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coral / mtk-gst-plugins-bad / 454c2523355d5a278cf1293c446f308ce4d48c61 / . / ext / wavpack / gstwavpackenc.c
blob: 60ab40a2169b1f863840f51327856a3e969e278f [file] [log] [blame]
/* GStreamer Wavpack encoder plugin
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackdec.c: Wavpack audio encoder
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* 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.
*/
/*
* TODO: - add multichannel handling. channel_mask is:
* front left
* front right
* center
* LFE
* back left
* back right
* front left center
* front right center
* back left
* back center
* side left
* side right
* ...
* - add 32 bit float mode. CONFIG_FLOAT_DATA
*/
#include <string.h>
#include <gst/gst.h>
#include <glib/gprintf.h>
#include <wavpack/wavpack.h>
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
#include "md5.h"
static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
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 void gst_wavpack_enc_finalize (GObject * object);
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,
GValue * value, GParamSpec * pspec);
enum
{
ARG_0,
ARG_MODE,
ARG_BITRATE,
ARG_BITSPERSAMPLE,
ARG_CORRECTION_MODE,
ARG_MD5,
ARG_EXTRA_PROCESSING,
ARG_JOINT_STEREO_MODE,
};
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
#define GST_CAT_DEFAULT gst_wavpack_enc_debug
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) 32, "
"endianness = (int) LITTLE_ENDIAN, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
"audio/x-raw-int, "
"width = (int) 24, "
"depth = (int) 24, "
"endianness = (int) LITTLE_ENDIAN, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
"audio/x-raw-int, "
"width = (int) 16, "
"depth = (int) 16, "
"endianness = (int) LITTLE_ENDIAN, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
"audio/x-raw-int, "
"width = (int) 8, "
"depth = (int) 8, "
"endianness = (int) LITTLE_ENDIAN, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) FALSE")
);
static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) FALSE")
);
#define DEFAULT_MODE 1
#define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
static GType
gst_wavpack_enc_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{0, "Fast Compression", "0"},
{1, "Default", "1"},
{2, "High Compression", "2"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncMode", values);
}
return qtype;
}
#define DEFAULT_CORRECTION_MODE 0
#define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
static GType
gst_wavpack_enc_correction_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{0, "Create no correction file (default)", "0"},
{1, "Create correction file", "1"},
{2, "Create optimized correction file", "2"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
}
return qtype;
}
#define DEFAULT_JS_MODE 0
#define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
static GType
gst_wavpack_enc_joint_stereo_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{0, "auto (default)", "0"},
{1, "left/right", "1"},
{2, "mid/side", "2"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
}
return qtype;
}
GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT);
static void
gst_wavpack_enc_base_init (gpointer klass)
{
static GstElementDetails element_details = {
"Wavpack audio encoder",
"Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>"
};
GstElementClass *element_class = GST_ELEMENT_CLASS (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));
/* set element details */
gst_element_class_set_details (element_class, &element_details);
}
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);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_enc_finalize);
/* set property handlers */
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_get_property);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
"Speed versus compression tradeoff.",
GST_TYPE_WAVPACK_ENC_MODE, DEFAULT_MODE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_double ("bitrate", "Bitrate",
"Try to encode with this average bitrate (bits/sec). "
"This enables lossy encoding! A value smaller than 24000.0 disables this.",
0.0, 9600000.0, 0.0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
g_param_spec_double ("bits-per-sample", "Bits per sample",
"Try to encode with this amount of bits per sample. "
"This enables lossy encoding! A value smaller than 2.0 disables this.",
0.0, 24.0, 0.0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
g_param_spec_enum ("correction_mode", "Correction file mode",
"Use this mode for correction file creation. Only works in lossy mode!",
GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, DEFAULT_CORRECTION_MODE,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_MD5,
g_param_spec_boolean ("md5", "MD5",
"Store MD5 hash of raw samples within the file.", FALSE,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
g_param_spec_boolean ("extra_processing", "Extra processing",
"Extra encode processing.", FALSE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
g_param_spec_enum ("joint_stereo_mode", "Joint-Stereo mode",
"Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
DEFAULT_JS_MODE, G_PARAM_READWRITE));
}
static void
gst_wavpack_enc_init (GstWavpackEnc * wavpack_enc, GstWavpackEncClass * gclass)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpack_enc);
/* setup sink pad, add handlers */
wavpack_enc->sinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
gst_pad_set_setcaps_function (wavpack_enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
gst_pad_set_chain_function (wavpack_enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
gst_pad_set_event_function (wavpack_enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
gst_element_add_pad (GST_ELEMENT (wavpack_enc), wavpack_enc->sinkpad);
/* setup src pad */
wavpack_enc->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
gst_element_add_pad (GST_ELEMENT (wavpack_enc), wavpack_enc->srcpad);
/* initialize object attributes */
wavpack_enc->wp_config = NULL;
wavpack_enc->wp_context = NULL;
wavpack_enc->first_block = NULL;
wavpack_enc->first_block_size = 0;
wavpack_enc->md5_context = NULL;
wavpack_enc->samplerate = 0;
wavpack_enc->width = 0;
wavpack_enc->channels = 0;
wavpack_enc->wv_id = (write_id *) g_malloc0 (sizeof (write_id));
wavpack_enc->wv_id->correction = FALSE;
wavpack_enc->wv_id->wavpack_enc = wavpack_enc;
wavpack_enc->wvc_id = (write_id *) g_malloc0 (sizeof (write_id));
wavpack_enc->wvc_id->correction = TRUE;
wavpack_enc->wvc_id->wavpack_enc = wavpack_enc;
/* set default values of params */
wavpack_enc->mode = 1;
wavpack_enc->bitrate = 0.0;
wavpack_enc->correction_mode = 0;
wavpack_enc->md5 = FALSE;
wavpack_enc->extra_processing = FALSE;
wavpack_enc->joint_stereo_mode = 0;
}
static void
gst_wavpack_enc_finalize (GObject * object)
{
GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (object);
/* free the blockout helpers */
g_free (wavpack_enc->wv_id);
g_free (wavpack_enc->wvc_id);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
int depth = 0;
/* check caps and put relevant parts into our object attributes */
if ((!gst_structure_get_int (structure, "channels", &wavpack_enc->channels))
|| (!gst_structure_get_int (structure, "rate", &wavpack_enc->samplerate))
|| (!gst_structure_get_int (structure, "width", &wavpack_enc->width))
|| (!(gst_structure_get_int (structure, "depth", &depth))
|| depth != wavpack_enc->width)) {
GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
("got invalid caps: %", GST_PTR_FORMAT, caps));
gst_object_unref (wavpack_enc);
return FALSE;
}
/* 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, wavpack_enc->channels,
"rate", G_TYPE_INT, wavpack_enc->samplerate,
"width", G_TYPE_INT, wavpack_enc->width,
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_pad_set_caps (wavpack_enc->srcpad, caps)) {
GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
("setting caps failed: %", GST_PTR_FORMAT, caps));
gst_caps_unref (caps);
gst_object_unref (wavpack_enc);
return FALSE;
}
gst_pad_use_fixed_caps (wavpack_enc->srcpad);
gst_caps_unref (caps);
gst_object_unref (wavpack_enc);
return TRUE;
}
static void
gst_wavpack_enc_set_wp_config (GstWavpackEnc * wavpack_enc)
{
wavpack_enc->wp_config = (WavpackConfig *) g_malloc0 (sizeof (WavpackConfig));
/* set general stream informations in the WavpackConfig */
wavpack_enc->wp_config->bytes_per_sample = (wavpack_enc->width + 7) >> 3;
wavpack_enc->wp_config->bits_per_sample = wavpack_enc->width;
wavpack_enc->wp_config->num_channels = wavpack_enc->channels;
/* TODO: handle more than 2 channels correctly! */
if (wavpack_enc->channels == 1) {
wavpack_enc->wp_config->channel_mask = 0x4;
} else if (wavpack_enc->channels == 2) {
wavpack_enc->wp_config->channel_mask = 0x2 | 0x1;
}
wavpack_enc->wp_config->sample_rate = wavpack_enc->samplerate;
/*
* Set parameters in WavpackConfig
*/
/* Encoding mode */
switch (wavpack_enc->mode) {
case 0:
wavpack_enc->wp_config->flags |= CONFIG_FAST_FLAG;
break;
case 1: /* default */
break;
case 2:
wavpack_enc->wp_config->flags |= CONFIG_HIGH_FLAG;
break;
}
/* Bitrate, enables lossy mode */
if (wavpack_enc->bitrate >= 2.0) {
wavpack_enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
if (wavpack_enc->bitrate >= 24000.0) {
wavpack_enc->wp_config->bitrate = wavpack_enc->bitrate / 1000.0;
wavpack_enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
} else {
wavpack_enc->wp_config->bitrate = wavpack_enc->bitrate;
}
}
/* Correction Mode, only in lossy mode */
if (wavpack_enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
if (wavpack_enc->correction_mode > 0) {
wavpack_enc->wvcsrcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wavpack_enc), "wvcsrc"), "wvcsrc");
/* try to add correction src pad, don't set correction mode on failure */
GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, FALSE, NULL);
gst_element_no_more_pads (GST_ELEMENT (wavpack_enc));
if (!gst_pad_set_caps (wavpack_enc->wvcsrcpad, caps)) {
wavpack_enc->correction_mode = 0;
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL),
("setting correction caps failed: %", GST_PTR_FORMAT, caps));
} else {
gst_pad_use_fixed_caps (wavpack_enc->wvcsrcpad);
if (gst_element_add_pad (GST_ELEMENT (wavpack_enc),
wavpack_enc->wvcsrcpad)) {
wavpack_enc->wp_config->flags |= CONFIG_CREATE_WVC;
if (wavpack_enc->correction_mode == 2) {
wavpack_enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
}
} else {
wavpack_enc->correction_mode = 0;
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL),
("add correction pad failed. no correction file will be created."));
}
gst_caps_unref (caps);
}
}
} else {
if (wavpack_enc->correction_mode > 0) {
wavpack_enc->correction_mode = 0;
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, SETTINGS, (NULL),
("settings correction mode only has effect if a bitrate is provided."));
}
}
gst_element_no_more_pads (GST_ELEMENT (wavpack_enc));
/* MD5, setup MD5 context */
if ((wavpack_enc->md5) && !(wavpack_enc->md5_context)) {
wavpack_enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
wavpack_enc->md5_context = (MD5_CTX *) g_malloc0 (sizeof (MD5_CTX));
MD5Init (wavpack_enc->md5_context);
}
/* Extra encode processing */
if (wavpack_enc->extra_processing) {
wavpack_enc->wp_config->flags |= CONFIG_EXTRA_MODE;
}
/* Joint stereo mode */
switch (wavpack_enc->joint_stereo_mode) {
case 0: /* default */
break;
case 1:
wavpack_enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
wavpack_enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
break;
case 2:
wavpack_enc->wp_config->flags |=
(CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
break;
}
}
static int32_t *
gst_wavpack_enc_format_samples (const uchar * src_data, uint32_t sample_count,
guint width)
{
int32_t *data = (int32_t *) g_malloc0 (sizeof (int32_t) * sample_count);
/* put all samples into an int32_t*, no matter what
* width we have and convert them from little endian
* to host byte order */
switch (width) {
int i;
case 8:
for (i = 0; i < sample_count; i++)
data[i] = (int32_t) (int8_t) src_data[i];
break;
case 16:
for (i = 0; i < sample_count; i++)
data[i] = (int32_t) src_data[2 * i]
| ((int32_t) (int8_t) src_data[2 * i + 1] << 8);
break;
case 24:
for (i = 0; i < sample_count; i++)
data[i] = (int32_t) src_data[3 * i]
| ((int32_t) src_data[3 * i + 1] << 8)
| ((int32_t) (int8_t) src_data[3 * i + 2] << 16);
break;
case 32:
for (i = 0; i < sample_count; i++)
data[i] = (int32_t) src_data[4 * i]
| ((int32_t) src_data[4 * i + 1] << 8)
| ((int32_t) src_data[4 * i + 2] << 16)
| ((int32_t) (int8_t) src_data[4 * i + 3] << 24);
break;
}
return data;
}
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
write_id *wid = (write_id *) id;
GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (wid->wavpack_enc);
GstFlowReturn ret;
GstBuffer *buffer;
guchar *block = (guchar *) data;
if (wid->correction == FALSE) {
/* we got something that should be pushed to the (non-correction) src pad */
/* try to allocate a buffer, compatible with the pad, fail otherwise */
ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->srcpad,
GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (wavpack_enc->srcpad),
&buffer);
if (ret != GST_FLOW_OK) {
wavpack_enc->srcpad_last_return = ret;
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
("Dropped one block (%d bytes) of encoded data while allocating buffer! Reason: '%s'\n",
count, gst_flow_get_name (ret)));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p')
&& (block[3] == 'k')) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_DEBUG ("got %d bytes of encoded wavpack data", count);
gst_wavpack_read_header (&wph, block);
/* if it's the first wavpack block save it for later reference
* i.e. sample count correction and send a NEW_SEGMENT event */
if (wph.block_index == 0) {
GstEvent *event = gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0);
gst_pad_push_event (wavpack_enc->srcpad, event);
wavpack_enc->first_block = g_malloc0 (count);
g_memmove (wavpack_enc->first_block, block, count);
wavpack_enc->first_block_size = count;
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
wavpack_enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
wavpack_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 ("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 */
ret = gst_pad_push (wavpack_enc->srcpad, buffer);
wavpack_enc->srcpad_last_return = ret;
if (ret == GST_FLOW_OK) {
return TRUE;
} else {
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
("Dropped one block (%d bytes) of encoded data while pushing! Reason: '%s'\n",
count, gst_flow_get_name (ret)));
return FALSE;
}
} else if (wid->correction == TRUE) {
/* we got something that should be pushed to the correction src pad */
/* is the correction pad linked? */
if (!gst_pad_is_linked (wavpack_enc->wvcsrcpad)) {
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
("Dropped one block (%d bytes) of encoded correction data because of unlinked pad",
count));
wavpack_enc->wvcsrcpad_last_return = GST_FLOW_NOT_LINKED;
return FALSE;
}
/* try to allocate a buffer, compatible with the pad, fail otherwise */
ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->wvcsrcpad,
GST_BUFFER_OFFSET_NONE, count,
GST_PAD_CAPS (wavpack_enc->wvcsrcpad), &buffer);
if (ret != GST_FLOW_OK) {
wavpack_enc->wvcsrcpad_last_return = ret;
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
("Dropped one block (%d bytes) of encoded correction data while allocating buffer! Reason: '%s'\n",
count, gst_flow_get_name (ret)));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p')
&& (block[3] == 'k')) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_DEBUG ("got %d bytes of encoded wavpack correction data", count);
gst_wavpack_read_header (&wph, block);
/* if it's the first wavpack block send a NEW_SEGMENT
* event */
if (wph.block_index == 0) {
GstEvent *event = gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0);
gst_pad_push_event (wavpack_enc->wvcsrcpad, event);
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
wavpack_enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
wavpack_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 ("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 */
ret = gst_pad_push (wavpack_enc->wvcsrcpad, buffer);
wavpack_enc->wvcsrcpad_last_return = ret;
if (ret == GST_FLOW_OK)
return TRUE;
else {
GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
("Dropped one block (%d bytes) of encoded correction data while pushing! Reason: '%s'\n",
count, gst_flow_get_name (ret)));
return FALSE;
}
} else {
/* (correction != TRUE) && (correction != FALSE), wtf? ignore this */
g_assert_not_reached ();
return TRUE;
}
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count =
GST_BUFFER_SIZE (buf) / ((wavpack_enc->width + 7) >> 3);
int32_t *data;
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 */
wavpack_enc->srcpad_last_return = wavpack_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 (!wavpack_enc->wp_context) {
/* create raw context */
wavpack_enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, wavpack_enc->wv_id,
(wavpack_enc->correction_mode > 0) ? wavpack_enc->wvc_id : NULL);
if (!wavpack_enc->wp_context) {
GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
gst_object_unref (wavpack_enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (wavpack_enc);
/* set the configuration to the context now that we know everything
* and initialize the encoder */
if (!WavpackSetConfiguration (wavpack_enc->wp_context,
wavpack_enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (wavpack_enc->wp_context)) {
GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
WavpackCloseFile (wavpack_enc->wp_context);
gst_object_unref (wavpack_enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
GST_DEBUG ("setup of encoding context successfull");
}
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (wavpack_enc->md5) {
MD5Update (wavpack_enc->md5_context, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
}
/* put all samples into an int32_t*, no matter what
* width we have and convert them from little endian
* to host byte order */
data =
gst_wavpack_enc_format_samples (GST_BUFFER_DATA (buf), sample_count,
wavpack_enc->width);
gst_buffer_unref (buf);
/* encode and handle return values from encoding */
if (WavpackPackSamples (wavpack_enc->wp_context, data,
sample_count / wavpack_enc->channels)) {
GST_DEBUG ("encoding samples successfull");
ret = GST_FLOW_OK;
} else {
if ((wavpack_enc->srcpad_last_return == GST_FLOW_RESEND) ||
(wavpack_enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
ret = GST_FLOW_RESEND;
} else if ((wavpack_enc->srcpad_last_return == GST_FLOW_OK) ||
(wavpack_enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((wavpack_enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(wavpack_enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
} else if ((wavpack_enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
(wavpack_enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
ret = GST_FLOW_WRONG_STATE;
} else {
GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
}
}
g_free (data);
gst_object_unref (wavpack_enc);
return ret;
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * wavpack_enc)
{
GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
0, GST_BUFFER_OFFSET_NONE, 0);
gboolean ret;
g_return_if_fail (wavpack_enc);
g_return_if_fail (wavpack_enc->first_block);
/* update the sample count in the first block */
WavpackUpdateNumSamples (wavpack_enc->wp_context, wavpack_enc->first_block);
/* try to seek to the beginning of the output */
ret = gst_pad_push_event (wavpack_enc->srcpad, event);
if (ret) {
/* try to rewrite the first block */
ret = gst_wavpack_enc_push_block (wavpack_enc->wv_id,
wavpack_enc->first_block, wavpack_enc->first_block_size);
if (ret) {
GST_DEBUG ("rewriting of first block succeeded!");
} else {
GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, WRITE, (NULL),
("rewriting of first block failed while pushing!"));
}
} else {
GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL),
("rewriting of first block failed. Seeking to first block failed!"));
}
}
static gboolean
gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackEnc *wavpack_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 (wavpack_enc->wp_context);
/* write the MD5 sum if we have to write one */
if ((wavpack_enc->md5) && (wavpack_enc->md5_context)) {
guchar md5_digest[16];
MD5Final (md5_digest, wavpack_enc->md5_context);
WavpackStoreMD5Sum (wavpack_enc->wp_context, md5_digest);
}
/* Try to rewrite the first frame with the correct sample number */
if (wavpack_enc->first_block)
gst_wavpack_enc_rewrite_first_block (wavpack_enc);
/* close the context if not already happened */
if (wavpack_enc->wp_context) {
WavpackCloseFile (wavpack_enc->wp_context);
wavpack_enc->wp_context = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
case GST_EVENT_NEWSEGMENT:
if (wavpack_enc->wp_context) {
GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL),
("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 (wavpack_enc);
return ret;
}
static GstStateChangeReturn
gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackEnc *wavpack_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 */
wavpack_enc->srcpad_last_return = wavpack_enc->wvcsrcpad_last_return =
GST_FLOW_OK;
case GST_STATE_CHANGE_READY_TO_PAUSED:
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:
/* close and free everything stream related */
if (wavpack_enc->wp_context) {
WavpackCloseFile (wavpack_enc->wp_context);
wavpack_enc->wp_context = NULL;
}
if (wavpack_enc->wp_config) {
g_free (wavpack_enc->wp_config);
wavpack_enc->wp_config = NULL;
}
if (wavpack_enc->first_block) {
g_free (wavpack_enc->first_block);
wavpack_enc->first_block = NULL;
wavpack_enc->first_block_size = 0;
}
if (wavpack_enc->md5_context) {
g_free (wavpack_enc->md5_context);
wavpack_enc->md5_context = NULL;
}
/* 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 */
wavpack_enc->srcpad_last_return = wavpack_enc->wvcsrcpad_last_return =
GST_FLOW_OK;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
wavpack_enc->mode = g_value_get_enum (value);
break;
case ARG_BITRATE:{
gdouble val = g_value_get_double (value);
if ((val >= 24000.0) && (val <= 9600000.0)) {
wavpack_enc->bitrate = val;
} else {
wavpack_enc->bitrate = 0.0;
}
break;
}
case ARG_BITSPERSAMPLE:{
gdouble val = g_value_get_double (value);
if ((val >= 2.0) && (val <= 24.0)) {
wavpack_enc->bitrate = val;
} else {
wavpack_enc->bitrate = 0.0;
}
break;
}
case ARG_CORRECTION_MODE:
wavpack_enc->correction_mode = g_value_get_enum (value);
break;
case ARG_MD5:
wavpack_enc->md5 = g_value_get_boolean (value);
break;
case ARG_EXTRA_PROCESSING:
wavpack_enc->extra_processing = g_value_get_boolean (value);
break;
case ARG_JOINT_STEREO_MODE:
wavpack_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 *wavpack_enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
g_value_set_enum (value, wavpack_enc->mode);
break;
case ARG_BITRATE:
if (wavpack_enc->bitrate >= 24000.0) {
g_value_set_double (value, wavpack_enc->bitrate);
} else {
g_value_set_double (value, 0.0);
}
break;
case ARG_BITSPERSAMPLE:
if (wavpack_enc->bitrate <= 24.0) {
g_value_set_double (value, wavpack_enc->bitrate);
} else {
g_value_set_double (value, 0.0);
}
break;
case ARG_CORRECTION_MODE:
g_value_set_enum (value, wavpack_enc->correction_mode);
break;
case ARG_MD5:
g_value_set_boolean (value, wavpack_enc->md5);
break;
case ARG_EXTRA_PROCESSING:
g_value_set_boolean (value, wavpack_enc->extra_processing);
break;
case ARG_JOINT_STEREO_MODE:
g_value_set_enum (value, wavpack_enc->joint_stereo_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_wavpack_enc_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackenc",
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
"wavpack encoder");
return TRUE;
}