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wavpackenc: port to audioencoder

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Also adjust unit test to slightly modified behaviour.
This commit is contained in:
Mark Nauwelaerts 2012-02-27 23:45:54 +01:00
parent 9beda57c3a
commit b863df570f
3 changed files with 211 additions and 287 deletions
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478
ext/wavpack/gstwavpackenc.c
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@ -55,12 +55,18 @@
#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,
@ -86,7 +92,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32], "
"depth = (int) { 24, 32 }, "
"endianness = (int) BYTE_ORDER, "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
@ -196,21 +202,8 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
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);
GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
static void
gst_wavpack_enc_base_init (gpointer klass)
@ -220,8 +213,7 @@ gst_wavpack_enc_base_init (gpointer klass)
/* add pad templates */
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);
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",
@ -230,23 +222,24 @@ gst_wavpack_enc_base_init (gpointer klass)
"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;
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (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->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",
@ -304,6 +297,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
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);
@ -330,18 +326,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
static void
gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
{
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;
@ -365,37 +350,51 @@ gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
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);
}
static gboolean
gst_wavpack_enc_start (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (enc, "start");
return TRUE;
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
gst_wavpack_enc_stop (GstAudioEncoder * enc)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gst_wavpack_enc_reset (wpenc);
return TRUE;
}
static gboolean
gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos;
GstCaps *caps;
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);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->depth = GST_AUDIO_INFO_DEPTH (info);
enc->samplerate = GST_AUDIO_INFO_RATE (info);
pos = info->position;
g_assert (pos);
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;
goto invalid_channels;
}
enc->channel_mask =
@ -403,7 +402,6 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
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",
@ -414,18 +412,28 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
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);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), 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
@ -547,21 +555,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
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;
}
buffer = gst_buffer_new_and_alloc (count);
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
@ -597,12 +598,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
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) {
@ -612,29 +615,23 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
}
}
}
/* 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;
samples = 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 %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_pad_push (pad, buffer);
} else {
GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
GST_BUFFER_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",
@ -664,18 +661,25 @@ gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
}
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;
/* 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_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) {
@ -683,13 +687,8 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
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);
@ -699,76 +698,12 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
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;
goto config_failed;
}
GST_DEBUG ("setup of encoding context successfull");
GST_DEBUG_OBJECT (enc, "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),
@ -785,7 +720,7 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * 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");
GST_DEBUG_OBJECT (enc, "encoding samples successful");
ret = GST_FLOW_OK;
} else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
@ -801,15 +736,35 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
(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;
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
@ -826,7 +781,7 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
ret = gst_pad_push_event (enc->srcpad, event);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event);
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
@ -834,111 +789,84 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
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");
/* 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;
}
/* 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);
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");
}
/* 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;
}
return GST_FLOW_OK;
}
static gboolean
gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GST_DEBUG_OBJECT (enc, "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_buffer_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)) {
guint8 md5_digest[16];
gsize digest_len = sizeof (md5_digest);
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");
}
/* 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 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:
/* 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 FALSE;
}
static void

7
ext/wavpack/gstwavpackenc.h
View file

@ -23,6 +23,7 @@
#define __GST_WAVPACK_ENC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <wavpack/wavpack.h>
@ -50,10 +51,9 @@ typedef struct
struct _GstWavpackEnc
{
GstElement element;
GstAudioEncoder element;
/*< private > */
GstPad *sinkpad, *srcpad;
GstPad *wvcsrcpad;
GstFlowReturn srcpad_last_return;
@ -86,6 +86,7 @@ struct _GstWavpackEnc
GstBuffer *pending_buffer;
gint32 pending_offset;
GstEvent *pending_segment;
GstClockTime timestamp_offset;
GstClockTime next_ts;
@ -93,7 +94,7 @@ struct _GstWavpackEnc
struct _GstWavpackEncClass
{
GstElementClass parent;
GstAudioEncoderClass parent;
};
GType gst_wavpack_enc_get_type (void);

13
tests/check/elements/wavpackenc.c
View file

@ -32,14 +32,14 @@ static GstBus *bus;
#define RAW_CAPS_STRING "audio/x-raw-int, " \
"width = (int) 32, " \
"depth = (int) 16, " \
"depth = (int) 32, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true"
#define WAVPACK_CAPS_STRING "audio/x-wavpack, " \
"width = (int) 16, " \
"width = (int) 32, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"framed = (boolean) true"
@ -48,7 +48,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"width = (int) 32, "
"channels = (int) 1, "
"rate = (int) 44100, " "framed = (boolean) true"));
@ -57,7 +57,7 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"depth = (int) 32, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true"));
@ -118,13 +118,10 @@ GST_START_TEST (test_encode_silence)
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackenc, bus);
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE);
@ -134,9 +131,7 @@ GST_START_TEST (test_encode_silence)
fail_if (outbuffer == NULL);
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
"Failed to encode to valid Wavpack frames");