode */
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
+ gint samples = 0;
- pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
+ pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow =
- (wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
- srcpad_last_return;
-
- *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
- count, GST_PAD_CAPS (pad), &buffer);
-
- if (*flow != GST_FLOW_OK) {
- GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
- GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
- return FALSE;
- }
+ (wid->correction) ? &enc->
+ wvcsrcpad_last_return : &enc->srcpad_last_return;
- g_memmove (GST_BUFFER_DATA (buffer), block, count);
+ buffer = gst_buffer_new_and_alloc (count);
+ gst_buffer_fill (buffer, 0, data, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
} else if (enc->pending_offset == wph.block_index) {
- enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer);
+ enc->pending_buffer = gst_buffer_append (enc->pending_buffer, buffer);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (enc->pending_buffer);
enc->pending_offset = wph.block_index;
}
- if (!(wph.flags & FINAL_BLOCK))
+ /* Is this the not-final block of multi-channel data? If so, just
+ * accumulate and return here. */
+ if (!(wph.flags & FINAL_BLOCK) && ((block[32] & ID_OPTIONAL_DATA) == 0))
return TRUE;
buffer = enc->pending_buffer;
enc->pending_buffer = NULL;
enc->pending_offset = 0;
- /* if it's the first wavpack block, send a NEW_SEGMENT event */
- if (wph.block_index == 0) {
- gst_pad_push_event (pad,
- gst_event_new_new_segment (FALSE,
- 1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
+ /* only send segment on correction pad,
+ * regular pad is handled normally by baseclass */
+ if (wid->correction && enc->pending_segment) {
+ gst_pad_push_event (pad, enc->pending_segment);
+ enc->pending_segment = NULL;
+ }
+ if (wph.block_index == 0) {
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
- enc->first_block =
- g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
- enc->first_block_size = GST_BUFFER_SIZE (buffer);
+ GstMapInfo map;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ enc->first_block = g_memdup (map.data, map.size);
+ enc->first_block_size = map.size;
+ gst_buffer_unmap (buffer, &map);
}
}
}
+ samples = wph.block_samples;
- /* set buffer timestamp, duration, offset, offset_end from
- * the wavpack header */
- GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
- gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
- enc->samplerate);
- GST_BUFFER_DURATION (buffer) =
- gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
- enc->samplerate);
+ /* decorate buffer */
+ /* NOTE: this will get overwritten by baseclass, but stay for those
+ * that are pushed directly
+ * FIXME: add setting to baseclass to avoid overwriting it ?? */
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
-
- GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
- GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
- /* push the buffer and forward errors */
- GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
- GST_BUFFER_SIZE (buffer));
- *flow = gst_pad_push (pad, buffer);
+ if (wid->correction || wid->passthrough) {
+ /* push the buffer and forward errors */
+ GST_DEBUG_OBJECT (enc, "pushing buffer with %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (buffer));
+ *flow = gst_pad_push (pad, buffer);
+ } else {
+ GST_DEBUG_OBJECT (enc, "handing frame of %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (buffer));
+ *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
+ samples);
+ }
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
}
static GstFlowReturn
-gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
+gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
+ uint32_t sample_count;
GstFlowReturn ret;
+ GstMapInfo map;
+
+ /* base class ensures configuration */
+ g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
- GST_DEBUG ("got %u raw samples", sample_count);
+ if (G_UNLIKELY (!buf))
+ return gst_wavpack_enc_drain (enc);
+
+ sample_count = gst_buffer_get_size (buf) / 4;
+ GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
- if (!enc->wp_context) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("error creating Wavpack context"));
- gst_object_unref (enc);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
+ if (!enc->wp_context)
+ goto context_failed;
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
- ("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
- gst_object_unref (enc);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
- GST_DEBUG ("setup of encoding context successfull");
- }
-
- /* Save the timestamp of the first buffer. This will be later
- * used as offset for all following buffers */
- if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) {
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
- enc->next_ts = GST_BUFFER_TIMESTAMP (buf);
- } else {
- enc->timestamp_offset = 0;
- enc->next_ts = 0;
+ goto config_failed;
}
+ GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
}
- /* Check if we have a continous stream, if not drop some samples or the buffer or
- * insert some silence samples */
- if (enc->next_ts != GST_CLOCK_TIME_NONE &&
- GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
- guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
- guint64 diff_bytes;
-
- GST_WARNING_OBJECT (enc, "Buffer is older than previous "
- "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
- "), cannot handle. Clipping buffer.",
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (enc->next_ts));
-
- diff_bytes =
- GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2;
- if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
- gst_buffer_unref (buf);
- return GST_FLOW_OK;
- }
- buf = gst_buffer_make_metadata_writable (buf);
- GST_BUFFER_DATA (buf) += diff_bytes;
- GST_BUFFER_SIZE (buf) -= diff_bytes;
-
- GST_BUFFER_TIMESTAMP (buf) += diff;
- if (GST_BUFFER_DURATION_IS_VALID (buf))
- GST_BUFFER_DURATION (buf) -= diff;
- }
-
- /* Allow a diff of at most 5 ms */
- if (enc->next_ts != GST_CLOCK_TIME_NONE
- && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) {
- GST_WARNING_OBJECT (enc,
- "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND);
-
- WavpackFlushSamples (enc->wp_context);
- enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts);
- }
- }
-
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
- && GST_BUFFER_DURATION_IS_VALID (buf))
- enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
- else
- enc->next_ts = GST_CLOCK_TIME_NONE;
-
if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf);
- gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
- sample_count);
+ gst_buffer_map (buf, &map, GST_MAP_WRITE);
+ gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
+ gst_buffer_unmap (buf, &map);
}
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
- g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf),
- GST_BUFFER_SIZE (buf));
+ g_checksum_update (enc->md5_context, map.data, map.size);
}
/* encode and handle return values from encoding */
- if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
+ if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
sample_count / enc->channels)) {
- GST_DEBUG ("encoding samples successful");
+ GST_DEBUG_OBJECT (enc, "encoding samples successful");
+ gst_buffer_unmap (buf, &map);
ret = GST_FLOW_OK;
} else {
- if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
- (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
- ret = GST_FLOW_RESEND;
- } else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
+ gst_buffer_unmap (buf, &map);
+ if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
- } else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
- (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
- ret = GST_FLOW_WRONG_STATE;
+ } else if ((enc->srcpad_last_return == GST_FLOW_FLUSHING) &&
+ (enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
+ ret = GST_FLOW_FLUSHING;
} else {
- GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
- ("encoding samples failed"));
- ret = GST_FLOW_ERROR;
+ goto encoding_failed;
}
}
- gst_buffer_unref (buf);
- gst_object_unref (enc);
+exit:
return ret;
+
+ /* ERRORS */
+encoding_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
+ ("encoding samples failed"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
+config_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
+ ("error setting up wavpack encoding context"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
+context_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+ ("error creating Wavpack context"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{
- GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
- 0, GST_BUFFER_OFFSET_NONE, 0);
+ GstSegment segment;
gboolean ret;
+ GstQuery *query;
+ gboolean seekable = FALSE;
g_return_if_fail (enc);
g_return_if_fail (enc->first_block);
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
- ret = gst_pad_push_event (enc->srcpad, event);
+ query = gst_query_new_seeking (GST_FORMAT_BYTES);
+ if (gst_pad_peer_query (GST_AUDIO_ENCODER_SRC_PAD (enc), query)) {
+ GstFormat format;
+
+ gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
+ if (format != GST_FORMAT_BYTES)
+ seekable = FALSE;
+ } else {
+ GST_LOG_OBJECT (enc, "SEEKING query not handled");
+ }
+ gst_query_unref (query);
+
+ if (!seekable) {
+ GST_DEBUG_OBJECT (enc, "downstream not seekable; not rewriting");
+ return;
+ }
+
+ gst_segment_init (&segment, GST_FORMAT_BYTES);
+ ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ gst_event_new_segment (&segment));
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
enc->wv_id.passthrough = FALSE;
+ g_free (enc->first_block);
+ enc->first_block = NULL;
} else {
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
"Seeking to first block failed!");
}
}
-static gboolean
-gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
+static GstFlowReturn
+gst_wavpack_enc_drain (GstWavpackEnc * enc)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- gboolean ret = TRUE;
+ if (!enc->wp_context)
+ return GST_FLOW_OK;
- GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
+ GST_DEBUG_OBJECT (enc, "draining");
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- /* Encode all remaining samples and flush them to the src pads */
- WavpackFlushSamples (enc->wp_context);
-
- /* Drop all remaining data, this is no complete block otherwise
- * it would've been pushed already */
- if (enc->pending_buffer) {
- gst_buffer_unref (enc->pending_buffer);
- enc->pending_buffer = NULL;
- enc->pending_offset = 0;
- }
+ /* Encode all remaining samples and flush them to the src pads */
+ WavpackFlushSamples (enc->wp_context);
- /* write the MD5 sum if we have to write one */
- if ((enc->md5) && (enc->md5_context)) {
- guint8 md5_digest[16];
- gsize digest_len = sizeof (md5_digest);
+ /* Drop all remaining data, this is no complete block otherwise
+ * it would've been pushed already */
+ if (enc->pending_buffer) {
+ gst_buffer_unref (enc->pending_buffer);
+ enc->pending_buffer = NULL;
+ enc->pending_offset = 0;
+ }
- g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
- if (digest_len == sizeof (md5_digest))
- WavpackStoreMD5Sum (enc->wp_context, md5_digest);
- else
- GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
- }
+ /* write the MD5 sum if we have to write one */
+ if ((enc->md5) && (enc->md5_context)) {
+ guint8 md5_digest[16];
+ gsize digest_len = sizeof (md5_digest);
- /* Try to rewrite the first frame with the correct sample number */
- if (enc->first_block)
- gst_wavpack_enc_rewrite_first_block (enc);
+ g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
+ if (digest_len == sizeof (md5_digest)) {
+ WavpackStoreMD5Sum (enc->wp_context, md5_digest);
+ WavpackFlushSamples (enc->wp_context);
+ } else
+ GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
+ }
- /* close the context if not already happened */
- if (enc->wp_context) {
- WavpackCloseFile (enc->wp_context);
- enc->wp_context = NULL;
- }
+ /* Try to rewrite the first frame with the correct sample number */
+ if (enc->first_block)
+ gst_wavpack_enc_rewrite_first_block (enc);
- ret = gst_pad_event_default (pad, event);
- break;
- case GST_EVENT_NEWSEGMENT:
- if (enc->wp_context) {
- GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
- "already started");
- }
- /* drop NEWSEGMENT events, we create our own when pushing
- * the first buffer to the pads */
- gst_event_unref (event);
- ret = TRUE;
- break;
- default:
- ret = gst_pad_event_default (pad, event);
- break;
+ /* close the context if not already happened */
+ if (enc->wp_context) {
+ WavpackCloseFile (enc->wp_context);
+ enc->wp_context = NULL;
}
- gst_object_unref (enc);
- return ret;
+ return GST_FLOW_OK;
}
-static GstStateChangeReturn
-gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
+static gboolean
+gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
- GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
- * as they're only set to something else in WavpackPackSamples() or more
- * specific gst_wavpack_enc_push_block() and nothing happened there yet */
- enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- default:
- break;
- }
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
+ GST_EVENT_TYPE_NAME (event));
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- gst_wavpack_enc_reset (enc);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEGMENT:
+ if (enc->wp_context) {
+ GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
+ "already started");
+ }
+ /* peek and hold NEWSEGMENT events for sending on correction pad */
+ if (enc->pending_segment)
+ gst_event_unref (enc->pending_segment);
+ enc->pending_segment = gst_event_ref (event);
break;
default:
break;
}
- return ret;
+ /* baseclass handles rest */
+ return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
static void
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE;
- GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
+ GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
"Wavpack encoder");
return TRUE;
projects / platform / upstream / gst-plugins-good.git / blobdiff