static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstAudioChannelPosition *pos;
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;
}
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;
}
if (pos == NULL) {
GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
("input has no valid channel layout"));
gst_object_unref (enc);
return FALSE;
}
enc->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
enc->need_channel_remap =
gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping);
g_free (pos);
/* 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);
if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
GST_WARNING_OBJECT (enc, "setting channel layout failed");
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);
gst_caps_unref (caps);
gst_object_unref (enc);
return TRUE;
}
static gboolean
gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
gboolean ret = TRUE;
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
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_object_unref (enc->pending_buffer);
enc->pending_buffer = NULL;
enc->pending_offset = 0;
}
/* write the MD5 sum if we have to write one */
if ((enc->md5) && (enc->md5_context)) {
guchar md5_digest[16];
MD5Final (md5_digest, enc->md5_context);
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
}
/* Try to rewrite the first frame with the correct sample number */
if (enc->first_block)
gst_wavpack_enc_rewrite_first_block (enc);
/* close the context if not already happened */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
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;
}
gst_object_unref (enc);
return ret;
}
static void
gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
enc->mode = g_value_get_enum (value);
break;
case ARG_BITRATE:{
guint val = g_value_get_uint (value);
if ((val >= 24000) && (val <= 9600000)) {
enc->bitrate = val;
enc->bps = 0.0;
} else {
enc->bitrate = 0;
enc->bps = 0.0;
}
break;
}
case ARG_BITSPERSAMPLE:{
gdouble val = g_value_get_double (value);
if ((val >= 2.0) && (val <= 24.0)) {
enc->bps = val;
enc->bitrate = 0;
} else {
enc->bps = 0.0;
enc->bitrate = 0;
}
break;
}
case ARG_CORRECTION_MODE:
enc->correction_mode = g_value_get_enum (value);
break;
case ARG_MD5:
enc->md5 = g_value_get_boolean (value);
break;
case ARG_EXTRA_PROCESSING:
enc->extra_processing = g_value_get_uint (value);
break;
case ARG_JOINT_STEREO_MODE:
enc->joint_stereo_mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
g_value_set_enum (value, enc->mode);
break;
case ARG_BITRATE:
if (enc->bps == 0.0) {
g_value_set_uint (value, enc->bitrate);
} else {
g_value_set_uint (value, 0);
}
break;
case ARG_BITSPERSAMPLE:
if (enc->bitrate == 0) {
g_value_set_double (value, enc->bps);
} else {
g_value_set_double (value, 0.0);
}
break;
case ARG_CORRECTION_MODE:
g_value_set_enum (value, enc->correction_mode);
break;
case ARG_MD5:
g_value_set_boolean (value, enc->md5);
break;
case ARG_EXTRA_PROCESSING:
g_value_set_uint (value, enc->extra_processing);
break;
case ARG_JOINT_STEREO_MODE:
g_value_set_enum (value, enc->joint_stereo_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
{
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;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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:
break;
default:
break;
}
return ret;
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstFlowReturn ret;
/* 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);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
/* create raw 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;
}
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
/* set the configuration to the context now that we know everything
* and initialize the encoder */
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;
}
}
/* 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);
}
/* 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));
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
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) ||
(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 {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
return ret;
}
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
GstFlowReturn *flow;
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
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;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
count, (wid->correction) ? "correction " : "");
gst_wavpack_read_header (&wph, block);
/* Only set when pushing the first buffer again, in that case
* we don't want to delay the buffer or push newsegment events
*/
if (!wid->passthrough) {
/* Only push complete blocks */
if (enc->pending_buffer == NULL) {
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);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
}
if (!(wph.flags & FINAL_BLOCK))
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));
/* 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);
}
}
}
/* 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);
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 (*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;
}
return TRUE;
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstFlowReturn ret;
/* 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);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
/* create raw 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;
}
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
/* set the configuration to the context now that we know everything
* and initialize the encoder */
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");
}
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);
}
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
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) ||
(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 {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
return ret;
}