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gstreamer/subprojects/gst-plugins-good/ext/wavpack/gstwavpackenc.c

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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-wavpackenc
* @title: wavpackenc
*
* WavpackEnc encodes raw audio into a framed Wavpack stream.
* [Wavpack](http://www.wavpack.com/) is an open-source audio codec that
* features both lossless and lossy encoding.
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
* as the Wavpack encoder only accepts input with 32 bit width.
* |[
* gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossless encoding (the file output will be fairly large).
* |[
* gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossy encoding at a certain bitrate (the file will be fairly small).
*
*/
/*
* 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"
#include "gstwavpackelements.h"
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 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,
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, "
"format = (string) " GST_AUDIO_NE (S32) ", "
"layout = (string) interleaved, "
"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"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"},
{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
{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;
}
#define gst_wavpack_enc_parent_class parent_class
G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
#define _do_init \
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0, \
"Wavpack encoder"); \
wavpack_element_init (plugin);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (wavpackenc, "wavpackenc", GST_RANK_NONE,
GST_TYPE_WAVPACK_ENC, _do_init);
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) (klass);
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (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);
/* set element details */
gst_element_class_set_static_metadata (element_class, "Wavpack audio encoder",
"Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* set property handlers */
gobject_class->set_property = gst_wavpack_enc_set_property;
gobject_class->get_property = gst_wavpack_enc_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
base_class->sink_event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
"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));
gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_MODE, 0);
gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, 0);
gst_type_mark_as_plugin_api (GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE, 0);
}
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_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = 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)
{
GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
/* 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;
/* require perfect ts */
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}
static gboolean
gst_wavpack_enc_start (GstAudioEncoder * enc)
{
GST_DEBUG_OBJECT (enc, "start");
return TRUE;
}
static gboolean
gst_wavpack_enc_stop (GstAudioEncoder * enc)
{
GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
gst_wavpack_enc_reset (wpenc);
return TRUE;
}
static gboolean
gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
GstAudioChannelPosition *pos;
GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
GstCaps *caps;
guint64 mask = 0;
/* 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);
/* If one channel is NONE they'll be all undefined */
if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
goto invalid_channels;
}
enc->channel_mask =
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);
/* wavpack caps hold gst mask, not wavpack mask */
gst_audio_channel_positions_to_mask (opos, enc->channels, FALSE, &mask);
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
"depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (mask)
gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
if (!gst_audio_encoder_set_output_format (benc, caps))
goto setting_src_caps_failed;
gst_caps_unref (caps);
/* 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
gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
{
enc->wp_config = g_new0 (WavpackConfig, 1);
/* set general stream information 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;
case GST_WAVPACK_ENC_MODE_VERY_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
break;
}
/* 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;
gint samples = 0;
pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow =
(wid->correction) ? &enc->
wvcsrcpad_last_return : &enc->srcpad_last_return;
buffer = gst_buffer_new_and_alloc (count);
gst_buffer_fill (buffer, 0, data, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
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_append (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;
}
/* Is this the not-final block of multi-channel data? If so, just
* accumulate and return here. */
if (!(wph.flags & FINAL_BLOCK) && ((block[32] & ID_OPTIONAL_DATA) == 0))
return TRUE;
buffer = enc->pending_buffer;
enc->pending_buffer = NULL;
enc->pending_offset = 0;
/* 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) {
GstMapInfo map;
gst_buffer_map (buffer, &map, GST_MAP_READ);
enc->first_block = g_memdup2 (map.data, map.size);
enc->first_block_size = map.size;
gst_buffer_unmap (buffer, &map);
}
}
}
samples = wph.block_samples;
/* decorate buffer */
/* NOTE: this will get overwritten by baseclass, but stay for those
* that are pushed directly
* FIXME: add setting to baseclass to avoid overwriting it ?? */
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
}
if (wid->correction || wid->passthrough) {
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buffer));
*flow = gst_pad_push (pad, buffer);
} else {
GST_DEBUG_OBJECT (enc, "handing frame of %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (buffer));
*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
samples);
}
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
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_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
uint32_t sample_count;
GstFlowReturn ret;
GstMapInfo map;
/* base class ensures configuration */
g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
if (G_UNLIKELY (!buf))
return gst_wavpack_enc_drain (enc);
sample_count = gst_buffer_get_size (buf) / 4;
GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
/* 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)
goto context_failed;
/* 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)) {
WavpackCloseFile (enc->wp_context);
goto config_failed;
}
GST_DEBUG_OBJECT (enc, "setup of encoding context successful");
}
if (enc->need_channel_remap) {
buf = gst_buffer_make_writable (buf);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
gst_buffer_unmap (buf, &map);
}
gst_buffer_map (buf, &map, GST_MAP_READ);
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
g_checksum_update (enc->md5_context, map.data, map.size);
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
sample_count / enc->channels)) {
GST_DEBUG_OBJECT (enc, "encoding samples successful");
gst_buffer_unmap (buf, &map);
ret = GST_FLOW_OK;
} else {
gst_buffer_unmap (buf, &map);
if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
} else if ((enc->srcpad_last_return == GST_FLOW_FLUSHING) &&
(enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
ret = GST_FLOW_FLUSHING;
} else {
goto encoding_failed;
}
}
exit:
return ret;
/* ERRORS */
encoding_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
goto exit;
}
config_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
ret = GST_FLOW_ERROR;
goto exit;
}
context_failed:
{
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
ret = GST_FLOW_ERROR;
goto exit;
}
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{
GstSegment segment;
gboolean ret;
GstQuery *query;
gboolean seekable = FALSE;
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 */
query = gst_query_new_seeking (GST_FORMAT_BYTES);
if (gst_pad_peer_query (GST_AUDIO_ENCODER_SRC_PAD (enc), query)) {
GstFormat format;
gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
if (format != GST_FORMAT_BYTES)
seekable = FALSE;
} else {
GST_LOG_OBJECT (enc, "SEEKING query not handled");
}
gst_query_unref (query);
if (!seekable) {
GST_DEBUG_OBJECT (enc, "downstream not seekable; not rewriting");
return;
}
gst_segment_init (&segment, GST_FORMAT_BYTES);
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (&segment));
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
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;
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 GstFlowReturn
gst_wavpack_enc_drain (GstWavpackEnc * enc)
{
if (!enc->wp_context)
return GST_FLOW_OK;
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_SEGMENT:
if (enc->wp_context) {
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
"already started");
}
/* peek and hold NEWSEGMENT events for sending on correction pad */
if (enc->pending_segment)
gst_event_unref (enc->pending_segment);
enc->pending_segment = gst_event_ref (event);
break;
default:
break;
}
/* baseclass handles rest */
return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
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;
}
}
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