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gstreamer/ext/wavpack/gstwavpackenc.c
Wim Taymans 9365f12d6e GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING
2012-02-08 16:43:30 +01:00

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/* 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.
*/
/**
* SECTION:element-wavpackenc
*
* WavpackEnc encodes raw audio into a framed Wavpack stream.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch 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).
* |[
* gst-launch 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
* ]| 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>
*/
/*
* TODO: - 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"
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_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) [ 1, 32], "
"endianness = (int) BYTE_ORDER, "
"channels = (int) [ 1, 8 ], "
"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) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
);
enum
{
GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
GST_WAVPACK_ENC_MODE_FAST,
GST_WAVPACK_ENC_MODE_DEFAULT,
GST_WAVPACK_ENC_MODE_HIGH,
GST_WAVPACK_ENC_MODE_VERY_HIGH
};
#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[] = {
#if 0
/* Very Fast Compression is not supported yet, but will be supported
* in future wavpack versions */
{GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
#endif
{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}
};
qtype = g_enum_register_static ("GstWavpackEncMode", values);
}
return qtype;
}
enum
{
GST_WAVPACK_CORRECTION_MODE_OFF = 0,
GST_WAVPACK_CORRECTION_MODE_ON,
GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
};
#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[] = {
{GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
{GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
{GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
"Create optimized correction file", "optimized"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
}
return qtype;
}
enum
{
GST_WAVPACK_JS_MODE_AUTO = 0,
GST_WAVPACK_JS_MODE_LEFT_RIGHT,
GST_WAVPACK_JS_MODE_MID_SIDE
};
#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[] = {
{GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
{GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
{GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
}
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);
static void
gst_wavpack_enc_base_init (gpointer klass)
{
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_simple (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;
/* 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, GST_WAVPACK_ENC_MODE_DEFAULT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_uint ("bitrate", "Bitrate",
"Try to encode with this average bitrate (bits/sec). "
"This enables lossy encoding, values smaller than 24000 disable it again.",
0, 9600000, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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, values smaller than 2.0 disable it again.",
0.0, 24.0, 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
g_param_spec_enum ("correction-mode", "Correction stream mode",
"Use this mode for the correction stream. Only works in lossy mode!",
GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
g_param_spec_uint ("extra-processing", "Extra processing",
"Use better but slower filters for better compression/quality.",
0, 6, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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,
GST_WAVPACK_JS_MODE_AUTO,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_wavpack_enc_reset (GstWavpackEnc * enc)
{
/* close and free everything stream related if we already did something */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
if (enc->wp_config) {
g_free (enc->wp_config);
enc->wp_config = NULL;
}
if (enc->first_block) {
g_free (enc->first_block);
enc->first_block = NULL;
}
enc->first_block_size = 0;
if (enc->md5_context) {
g_checksum_free (enc->md5_context);
enc->md5_context = NULL;
}
if (enc->pending_buffer) {
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = NULL;
enc->pending_offset = 0;
}
/* 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;
/* reset stream information */
enc->samplerate = 0;
enc->depth = 0;
enc->channels = 0;
enc->channel_mask = 0;
enc->need_channel_remap = FALSE;
enc->timestamp_offset = GST_CLOCK_TIME_NONE;
enc->next_ts = GST_CLOCK_TIME_NONE;
}
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);
/* initialize object attributes */
enc->wp_config = NULL;
enc->wp_context = NULL;
enc->first_block = NULL;
enc->md5_context = NULL;
gst_wavpack_enc_reset (enc);
enc->wv_id.correction = FALSE;
enc->wv_id.wavpack_enc = enc;
enc->wv_id.passthrough = FALSE;
enc->wvc_id.correction = TRUE;
enc->wvc_id.wavpack_enc = enc;
enc->wvc_id.passthrough = FALSE;
/* set default values of params */
enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
enc->bitrate = 0;
enc->bps = 0.0;
enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
enc->md5 = FALSE;
enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
}
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 void
gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
{
enc->wp_config = g_new0 (WavpackConfig, 1);
/* set general stream informations in the WavpackConfig */
enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
enc->wp_config->bits_per_sample = enc->depth;
enc->wp_config->num_channels = enc->channels;
enc->wp_config->channel_mask = enc->channel_mask;
enc->wp_config->sample_rate = enc->samplerate;
/*
* Set parameters in WavpackConfig
*/
/* Encoding mode */
switch (enc->mode) {
#if 0
case GST_WAVPACK_ENC_MODE_VERY_FAST:
enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG;
enc->wp_config->flags |= CONFIG_FAST_FLAG;
break;
#endif
case GST_WAVPACK_ENC_MODE_FAST:
enc->wp_config->flags |= CONFIG_FAST_FLAG;
break;
case GST_WAVPACK_ENC_MODE_DEFAULT:
break;
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 */
if (enc->bitrate) {
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
enc->wp_config->bitrate = enc->bitrate / 1000.0;
} else if (enc->bps) {
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
enc->wp_config->bitrate = enc->bps;
}
/* Correction Mode, only in lossy mode */
if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
enc->wvcsrcpad =
gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
/* try to add correction src pad, don't set correction mode on failure */
GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
enc->correction_mode = 0;
GST_WARNING_OBJECT (enc, "setting correction caps failed");
} else {
gst_pad_use_fixed_caps (enc->wvcsrcpad);
gst_pad_set_active (enc->wvcsrcpad, TRUE);
gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad);
enc->wp_config->flags |= CONFIG_CREATE_WVC;
if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) {
enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
}
}
gst_caps_unref (caps);
}
} else {
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
enc->correction_mode = 0;
GST_WARNING_OBJECT (enc, "setting correction mode only has "
"any effect if a bitrate is provided.");
}
}
gst_element_no_more_pads (GST_ELEMENT (enc));
/* MD5, setup MD5 context */
if ((enc->md5) && !(enc->md5_context)) {
enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
enc->md5_context = g_checksum_new (G_CHECKSUM_MD5);
}
/* Extra encode processing */
if (enc->extra_processing) {
enc->wp_config->flags |= CONFIG_EXTRA_MODE;
enc->wp_config->xmode = enc->extra_processing;
}
/* Joint stereo mode */
switch (enc->joint_stereo_mode) {
case GST_WAVPACK_JS_MODE_AUTO:
break;
case GST_WAVPACK_JS_MODE_LEFT_RIGHT:
enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
break;
case GST_WAVPACK_JS_MODE_MID_SIDE:
enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
break;
}
}
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 void
gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
gint nsamples)
{
gint i, j;
gint32 tmp[8];
for (i = 0; i < nsamples / enc->channels; i++) {
for (j = 0; j < enc->channels; j++) {
tmp[enc->channel_mapping[j]] = data[j];
}
for (j = 0; j < enc->channels; j++) {
data[j] = tmp[j];
}
data += enc->channels;
}
}
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_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;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
return ret;
}
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);
gboolean ret;
g_return_if_fail (enc);
g_return_if_fail (enc->first_block);
/* update the sample count in the 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);
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
enc->wv_id.passthrough = TRUE;
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
enc->wv_id.passthrough = FALSE;
} 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)
{
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_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:
break;
default:
break;
}
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;
}
}
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, "wavpack_enc", 0,
"Wavpack encoder");
return TRUE;
}
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