/* GStreamer Wavpack encoder plugin * Copyright (c) 2006 Sebastian Dröge * * gstwavpackdec.c: Wavpack audio encoder * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-wavpackenc * * * Wavpackenc encodes raw audio into a framed Wavpack stream. * Wavpack is an open-source * audio codec that features both lossless and lossy encoding. * Example launch line * * * gst-launch audiotestsrc num-buffers=500 ! wavpackenc ! filesink location=sinewave.wv * * This pipeline encodes audio from audiotestsrc into a Wavpack file. * * * * gst-launch cdda://1 ! wavpackenc ! filesink location=track1.wv * * This pipeline encodes audio from an audio CD into a Wavpack file using * lossless encoding (the file output will be fairly large). * * * * gst-launch cdda://1 ! 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 multichannel handling. channel_mask is: * front left * front right * center * LFE * back left * back right * front left center * front right center * back left * back center * side left * side right * ... * - add 32 bit float mode. CONFIG_FLOAT_DATA */ #include #include #include #include #include "gstwavpackenc.h" #include "gstwavpackcommon.h" #include "md5.h" static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps); static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count); static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event); static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition); static void gst_wavpack_enc_finalize (GObject * object); static void gst_wavpack_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); enum { ARG_0, ARG_MODE, ARG_BITRATE, ARG_BITSPERSAMPLE, ARG_CORRECTION_MODE, ARG_MD5, ARG_EXTRA_PROCESSING, ARG_JOINT_STEREO_MODE }; GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug); #define GST_CAT_DEFAULT gst_wavpack_enc_debug static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "width = (int) 32, " "depth = (int) 32, " "endianness = (int) LITTLE_ENDIAN, " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;" "audio/x-raw-int, " "width = (int) 24, " "depth = (int) 24, " "endianness = (int) LITTLE_ENDIAN, " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;" "audio/x-raw-int, " "width = (int) 16, " "depth = (int) 16, " "endianness = (int) LITTLE_ENDIAN, " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;" "audio/x-raw-int, " "width = (int) 8, " "depth = (int) 8, " "endianness = (int) LITTLE_ENDIAN, " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE") ); static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " "width = (int) { 8, 16, 24, 32 }, " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) 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") ); #define DEFAULT_MODE 1 #define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ()) static GType gst_wavpack_enc_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {0, "Fast Compression", "0"}, {1, "Default", "1"}, {2, "High Compression", "2"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncMode", values); } return qtype; } #define DEFAULT_CORRECTION_MODE 0 #define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ()) static GType gst_wavpack_enc_correction_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {0, "Create no correction file (default)", "0"}, {1, "Create correction file", "1"}, {2, "Create optimized correction file", "2"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values); } return qtype; } #define DEFAULT_JS_MODE 0 #define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ()) static GType gst_wavpack_enc_joint_stereo_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {0, "auto (default)", "0"}, {1, "left/right", "1"}, {2, "mid/side", "2"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncJSMode", values); } return qtype; } GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT); static void gst_wavpack_enc_base_init (gpointer klass) { static GstElementDetails element_details = { "Wavpack audio encoder", "Codec/Encoder/Audio", "Encodes audio with the Wavpack lossless/lossy audio codec", "Sebastian Dröge " }; GstElementClass *element_class = GST_ELEMENT_CLASS (klass); /* add pad templates */ gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&wvcsrc_factory)); /* set element details */ gst_element_class_set_details (element_class, &element_details); } static void gst_wavpack_enc_class_init (GstWavpackEncClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); /* set state change handler */ gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_enc_finalize); /* set property handlers */ gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_wavpack_enc_get_property); /* install all properties */ g_object_class_install_property (gobject_class, ARG_MODE, g_param_spec_enum ("mode", "Encoding mode", "Speed versus compression tradeoff.", GST_TYPE_WAVPACK_ENC_MODE, DEFAULT_MODE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_BITRATE, g_param_spec_double ("bitrate", "Bitrate", "Try to encode with this average bitrate (bits/sec). " "This enables lossy encoding! A value smaller than 24000.0 disables this.", 0.0, 9600000.0, 0.0, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE, g_param_spec_double ("bits-per-sample", "Bits per sample", "Try to encode with this amount of bits per sample. " "This enables lossy encoding! A value smaller than 2.0 disables this.", 0.0, 24.0, 0.0, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE, g_param_spec_enum ("correction-mode", "Correction file mode", "Use this mode for correction file creation. Only works in lossy mode!", GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, DEFAULT_CORRECTION_MODE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_MD5, g_param_spec_boolean ("md5", "MD5", "Store MD5 hash of raw samples within the file.", FALSE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING, g_param_spec_boolean ("extra-processing", "Extra processing", "Extra encode processing.", FALSE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE, g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode", "Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE, DEFAULT_JS_MODE, G_PARAM_READWRITE)); } static void gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) { enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); gst_pad_set_setcaps_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps)); gst_pad_set_chain_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain)); gst_pad_set_event_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event)); gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); /* setup src pad */ enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); /* initialize object attributes */ enc->wp_config = NULL; enc->wp_context = NULL; enc->first_block = NULL; enc->first_block_size = 0; enc->md5_context = NULL; enc->samplerate = 0; enc->width = 0; enc->channels = 0; enc->wv_id = g_new0 (GstWavpackEncWriteID, 1); enc->wv_id->correction = FALSE; enc->wv_id->wavpack_enc = enc; enc->wvc_id = g_new0 (GstWavpackEncWriteID, 1); enc->wvc_id->correction = TRUE; enc->wvc_id->wavpack_enc = enc; /* set default values of params */ enc->mode = 1; enc->bitrate = 0.0; enc->correction_mode = 0; enc->md5 = FALSE; enc->extra_processing = FALSE; enc->joint_stereo_mode = 0; } static void gst_wavpack_enc_finalize (GObject * object) { GstWavpackEnc *enc = GST_WAVPACK_ENC (object); /* free the blockout helpers */ g_free (enc->wv_id); g_free (enc->wvc_id); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); GstStructure *structure = gst_caps_get_structure (caps, 0); int depth = 0; /* check caps and put relevant parts into our object attributes */ if (!gst_structure_get_int (structure, "channels", &enc->channels) || !gst_structure_get_int (structure, "rate", &enc->samplerate) || !gst_structure_get_int (structure, "width", &enc->width) || !(gst_structure_get_int (structure, "depth", &depth) || depth != enc->width)) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("got invalid caps: %", GST_PTR_FORMAT, caps)); gst_object_unref (enc); return FALSE; } /* set fixed src pad caps now that we know what we will get */ caps = gst_caps_new_simple ("audio/x-wavpack", "channels", G_TYPE_INT, enc->channels, "rate", G_TYPE_INT, enc->samplerate, "width", G_TYPE_INT, enc->width, "framed", G_TYPE_BOOLEAN, TRUE, NULL); if (!gst_pad_set_caps (enc->srcpad, caps)) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("setting caps failed: %", GST_PTR_FORMAT, caps)); gst_caps_unref (caps); gst_object_unref (enc); return FALSE; } gst_pad_use_fixed_caps (enc->srcpad); gst_caps_unref (caps); gst_object_unref (enc); return TRUE; } static void gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc) { enc->wp_config = g_new0 (WavpackConfig, 1); /* set general stream informations in the WavpackConfig */ enc->wp_config->bytes_per_sample = (enc->width + 7) >> 3; enc->wp_config->bits_per_sample = enc->width; enc->wp_config->num_channels = enc->channels; /* TODO: handle more than 2 channels correctly! */ if (enc->channels == 1) { enc->wp_config->channel_mask = 0x4; } else if (enc->channels == 2) { enc->wp_config->channel_mask = 0x2 | 0x1; } enc->wp_config->sample_rate = enc->samplerate; /* * Set parameters in WavpackConfig */ /* Encoding mode */ switch (enc->mode) { case 0: enc->wp_config->flags |= CONFIG_FAST_FLAG; break; case 1: /* default */ break; case 2: enc->wp_config->flags |= CONFIG_HIGH_FLAG; break; } /* Bitrate, enables lossy mode */ if (enc->bitrate >= 2.0) { enc->wp_config->flags |= CONFIG_HYBRID_FLAG; if (enc->bitrate >= 24000.0) { enc->wp_config->bitrate = enc->bitrate / 1000.0; enc->wp_config->flags |= CONFIG_BITRATE_KBPS; } else { enc->wp_config->bitrate = enc->bitrate; } } /* Correction Mode, only in lossy mode */ if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) { if (enc->correction_mode > 0) { enc->wvcsrcpad = gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc"); /* try to add correction src pad, don't set correction mode on failure */ GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction", "framed", G_TYPE_BOOLEAN, TRUE, NULL); gst_element_no_more_pads (GST_ELEMENT (enc)); 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_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad); enc->wp_config->flags |= CONFIG_CREATE_WVC; if (enc->correction_mode == 2) { enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC; } } gst_caps_unref (caps); } } else { if (enc->correction_mode > 0) { 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_new0 (MD5_CTX, 1); MD5Init (enc->md5_context); } /* Extra encode processing */ if (enc->extra_processing) { enc->wp_config->flags |= CONFIG_EXTRA_MODE; } /* Joint stereo mode */ switch (enc->joint_stereo_mode) { case 0: /* default */ break; case 1: enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE; enc->wp_config->flags &= ~CONFIG_JOINT_STEREO; break; case 2: enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO); break; } } static int32_t * gst_wavpack_enc_format_samples (const uchar * src_data, uint32_t sample_count, guint width) { int32_t *data = g_new0 (int32_t, sample_count); /* put all samples into an int32_t*, no matter what * width we have and convert them from little endian * to host byte order */ switch (width) { int i; case 8: for (i = 0; i < sample_count; i++) data[i] = (int32_t) (int8_t) src_data[i]; break; case 16: for (i = 0; i < sample_count; i++) data[i] = (int32_t) src_data[2 * i] | ((int32_t) (int8_t) src_data[2 * i + 1] << 8); break; case 24: for (i = 0; i < sample_count; i++) data[i] = (int32_t) src_data[3 * i] | ((int32_t) src_data[3 * i + 1] << 8) | ((int32_t) (int8_t) src_data[3 * i + 2] << 16); break; case 32: for (i = 0; i < sample_count; i++) data[i] = (int32_t) src_data[4 * i] | ((int32_t) src_data[4 * i + 1] << 8) | ((int32_t) src_data[4 * i + 2] << 16) | ((int32_t) (int8_t) src_data[4 * i + 3] << 24); break; } return data; } static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count) { GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id; GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc); GstFlowReturn *flow; GstBuffer *buffer; GstPad *pad; guchar *block = (guchar *) data; pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad; flow = (wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> srcpad_last_return; *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (pad), &buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } g_memmove (GST_BUFFER_DATA (buffer), block, count); if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { /* if it's a Wavpack block set buffer timestamp and duration, etc */ WavpackHeader wph; GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata", count, (wid->correction) ? "correction " : ""); gst_wavpack_read_header (&wph, block); /* if it's the first wavpack block, send a NEW_SEGMENT event */ if (wph.block_index == 0) { gst_pad_push_event (pad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0)); /* save header for later reference, so we can re-send it later on * EOS with fixed up values for total sample count etc. */ if (enc->first_block == NULL && !wid->correction) { enc->first_block = g_memdup (block, count); enc->first_block_size = count; } } /* set buffer timestamp, duration, offset, offset_end from * the wavpack header */ GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale_int (GST_SECOND, wph.block_index, enc->samplerate); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, enc->samplerate); GST_BUFFER_OFFSET (buffer) = wph.block_index; GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples; } else { /* if it's something else set no timestamp and duration on the buffer */ GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count); GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; } /* push the buffer and forward errors */ *flow = gst_pad_push (pad, buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } return TRUE; } static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); uint32_t sample_count = GST_BUFFER_SIZE (buf) / ((enc->width + 7) >> 3); int32_t *data; GstFlowReturn ret; /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; GST_DEBUG ("got %u raw samples", sample_count); /* check if we already have a valid WavpackContext, otherwise make one */ if (!enc->wp_context) { /* create raw context */ enc->wp_context = WavpackOpenFileOutput (gst_wavpack_enc_push_block, enc->wv_id, (enc->correction_mode > 0) ? enc->wvc_id : NULL); if (!enc->wp_context) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("error creating Wavpack context")); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } /* set the WavpackConfig according to our parameters */ gst_wavpack_enc_set_wp_config (enc); /* set the configuration to the context now that we know everything * and initialize the encoder */ if (!WavpackSetConfiguration (enc->wp_context, enc->wp_config, (uint32_t) (-1)) || !WavpackPackInit (enc->wp_context)) { GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), ("error setting up wavpack encoding context")); WavpackCloseFile (enc->wp_context); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } GST_DEBUG ("setup of encoding context successfull"); } /* if we want to append the MD5 sum to the stream update it here * with the current raw samples */ if (enc->md5) { MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); } /* put all samples into an int32_t*, no matter what * width we have and convert them from little endian * to host byte order */ data = gst_wavpack_enc_format_samples (GST_BUFFER_DATA (buf), sample_count, enc->width); gst_buffer_unref (buf); /* encode and handle return values from encoding */ if (WavpackPackSamples (enc->wp_context, data, sample_count / enc->channels)) { GST_DEBUG ("encoding samples successful"); ret = GST_FLOW_OK; } else { if ((enc->srcpad_last_return == GST_FLOW_RESEND) || (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { ret = GST_FLOW_RESEND; } else if ((enc->srcpad_last_return == GST_FLOW_OK) || (enc->wvcsrcpad_last_return == GST_FLOW_OK)) { ret = GST_FLOW_OK; } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && (enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) { ret = GST_FLOW_NOT_LINKED; } else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) && (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) { ret = GST_FLOW_WRONG_STATE; } else { GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL), ("encoding samples failed")); ret = GST_FLOW_ERROR; } } g_free (data); gst_object_unref (enc); return ret; } static void gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) { GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0); gboolean ret; g_return_if_fail (enc); g_return_if_fail (enc->first_block); /* update the sample count in the first block */ WavpackUpdateNumSamples (enc->wp_context, enc->first_block); /* try to seek to the beginning of the output */ ret = gst_pad_push_event (enc->srcpad, event); if (ret) { /* try to rewrite the first block */ GST_DEBUG_OBJECT (enc, "rewriting first block ..."); ret = gst_wavpack_enc_push_block (enc->wv_id, enc->first_block, enc->first_block_size); } else { GST_WARNING_OBJECT (enc, "rewriting of first block failed. " "Seeking to first block failed!"); } } static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); gboolean ret = TRUE; GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* Encode all remaining samples and flush them to the src pads */ WavpackFlushSamples (enc->wp_context); /* write the MD5 sum if we have to write one */ if ((enc->md5) && (enc->md5_context)) { guchar md5_digest[16]; MD5Final (md5_digest, enc->md5_context); WavpackStoreMD5Sum (enc->wp_context, md5_digest); } /* Try to rewrite the first frame with the correct sample number */ if (enc->first_block) gst_wavpack_enc_rewrite_first_block (enc); /* close the context if not already happened */ if (enc->wp_context) { WavpackCloseFile (enc->wp_context); enc->wp_context = NULL; } ret = gst_pad_event_default (pad, event); break; case GST_EVENT_NEWSEGMENT: if (enc->wp_context) { GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding " "already started"); } /* drop NEWSEGMENT events, we create our own when pushing * the first buffer to the pads */ gst_event_unref (event); ret = TRUE; break; default: ret = gst_pad_event_default (pad, event); break; } gst_object_unref (enc); return ret; } static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstWavpackEnc *enc = GST_WAVPACK_ENC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK * as they're only set to something else in WavpackPackSamples() or more * specific gst_wavpack_enc_push_block() and nothing happened there yet */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; case GST_STATE_CHANGE_READY_TO_PAUSED: case GST_STATE_CHANGE_PAUSED_TO_PLAYING: default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* close and free everything stream related */ if (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_free (enc->md5_context); enc->md5_context = NULL; } /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static void gst_wavpack_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWavpackEnc *enc = GST_WAVPACK_ENC (object); switch (prop_id) { case ARG_MODE: enc->mode = g_value_get_enum (value); break; case ARG_BITRATE:{ gdouble val = g_value_get_double (value); if ((val >= 24000.0) && (val <= 9600000.0)) { enc->bitrate = val; } else { enc->bitrate = 0.0; } break; } case ARG_BITSPERSAMPLE:{ gdouble val = g_value_get_double (value); if ((val >= 2.0) && (val <= 24.0)) { enc->bitrate = val; } else { enc->bitrate = 0.0; } 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_boolean (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->bitrate >= 24000.0) { g_value_set_double (value, enc->bitrate); } else { g_value_set_double (value, 0.0); } break; case ARG_BITSPERSAMPLE: if (enc->bitrate <= 24.0) { g_value_set_double (value, enc->bitrate); } 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_boolean (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; }