diff options
Diffstat (limited to 'ext/wavpack')
| -rw-r--r-- | ext/wavpack/Makefile.am | 22 | ||||
| -rwxr-xr-x | ext/wavpack/Makefile.in | 943 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpack.c | 54 | ||||
| -rwxr-xr-x | ext/wavpack/gstwavpackcommon.c | 268 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackcommon.h | 75 | ||||
| -rwxr-xr-x | ext/wavpack/gstwavpackdec.c | 483 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackdec.h | 78 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackenc.c | 1010 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackenc.h | 105 | ||||
| -rwxr-xr-x | ext/wavpack/gstwavpackstreamreader.c | 125 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackstreamreader.h | 36 |
11 files changed, 3199 insertions, 0 deletions
diff --git a/ext/wavpack/Makefile.am b/ext/wavpack/Makefile.am new file mode 100644 index 0000000..83163fc --- /dev/null +++ b/ext/wavpack/Makefile.am @@ -0,0 +1,22 @@ +plugin_LTLIBRARIES = libgstwavpack.la + +libgstwavpack_la_SOURCES = \ + gstwavpack.c \ + gstwavpackcommon.c \ + gstwavpackdec.c \ + gstwavpackenc.c \ + gstwavpackstreamreader.c + +libgstwavpack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \ + $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(WAVPACK_CFLAGS) +libgstwavpack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \ + $(GST_BASE_LIBS) $(GST_LIBS) $(WAVPACK_LIBS) +libgstwavpack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) +libgstwavpack_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS) + +noinst_HEADERS = \ + gstwavpackdec.h \ + gstwavpackenc.h \ + gstwavpackcommon.h \ + gstwavpackstreamreader.h + diff --git a/ext/wavpack/Makefile.in b/ext/wavpack/Makefile.in new file mode 100755 index 0000000..31482e4 --- /dev/null +++ b/ext/wavpack/Makefile.in @@ -0,0 +1,943 @@ +# Makefile.in generated by automake 1.14.1 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994-2013 Free Software Foundation, Inc. + +# This Makefile.in is free software; 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Merkatz <v.merkatz@gmx.net> + * + * gstwavpack.c: plugin loader + * + * 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. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gst/gst-i18n-plugin.h> + +#include "gstwavpackdec.h" +#include "gstwavpackenc.h" + +/* debug category for common code */ +GST_DEBUG_CATEGORY (wavpack_debug); + +static gboolean +plugin_init (GstPlugin * plugin) +{ + GST_DEBUG_CATEGORY_INIT (wavpack_debug, "wavpack", 0, "Wavpack elements"); + +#ifdef ENABLE_NLS + GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE, + LOCALEDIR); + bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); + bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8"); +#endif + + return (gst_wavpack_dec_plugin_init (plugin) + && gst_wavpack_enc_plugin_init (plugin)); +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + wavpack, + "Wavpack lossless/lossy audio format handling", + plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/ext/wavpack/gstwavpackcommon.c b/ext/wavpack/gstwavpackcommon.c new file mode 100755 index 0000000..7e3e492 --- /dev/null +++ b/ext/wavpack/gstwavpackcommon.c @@ -0,0 +1,268 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net> + * Copyright (c) 1998 - 2005 Conifer Software + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackcommon.c: common helper functions + * + * 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. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gstwavpackcommon.h" +#include <string.h> + +#include <gst/gst.h> + +GST_DEBUG_CATEGORY_EXTERN (wavpack_debug); +#define GST_CAT_DEFAULT wavpack_debug + +gboolean +gst_wavpack_read_header (WavpackHeader * header, guint8 * buf) +{ + memmove (header, buf, sizeof (WavpackHeader)); + +#ifndef WAVPACK_OLD_API + WavpackLittleEndianToNative (header, (char *) WavpackHeaderFormat); +#else + little_endian_to_native (header, WavpackHeaderFormat); +#endif + + return (memcmp (header->ckID, "wvpk", 4) == 0); +} + +/* inspired by the original one in wavpack */ +gboolean +gst_wavpack_read_metadata (GstWavpackMetadata * wpmd, guint8 * header_data, + guint8 ** p_data) +{ + WavpackHeader hdr; + guint8 *end; + + gst_wavpack_read_header (&hdr, header_data); + end = header_data + hdr.ckSize + 8; + + if (end - *p_data < 2) + return FALSE; + + wpmd->id = GST_READ_UINT8 (*p_data); + wpmd->byte_length = 2 * (guint) GST_READ_UINT8 (*p_data + 1); + + *p_data += 2; + + if ((wpmd->id & ID_LARGE) == ID_LARGE) { + guint extra; + + wpmd->id &= ~ID_LARGE; + + if (end - *p_data < 2) + return FALSE; + + extra = GST_READ_UINT16_LE (*p_data); + wpmd->byte_length += (extra << 9); + *p_data += 2; + } + + if ((wpmd->id & ID_ODD_SIZE) == ID_ODD_SIZE) { + wpmd->id &= ~ID_ODD_SIZE; + --wpmd->byte_length; + } + + if (wpmd->byte_length > 0) { + if (end - *p_data < wpmd->byte_length + (wpmd->byte_length & 1)) { + wpmd->data = NULL; + return FALSE; + } + + wpmd->data = *p_data; + *p_data += wpmd->byte_length + (wpmd->byte_length & 1); + } else { + wpmd->data = NULL; + } + + return TRUE; +} + +gint +gst_wavpack_get_default_channel_mask (gint nchannels) +{ + gint channel_mask = 0; + + /* Set the default channel mask for the given number of channels. + * It's the same as for WAVE_FORMAT_EXTENDED: + * http://www.microsoft.com/whdc/device/audio/multichaud.mspx + */ + switch (nchannels) { + case 11: + channel_mask |= 0x00400; + channel_mask |= 0x00200; + case 9: + channel_mask |= 0x00100; + case 8: + channel_mask |= 0x00080; + channel_mask |= 0x00040; + case 6: + channel_mask |= 0x00020; + channel_mask |= 0x00010; + case 4: + channel_mask |= 0x00008; + case 3: + channel_mask |= 0x00004; + case 2: + channel_mask |= 0x00002; + channel_mask |= 0x00001; + break; + case 1: + /* For mono use front center */ + channel_mask |= 0x00004; + break; + } + + return channel_mask; +} + +static const struct +{ + const guint32 ms_mask; + const GstAudioChannelPosition gst_pos; +} layout_mapping[] = { + { + 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { + 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { + 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { + 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, { + 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { + 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { + 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { + 0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, { + 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { + 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { + 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { + 0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, { + 0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, { + 0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, { + 0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, { + 0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, { + 0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, { + 0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT} +}; + +#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) + +gboolean +gst_wavpack_get_channel_positions (gint num_channels, gint layout, + GstAudioChannelPosition * pos) +{ + gint i, p; + + if (num_channels == 1 && layout == 0x00004) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; + return TRUE; + } + + p = 0; + for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) { + if ((layout & layout_mapping[i].ms_mask) != 0) { + if (p >= num_channels) { + GST_WARNING ("More bits set in the channel layout map than there " + "are channels! Broken file"); + return FALSE; + } + if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) { + GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel " + "layout map - ignoring those channels", layout_mapping[i].ms_mask); + /* what to do? just ignore it and let downstream deal with a channel + * layout that has INVALID positions in it for now ... */ + } + pos[p] = layout_mapping[i].gst_pos; + ++p; + } + } + + if (p != num_channels) { + GST_WARNING ("Only %d bits set in the channel layout map, but there are " + "supposed to be %d channels! Broken file", p, num_channels); + return FALSE; + } + + return TRUE; +} + +GstAudioChannelPosition * +gst_wavpack_get_default_channel_positions (gint nchannels) +{ + GstAudioChannelPosition *pos = g_new (GstAudioChannelPosition, nchannels); + gint i; + + if (nchannels == 1) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; + return pos; + } + + for (i = 0; i < nchannels; i++) + pos[i] = layout_mapping[i].gst_pos; + + return pos; +} + +gint +gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos, + gint nchannels) +{ + gint channel_mask = 0; + gint i, j; + + if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_MONO) { + channel_mask = 0x00000004; + return channel_mask; + } + + /* FIXME: not exactly efficient but otherwise we need an inverse + * mapping table too */ + for (i = 0; i < nchannels; i++) { + for (j = 0; j < MAX_CHANNEL_POSITIONS; j++) { + if (pos[i] == layout_mapping[j].gst_pos) { + channel_mask |= layout_mapping[j].ms_mask; + break; + } + } + } + + return channel_mask; +} + +gboolean +gst_wavpack_set_channel_mapping (GstAudioChannelPosition * pos, gint nchannels, + gint8 * channel_mapping) +{ + gint i, j; + gboolean ret = TRUE; + + for (i = 0; i < nchannels; i++) { + for (j = 0; j < MAX_CHANNEL_POSITIONS; j++) { + if (pos[i] == layout_mapping[j].gst_pos) { + channel_mapping[i] = j; + ret &= (i == j); + break; + } + } + } + + return !ret; +} diff --git a/ext/wavpack/gstwavpackcommon.h b/ext/wavpack/gstwavpackcommon.h new file mode 100644 index 0000000..f4a81bf --- /dev/null +++ b/ext/wavpack/gstwavpackcommon.h @@ -0,0 +1,75 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net> + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackcommon.h: common helper functions + * + * 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. + */ + +#ifndef __GST_WAVPACK_COMMON_H__ +#define __GST_WAVPACK_COMMON_H__ + +#include <gst/gst.h> +#include <gst/audio/audio.h> +#include <wavpack/wavpack.h> + +typedef struct +{ + guint32 byte_length; + guint8 *data; + guint8 id; +} GstWavpackMetadata; + +#define ID_UNIQUE 0x3f +#define ID_OPTIONAL_DATA 0x20 +#define ID_ODD_SIZE 0x40 +#define ID_LARGE 0x80 + +#define ID_DUMMY 0x0 +#define ID_ENCODER_INFO 0x1 +#define ID_DECORR_TERMS 0x2 +#define ID_DECORR_WEIGHTS 0x3 +#define ID_DECORR_SAMPLES 0x4 +#define ID_ENTROPY_VARS 0x5 +#define ID_HYBRID_PROFILE 0x6 +#define ID_SHAPING_WEIGHTS 0x7 +#define ID_FLOAT_INFO 0x8 +#define ID_INT32_INFO 0x9 +#define ID_WV_BITSTREAM 0xa +#define ID_WVC_BITSTREAM 0xb +#define ID_WVX_BITSTREAM 0xc +#define ID_CHANNEL_INFO 0xd + +#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1) +#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2) +#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3) +#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4) +#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5) +#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6) +#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7) + + +gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf); +gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta, + guint8 * header_data, guint8 ** p_data); +gint gst_wavpack_get_default_channel_mask (gint nchannels); +gboolean gst_wavpack_get_channel_positions (gint nchannels, gint layout, GstAudioChannelPosition *pos); +GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels); +gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels); +gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping); + +#endif diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c new file mode 100755 index 0000000..10659ec --- /dev/null +++ b/ext/wavpack/gstwavpackdec.c @@ -0,0 +1,483 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net> + * Copyright (c) 2006 Edward Hervey <bilboed@gmail.com> + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackdec.c: raw Wavpack bitstream decoder + * + * 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-wavpackdec + * + * WavpackDec decodes framed (for example by the WavpackParse element) + * Wavpack streams and decodes them to raw audio. + * <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-1.0 filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink + * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and + * tries to play it back using an automatically found audio sink. + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gst/gst.h> +#include <gst/audio/audio.h> + +#include <math.h> +#include <string.h> + +#include <wavpack/wavpack.h> +#include "gstwavpackdec.h" +#include "gstwavpackcommon.h" +#include "gstwavpackstreamreader.h" + + +GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug); +#define GST_CAT_DEFAULT gst_wavpack_dec_debug + +static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + 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 src_factory = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) S8, " + "layout = (string) interleaved, " + "channels = (int) [ 1, 8 ], " + "rate = (int) [ 6000, 192000 ]; " + "audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S16) ", " + "layout = (string) interleaved, " + "channels = (int) [ 1, 8 ], " + "rate = (int) [ 6000, 192000 ]; " + "audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S32) ", " + "layout = (string) interleaved, " + "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]") + ); + +static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec); +static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec); +static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec, + GstCaps * caps); +static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec, + GstBuffer * buffer); + +static void gst_wavpack_dec_finalize (GObject * object); +static void gst_wavpack_dec_post_tags (GstWavpackDec * dec); + +#define gst_wavpack_dec_parent_class parent_class +G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER); + +static void +gst_wavpack_dec_class_init (GstWavpackDecClass * klass) +{ + GObjectClass *gobject_class = (GObjectClass *) klass; + GstElementClass *element_class = (GstElementClass *) (klass); + GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass); + + 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 (&sink_factory)); + gst_element_class_set_static_metadata (element_class, "Wavpack audio decoder", + "Codec/Decoder/Audio", + "Decodes Wavpack audio data", + "Arwed v. Merkatz <v.merkatz@gmx.net>, " + "Sebastian Dröge <slomo@circular-chaos.org>"); + + gobject_class->finalize = gst_wavpack_dec_finalize; + + base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame); +} + +static void +gst_wavpack_dec_reset (GstWavpackDec * dec) +{ + dec->wv_id.buffer = NULL; + dec->wv_id.position = dec->wv_id.length = 0; + + dec->channels = 0; + dec->channel_mask = 0; + dec->sample_rate = 0; + dec->depth = 0; +} + +static void +gst_wavpack_dec_init (GstWavpackDec * dec) +{ + dec->context = NULL; + dec->stream_reader = gst_wavpack_stream_reader_new (); + + gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE); + + gst_wavpack_dec_reset (dec); +} + +static void +gst_wavpack_dec_finalize (GObject * object) +{ + GstWavpackDec *dec = GST_WAVPACK_DEC (object); + + g_free (dec->stream_reader); + dec->stream_reader = NULL; + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static gboolean +gst_wavpack_dec_start (GstAudioDecoder * dec) +{ + GST_DEBUG_OBJECT (dec, "start"); + + /* never mind a few errors */ + gst_audio_decoder_set_max_errors (dec, 16); + /* don't bother us with flushing */ + gst_audio_decoder_set_drainable (dec, FALSE); + /* aim for some perfect timestamping */ + gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND); + + return TRUE; +} + +static gboolean +gst_wavpack_dec_stop (GstAudioDecoder * dec) +{ + GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec); + + GST_DEBUG_OBJECT (dec, "stop"); + + if (wpdec->context) { + WavpackCloseFile (wpdec->context); + wpdec->context = NULL; + } + + gst_wavpack_dec_reset (wpdec); + + return TRUE; +} + +static void +gst_wavpack_dec_negotiate (GstWavpackDec * dec) +{ + GstAudioInfo info; + GstAudioFormat fmt; + GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, }; + + /* arrange for 1, 2 or 4-byte width == depth output */ + dec->width = dec->depth; + switch (dec->depth) { + case 8: + fmt = GST_AUDIO_FORMAT_S8; + break; + case 16: + fmt = _GST_AUDIO_FORMAT_NE (S16); + break; + case 24: + case 32: + fmt = _GST_AUDIO_FORMAT_NE (S32); + dec->width = 32; + break; + default: + fmt = GST_AUDIO_FORMAT_UNKNOWN; + g_assert_not_reached (); + break; + } + + g_assert (dec->channel_mask != 0); + + if (!gst_wavpack_get_channel_positions (dec->channels, + dec->channel_mask, pos)) + GST_WARNING_OBJECT (dec, "Failed to set channel layout"); + + gst_audio_info_init (&info); + gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos); + + gst_audio_channel_positions_to_valid_order (info.position, info.channels); + gst_audio_get_channel_reorder_map (info.channels, + info.position, pos, dec->channel_reorder_map); + + /* should always succeed */ + gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info); +} + +static gboolean +gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) +{ + /* pretty much nothing to do here, + * we'll parse it all from the stream and setup then */ + + return TRUE; +} + +static void +gst_wavpack_dec_post_tags (GstWavpackDec * dec) +{ + GstTagList *list; + GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES; + gint64 duration, size; + + /* try to estimate the average bitrate */ + if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + format_bytes, &size) && + gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + format_time, &duration) && size > 0 && duration > 0) { + guint64 bitrate; + + list = gst_tag_list_new_empty (); + + bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration); + gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, + (guint) bitrate, NULL); + gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list, + GST_TAG_MERGE_REPLACE); + } +} + +static GstFlowReturn +gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) +{ + GstWavpackDec *dec; + GstBuffer *outbuf = NULL; + GstFlowReturn ret = GST_FLOW_OK; + WavpackHeader wph; + int32_t decoded, unpacked_size; + gboolean format_changed; + gint width, depth, i, j, max; + gint32 *dec_data = NULL; + guint8 *out_data; + GstMapInfo map, omap; + + dec = GST_WAVPACK_DEC (bdec); + + g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR); + + gst_buffer_map (buf, &map, GST_MAP_READ); + + /* check input, we only accept framed input with complete chunks */ + if (map.size < sizeof (WavpackHeader)) + goto input_not_framed; + + if (!gst_wavpack_read_header (&wph, map.data)) + goto invalid_header; + + if (map.size < wph.ckSize + 4 * 1 + 4) + goto input_not_framed; + + if (!(wph.flags & INITIAL_BLOCK)) + goto input_not_framed; + + dec->wv_id.buffer = map.data; + dec->wv_id.length = map.size; + dec->wv_id.position = 0; + + /* create a new wavpack context if there is none yet but if there + * was already one (i.e. caps were set on the srcpad) check whether + * the new one has the same caps */ + if (!dec->context) { + gchar error_msg[80]; + + dec->context = WavpackOpenFileInputEx (dec->stream_reader, + &dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0); + + /* expect this to work */ + if (!dec->context) { + GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg); + goto context_failed; + } + } + + g_assert (dec->context != NULL); + + format_changed = + (dec->sample_rate != WavpackGetSampleRate (dec->context)) || + (dec->channels != WavpackGetNumChannels (dec->context)) || + (dec->depth != WavpackGetBytesPerSample (dec->context) * 8) || +#ifdef WAVPACK_OLD_API + (dec->channel_mask != dec->context->config.channel_mask); +#else + (dec->channel_mask != WavpackGetChannelMask (dec->context)); +#endif + + if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) || + format_changed) { + gint channel_mask; + + dec->sample_rate = WavpackGetSampleRate (dec->context); + dec->channels = WavpackGetNumChannels (dec->context); + dec->depth = WavpackGetBytesPerSample (dec->context) * 8; + +#ifdef WAVPACK_OLD_API + channel_mask = dec->context->config.channel_mask; +#else + channel_mask = WavpackGetChannelMask (dec->context); +#endif + if (channel_mask == 0) + channel_mask = gst_wavpack_get_default_channel_mask (dec->channels); + + dec->channel_mask = channel_mask; + + gst_wavpack_dec_negotiate (dec); + + /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something + * is decoded or after the format has changed */ + gst_wavpack_dec_post_tags (dec); + } + + /* alloc output buffer */ + dec_data = g_malloc (4 * wph.block_samples * dec->channels); + + /* decode */ + decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples); + if (decoded != wph.block_samples) + goto decode_error; + + unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels; + outbuf = gst_buffer_new_and_alloc (unpacked_size); + + /* legacy; pass along offset, whatever that might entail */ + GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf); + + gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); + out_data = omap.data; + + width = dec->width; + depth = dec->depth; + max = dec->channels * wph.block_samples; + if (width == 8) { + gint8 *outbuffer = (gint8 *) out_data; + gint *reorder_map = dec->channel_reorder_map; + + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]); + } + } else if (width == 16) { + gint16 *outbuffer = (gint16 *) out_data; + gint *reorder_map = dec->channel_reorder_map; + + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]); + } + } else if (dec->width == 32) { + gint32 *outbuffer = (gint32 *) out_data; + gint *reorder_map = dec->channel_reorder_map; + + if (width != depth) { + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = + (gint32) (dec_data[i + reorder_map[j]] << (width - depth)); + } + } else { + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]); + } + } + } else { + g_assert_not_reached (); + } + + gst_buffer_unmap (outbuf, &omap); + gst_buffer_unmap (buf, &map); + buf = NULL; + + g_free (dec_data); + + ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1); + +out: + if (buf) + gst_buffer_unmap (buf, &map); + + if (G_UNLIKELY (ret != GST_FLOW_OK)) { + GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret)); + } + + return ret; + +/* ERRORS */ +input_not_framed: + { + GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input")); + ret = GST_FLOW_ERROR; + goto out; + } +invalid_header: + { + GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header")); + ret = GST_FLOW_ERROR; + goto out; + } +context_failed: + { + GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL), + ("error creating Wavpack context"), ret); + goto out; + } +decode_error: + { + const gchar *reason = "unknown"; + + if (dec->context) { +#ifdef WAVPACK_OLD_API + reason = dec->context->error_message; +#else + reason = WavpackGetErrorMessage (dec->context); +#endif + } else { + reason = "couldn't create decoder context"; + } + GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), + ("decoding error: %s", reason), ret); + g_free (dec_data); + if (ret == GST_FLOW_OK) + gst_audio_decoder_finish_frame (bdec, NULL, 1); + goto out; + } +} + +gboolean +gst_wavpack_dec_plugin_init (GstPlugin * plugin) +{ + if (!gst_element_register (plugin, "wavpackdec", + GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC)) + return FALSE; + GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0, + "Wavpack decoder"); + return TRUE; +} diff --git a/ext/wavpack/gstwavpackdec.h b/ext/wavpack/gstwavpackdec.h new file mode 100644 index 0000000..8a002b4 --- /dev/null +++ b/ext/wavpack/gstwavpackdec.h @@ -0,0 +1,78 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2004 Arwed v. Merkatz <v.merkatz@gmx.net> + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackdec.h: raw Wavpack bitstream decoder + * + * 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. + */ + +#ifndef __GST_WAVPACK_DEC_H__ +#define __GST_WAVPACK_DEC_H__ + +#include <gst/gst.h> +#include <gst/audio/gstaudiodecoder.h> + +#include <wavpack/wavpack.h> + +#include "gstwavpackstreamreader.h" + +G_BEGIN_DECLS +#define GST_TYPE_WAVPACK_DEC \ + (gst_wavpack_dec_get_type()) +#define GST_WAVPACK_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_DEC,GstWavpackDec)) +#define GST_WAVPACK_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_DEC,GstWavpackDecClass)) +#define GST_IS_WAVPACK_DEC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_DEC)) +#define GST_IS_WAVPACK_DEC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_DEC)) +typedef struct _GstWavpackDec GstWavpackDec; +typedef struct _GstWavpackDecClass GstWavpackDecClass; + +struct _GstWavpackDec +{ + GstAudioDecoder element; + + /*< private > */ + + WavpackContext *context; + WavpackStreamReader *stream_reader; + + read_id wv_id; + + gint sample_rate; + gint depth; + gint width; + gint channels; + gint channel_mask; + + gint channel_reorder_map[64]; + +}; + +struct _GstWavpackDecClass +{ + GstAudioDecoderClass parent; +}; + +GType gst_wavpack_dec_get_type (void); + +gboolean gst_wavpack_dec_plugin_init (GstPlugin * plugin); + +G_END_DECLS +#endif /* __GST_WAVPACK_DEC_H__ */ diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c new file mode 100644 index 0000000..cb69739 --- /dev/null +++ b/ext/wavpack/gstwavpackenc.c @@ -0,0 +1,1010 @@ +/* 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 + * + * 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-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). + * </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 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"}, +#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; +} + +#define gst_wavpack_enc_parent_class parent_class +G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER); + +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_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_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)); +} + +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); +} + + +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 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; + 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_memdup (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 successfull"); + } + + 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; + } +} + +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; +} diff --git a/ext/wavpack/gstwavpackenc.h b/ext/wavpack/gstwavpackenc.h new file mode 100644 index 0000000..26a5b11 --- /dev/null +++ b/ext/wavpack/gstwavpackenc.h @@ -0,0 +1,105 @@ +/* GStreamer Wavpack encoder plugin + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackenc.h: 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. + */ + +#ifndef __GST_WAVPACK_ENC_H__ +#define __GST_WAVPACK_ENC_H__ + +#include <gst/gst.h> +#include <gst/audio/gstaudioencoder.h> + +#include <wavpack/wavpack.h> + +G_BEGIN_DECLS +#define GST_TYPE_WAVPACK_ENC \ + (gst_wavpack_enc_get_type()) +#define GST_WAVPACK_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_ENC,GstWavpackEnc)) +#define GST_WAVPACK_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_ENC,GstWavpackEnc)) +#define GST_IS_WAVPACK_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_ENC)) +#define GST_IS_WAVPACK_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_ENC)) +typedef struct _GstWavpackEnc GstWavpackEnc; +typedef struct _GstWavpackEncClass GstWavpackEncClass; + +typedef struct +{ + gboolean correction; + GstWavpackEnc *wavpack_enc; + gboolean passthrough; +} GstWavpackEncWriteID; + + +struct _GstWavpackEnc +{ + GstAudioEncoder element; + + /*< private > */ + GstPad *wvcsrcpad; + + GstFlowReturn srcpad_last_return; + GstFlowReturn wvcsrcpad_last_return; + + WavpackConfig *wp_config; + WavpackContext *wp_context; + + gint samplerate; + gint channels; + gint channel_mask; + gint8 channel_mapping[8]; + gboolean need_channel_remap; + gint depth; + + GstWavpackEncWriteID wv_id; + GstWavpackEncWriteID wvc_id; + + guint mode; + guint bitrate; + gdouble bps; + guint correction_mode; + gboolean md5; + GChecksum *md5_context; + guint extra_processing; + guint joint_stereo_mode; + + void *first_block; + int32_t first_block_size; + + GstBuffer *pending_buffer; + gint32 pending_offset; + GstEvent *pending_segment; + + GstClockTime timestamp_offset; + GstClockTime next_ts; +}; + +struct _GstWavpackEncClass +{ + GstAudioEncoderClass parent; +}; + +GType gst_wavpack_enc_get_type (void); + +gboolean gst_wavpack_enc_plugin_init (GstPlugin * plugin); + +G_END_DECLS +#endif /* __GST_WAVPACK_ENC_H__ */ diff --git a/ext/wavpack/gstwavpackstreamreader.c b/ext/wavpack/gstwavpackstreamreader.c new file mode 100755 index 0000000..dd29a37 --- /dev/null +++ b/ext/wavpack/gstwavpackstreamreader.c @@ -0,0 +1,125 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackstreamreader.c: stream reader used for decoding + * + * 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. + */ + +#include <string.h> +#include <math.h> +#include <gst/gst.h> + +#include "gstwavpackstreamreader.h" + +GST_DEBUG_CATEGORY_EXTERN (wavpack_debug); +#define GST_CAT_DEFAULT wavpack_debug + +static int32_t +gst_wavpack_stream_reader_read_bytes (void *id, void *data, int32_t bcount) +{ + read_id *rid = (read_id *) id; + uint32_t left = rid->length - rid->position; + uint32_t to_read = MIN (left, bcount); + + GST_DEBUG ("Trying to read %d of %d bytes from position %d", bcount, + rid->length, rid->position); + + if (to_read > 0) { + memmove (data, rid->buffer + rid->position, to_read); + rid->position += to_read; + return to_read; + } else { + GST_WARNING ("Couldn't read %d bytes", bcount); + return 0; + } +} + +static uint32_t +gst_wavpack_stream_reader_get_pos (void *id) +{ + GST_DEBUG ("Returning position %d", ((read_id *) id)->position); + return ((read_id *) id)->position; +} + +static int +gst_wavpack_stream_reader_set_pos_abs (void *id, uint32_t pos) +{ + GST_WARNING ("Should not be called: tried to set absolute position to %d", + pos); + return -1; +} + +static int +gst_wavpack_stream_reader_set_pos_rel (void *id, int32_t delta, int mode) +{ + GST_WARNING ("Should not be called: tried to set relative position to %d" + " with mode %d", delta, mode); + return -1; +} + +static int +gst_wavpack_stream_reader_push_back_byte (void *id, int c) +{ + read_id *rid = (read_id *) id; + + GST_DEBUG ("Pushing back one byte: 0x%x", c); + + if (rid->position == 0) + return rid->position; + + rid->position -= 1; + return rid->position; +} + +static uint32_t +gst_wavpack_stream_reader_get_length (void *id) +{ + GST_DEBUG ("Returning length %d", ((read_id *) id)->length); + + return ((read_id *) id)->length; +} + +static int +gst_wavpack_stream_reader_can_seek (void *id) +{ + GST_DEBUG ("Can't seek"); + return FALSE; +} + +static int32_t +gst_wavpack_stream_reader_write_bytes (void *id, void *data, int32_t bcount) +{ + GST_WARNING ("Should not be called, tried to write %d bytes", bcount); + return 0; +} + +WavpackStreamReader * +gst_wavpack_stream_reader_new (void) +{ + WavpackStreamReader *stream_reader = + (WavpackStreamReader *) g_malloc0 (sizeof (WavpackStreamReader)); + stream_reader->read_bytes = gst_wavpack_stream_reader_read_bytes; + stream_reader->get_pos = gst_wavpack_stream_reader_get_pos; + stream_reader->set_pos_abs = gst_wavpack_stream_reader_set_pos_abs; + stream_reader->set_pos_rel = gst_wavpack_stream_reader_set_pos_rel; + stream_reader->push_back_byte = gst_wavpack_stream_reader_push_back_byte; + stream_reader->get_length = gst_wavpack_stream_reader_get_length; + stream_reader->can_seek = gst_wavpack_stream_reader_can_seek; + stream_reader->write_bytes = gst_wavpack_stream_reader_write_bytes; + + return stream_reader; +} diff --git a/ext/wavpack/gstwavpackstreamreader.h b/ext/wavpack/gstwavpackstreamreader.h new file mode 100644 index 0000000..17412cc --- /dev/null +++ b/ext/wavpack/gstwavpackstreamreader.h @@ -0,0 +1,36 @@ +/* GStreamer Wavpack plugin + * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org> + * + * gstwavpackstreamreader.h: stream reader used for decoding + * + * 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. + */ + +#ifndef __GST_WAVPACK_STREAM_READER_H__ +#define __GST_WAVPACK_STREAM_READER_H__ + +#include <wavpack/wavpack.h> + +typedef struct +{ + guint8 *buffer; + uint32_t length; + uint32_t position; +} read_id; + +WavpackStreamReader *gst_wavpack_stream_reader_new (void); + +#endif |