/* 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 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits). * |[ * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv * ]| This pipeline encodes audio from an audio CD into a Wavpack file using * lossless encoding (the file output will be fairly large). * |[ * gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv * ]| This pipeline encodes audio from an audio CD into a Wavpack file using * lossy encoding at a certain bitrate (the file will be fairly small). * */ /* * TODO: - add 32 bit float mode. CONFIG_FLOAT_DATA */ #include #include #include #include #include "gstwavpackenc.h" #include "gstwavpackcommon.h" static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps); static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count); static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event); static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition); static void gst_wavpack_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); enum { ARG_0, ARG_MODE, ARG_BITRATE, ARG_BITSPERSAMPLE, ARG_CORRECTION_MODE, ARG_MD5, ARG_EXTRA_PROCESSING, ARG_JOINT_STEREO_MODE }; GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug); #define GST_CAT_DEFAULT gst_wavpack_enc_debug static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "width = (int) 32, " "depth = (int) [ 1, 32], " "endianness = (int) BYTE_ORDER, " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE") ); static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " "width = (int) [ 1, 32 ], " "channels = (int) [ 1, 2 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE") ); static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE") ); enum { GST_WAVPACK_ENC_MODE_VERY_FAST = 0, GST_WAVPACK_ENC_MODE_FAST, GST_WAVPACK_ENC_MODE_DEFAULT, GST_WAVPACK_ENC_MODE_HIGH, GST_WAVPACK_ENC_MODE_VERY_HIGH }; #define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ()) static GType gst_wavpack_enc_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { #if 0 /* Very Fast Compression is not supported yet, but will be supported * in future wavpack versions */ {GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"}, #endif {GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"}, {GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"}, {GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"}, #ifndef WAVPACK_OLD_API {GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"}, #endif {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncMode", values); } return qtype; } enum { GST_WAVPACK_CORRECTION_MODE_OFF = 0, GST_WAVPACK_CORRECTION_MODE_ON, GST_WAVPACK_CORRECTION_MODE_OPTIMIZED }; #define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ()) static GType gst_wavpack_enc_correction_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"}, {GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"}, {GST_WAVPACK_CORRECTION_MODE_OPTIMIZED, "Create optimized correction file", "optimized"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values); } return qtype; } enum { GST_WAVPACK_JS_MODE_AUTO = 0, GST_WAVPACK_JS_MODE_LEFT_RIGHT, GST_WAVPACK_JS_MODE_MID_SIDE }; #define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ()) static GType gst_wavpack_enc_joint_stereo_mode_get_type (void) { static GType qtype = 0; if (qtype == 0) { static const GEnumValue values[] = { {GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"}, {GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"}, {GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"}, {0, NULL, NULL} }; qtype = g_enum_register_static ("GstWavpackEncJSMode", values); } return qtype; } static void _do_init (GType object_type) { const GInterfaceInfo preset_interface_info = { NULL, /* interface_init */ NULL, /* interface_finalize */ NULL /* interface_data */ }; g_type_add_interface_static (object_type, GST_TYPE_PRESET, &preset_interface_info); } GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT, _do_init); static void gst_wavpack_enc_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); /* add pad templates */ gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&wvcsrc_factory)); /* set element details */ gst_element_class_set_details_simple (element_class, "Wavpack audio encoder", "Codec/Encoder/Audio", "Encodes audio with the Wavpack lossless/lossy audio codec", "Sebastian Dröge "); } static void gst_wavpack_enc_class_init (GstWavpackEncClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); /* set state change handler */ gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state); /* set property handlers */ gobject_class->set_property = GST_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, GST_WAVPACK_ENC_MODE_DEFAULT, G_PARAM_READWRITE)); 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_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_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_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_uint ("extra-processing", "Extra processing", "Use better but slower filters for better compression/quality.", 0, 6, 0, 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, GST_WAVPACK_JS_MODE_AUTO, G_PARAM_READWRITE)); } static void gst_wavpack_enc_reset (GstWavpackEnc * enc) { /* close and free everything stream related if we already did something */ if (enc->wp_context) { WavpackCloseFile (enc->wp_context); enc->wp_context = NULL; } if (enc->wp_config) { g_free (enc->wp_config); enc->wp_config = NULL; } if (enc->first_block) { g_free (enc->first_block); enc->first_block = NULL; } enc->first_block_size = 0; if (enc->md5_context) { g_checksum_free (enc->md5_context); enc->md5_context = NULL; } if (enc->pending_buffer) { gst_buffer_unref (enc->pending_buffer); enc->pending_buffer = NULL; enc->pending_offset = 0; } /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; /* reset stream information */ enc->samplerate = 0; enc->depth = 0; enc->channels = 0; enc->channel_mask = 0; enc->need_channel_remap = FALSE; enc->timestamp_offset = GST_CLOCK_TIME_NONE; enc->next_ts = GST_CLOCK_TIME_NONE; } static void gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) { enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); gst_pad_set_setcaps_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps)); gst_pad_set_chain_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain)); gst_pad_set_event_function (enc->sinkpad, GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event)); gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); /* setup src pad */ enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); /* initialize object attributes */ enc->wp_config = NULL; enc->wp_context = NULL; enc->first_block = NULL; enc->md5_context = NULL; gst_wavpack_enc_reset (enc); enc->wv_id.correction = FALSE; enc->wv_id.wavpack_enc = enc; enc->wv_id.passthrough = FALSE; enc->wvc_id.correction = TRUE; enc->wvc_id.wavpack_enc = enc; enc->wvc_id.passthrough = FALSE; /* set default values of params */ enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT; enc->bitrate = 0; enc->bps = 0.0; enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF; enc->md5 = FALSE; enc->extra_processing = 0; enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO; } static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); GstStructure *structure = gst_caps_get_structure (caps, 0); GstAudioChannelPosition *pos; if (!gst_structure_get_int (structure, "channels", &enc->channels) || !gst_structure_get_int (structure, "rate", &enc->samplerate) || !gst_structure_get_int (structure, "depth", &enc->depth)) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("got invalid caps: %" GST_PTR_FORMAT, caps)); gst_object_unref (enc); return FALSE; } pos = gst_audio_get_channel_positions (structure); /* If one channel is NONE they'll be all undefined */ if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) { g_free (pos); pos = NULL; } if (pos == NULL) { GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL), ("input has no valid channel layout")); gst_object_unref (enc); return FALSE; } enc->channel_mask = gst_wavpack_get_channel_mask_from_positions (pos, enc->channels); enc->need_channel_remap = gst_wavpack_set_channel_mapping (pos, enc->channels, enc->channel_mapping); g_free (pos); /* set fixed src pad caps now that we know what we will get */ caps = gst_caps_new_simple ("audio/x-wavpack", "channels", G_TYPE_INT, enc->channels, "rate", G_TYPE_INT, enc->samplerate, "width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL); if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask)) GST_WARNING_OBJECT (enc, "setting channel layout failed"); if (!gst_pad_set_caps (enc->srcpad, caps)) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("setting caps failed: %" GST_PTR_FORMAT, caps)); gst_caps_unref (caps); gst_object_unref (enc); return FALSE; } gst_pad_use_fixed_caps (enc->srcpad); gst_caps_unref (caps); gst_object_unref (enc); return TRUE; } static void gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc) { enc->wp_config = g_new0 (WavpackConfig, 1); /* set general stream informations in the WavpackConfig */ enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8; enc->wp_config->bits_per_sample = enc->depth; enc->wp_config->num_channels = enc->channels; enc->wp_config->channel_mask = enc->channel_mask; enc->wp_config->sample_rate = enc->samplerate; /* * Set parameters in WavpackConfig */ /* Encoding mode */ switch (enc->mode) { #if 0 case GST_WAVPACK_ENC_MODE_VERY_FAST: enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG; enc->wp_config->flags |= CONFIG_FAST_FLAG; break; #endif case GST_WAVPACK_ENC_MODE_FAST: enc->wp_config->flags |= CONFIG_FAST_FLAG; break; case GST_WAVPACK_ENC_MODE_DEFAULT: break; case GST_WAVPACK_ENC_MODE_HIGH: enc->wp_config->flags |= CONFIG_HIGH_FLAG; break; #ifndef WAVPACK_OLD_API case GST_WAVPACK_ENC_MODE_VERY_HIGH: enc->wp_config->flags |= CONFIG_HIGH_FLAG; enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG; break; #endif } /* Bitrate, enables lossy mode */ if (enc->bitrate) { enc->wp_config->flags |= CONFIG_HYBRID_FLAG; enc->wp_config->flags |= CONFIG_BITRATE_KBPS; enc->wp_config->bitrate = enc->bitrate / 1000.0; } else if (enc->bps) { enc->wp_config->flags |= CONFIG_HYBRID_FLAG; enc->wp_config->bitrate = enc->bps; } /* Correction Mode, only in lossy mode */ if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) { if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) { GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction", "framed", G_TYPE_BOOLEAN, TRUE, NULL); enc->wvcsrcpad = gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc"); /* try to add correction src pad, don't set correction mode on failure */ GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %" GST_PTR_FORMAT, caps); if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) { enc->correction_mode = 0; GST_WARNING_OBJECT (enc, "setting correction caps failed"); } else { gst_pad_use_fixed_caps (enc->wvcsrcpad); gst_pad_set_active (enc->wvcsrcpad, TRUE); gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad); enc->wp_config->flags |= CONFIG_CREATE_WVC; if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) { enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC; } } gst_caps_unref (caps); } } else { if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) { enc->correction_mode = 0; GST_WARNING_OBJECT (enc, "setting correction mode only has " "any effect if a bitrate is provided."); } } gst_element_no_more_pads (GST_ELEMENT (enc)); /* MD5, setup MD5 context */ if ((enc->md5) && !(enc->md5_context)) { enc->wp_config->flags |= CONFIG_MD5_CHECKSUM; enc->md5_context = g_checksum_new (G_CHECKSUM_MD5); } /* Extra encode processing */ if (enc->extra_processing) { enc->wp_config->flags |= CONFIG_EXTRA_MODE; enc->wp_config->xmode = enc->extra_processing; } /* Joint stereo mode */ switch (enc->joint_stereo_mode) { case GST_WAVPACK_JS_MODE_AUTO: break; case GST_WAVPACK_JS_MODE_LEFT_RIGHT: enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE; enc->wp_config->flags &= ~CONFIG_JOINT_STEREO; break; case GST_WAVPACK_JS_MODE_MID_SIDE: enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO); break; } } static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count) { GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id; GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc); GstFlowReturn *flow; GstBuffer *buffer; GstPad *pad; guchar *block = (guchar *) data; pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad; flow = (wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> srcpad_last_return; *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (pad), &buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } g_memmove (GST_BUFFER_DATA (buffer), block, count); if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { /* if it's a Wavpack block set buffer timestamp and duration, etc */ WavpackHeader wph; GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata", count, (wid->correction) ? "correction " : ""); gst_wavpack_read_header (&wph, block); /* Only set when pushing the first buffer again, in that case * we don't want to delay the buffer or push newsegment events */ if (!wid->passthrough) { /* Only push complete blocks */ if (enc->pending_buffer == NULL) { enc->pending_buffer = buffer; enc->pending_offset = wph.block_index; } else if (enc->pending_offset == wph.block_index) { enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer); } else { GST_ERROR ("Got incomplete block, dropping"); gst_buffer_unref (enc->pending_buffer); enc->pending_buffer = buffer; enc->pending_offset = wph.block_index; } if (!(wph.flags & FINAL_BLOCK)) return TRUE; buffer = enc->pending_buffer; enc->pending_buffer = NULL; enc->pending_offset = 0; /* if it's the first wavpack block, send a NEW_SEGMENT event */ if (wph.block_index == 0) { gst_pad_push_event (pad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0)); /* save header for later reference, so we can re-send it later on * EOS with fixed up values for total sample count etc. */ if (enc->first_block == NULL && !wid->correction) { enc->first_block = g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); enc->first_block_size = GST_BUFFER_SIZE (buffer); } } } /* set buffer timestamp, duration, offset, offset_end from * the wavpack header */ GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset + gst_util_uint64_scale_int (GST_SECOND, wph.block_index, enc->samplerate); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, enc->samplerate); GST_BUFFER_OFFSET (buffer) = wph.block_index; GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples; } else { /* if it's something else set no timestamp and duration on the buffer */ GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count); GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; } /* push the buffer and forward errors */ GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", GST_BUFFER_SIZE (buffer)); *flow = gst_pad_push (pad, buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } return TRUE; } static void gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data, gint nsamples) { gint i, j; gint32 tmp[8]; for (i = 0; i < nsamples / enc->channels; i++) { for (j = 0; j < enc->channels; j++) { tmp[enc->channel_mapping[j]] = data[j]; } for (j = 0; j < enc->channels; j++) { data[j] = tmp[j]; } data += enc->channels; } } static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4; GstFlowReturn ret; /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; GST_DEBUG ("got %u raw samples", sample_count); /* check if we already have a valid WavpackContext, otherwise make one */ if (!enc->wp_context) { /* create raw context */ enc->wp_context = WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id, (enc->correction_mode > 0) ? &enc->wvc_id : NULL); if (!enc->wp_context) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("error creating Wavpack context")); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } /* set the WavpackConfig according to our parameters */ gst_wavpack_enc_set_wp_config (enc); /* set the configuration to the context now that we know everything * and initialize the encoder */ if (!WavpackSetConfiguration (enc->wp_context, enc->wp_config, (uint32_t) (-1)) || !WavpackPackInit (enc->wp_context)) { GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), ("error setting up wavpack encoding context")); WavpackCloseFile (enc->wp_context); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } GST_DEBUG ("setup of encoding context successfull"); } /* Save the timestamp of the first buffer. This will be later * used as offset for all following buffers */ if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) { if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf); enc->next_ts = GST_BUFFER_TIMESTAMP (buf); } else { enc->timestamp_offset = 0; enc->next_ts = 0; } } /* Check if we have a continous stream, if not drop some samples or the buffer or * insert some silence samples */ if (enc->next_ts != GST_CLOCK_TIME_NONE && GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) { guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf); guint64 diff_bytes; GST_WARNING_OBJECT (enc, "Buffer is older than previous " "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT "), cannot handle. Clipping buffer.", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (enc->next_ts)); diff_bytes = GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2; if (diff_bytes >= GST_BUFFER_SIZE (buf)) { gst_buffer_unref (buf); return GST_FLOW_OK; } buf = gst_buffer_make_metadata_writable (buf); GST_BUFFER_DATA (buf) += diff_bytes; GST_BUFFER_SIZE (buf) -= diff_bytes; GST_BUFFER_TIMESTAMP (buf) += diff; if (GST_BUFFER_DURATION_IS_VALID (buf)) GST_BUFFER_DURATION (buf) -= diff; } /* Allow a diff of at most 5 ms */ if (enc->next_ts != GST_CLOCK_TIME_NONE && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts && GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) { GST_WARNING_OBJECT (enc, "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND); WavpackFlushSamples (enc->wp_context); enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts); } } if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) && GST_BUFFER_DURATION_IS_VALID (buf)) enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); else enc->next_ts = GST_CLOCK_TIME_NONE; if (enc->need_channel_remap) { buf = gst_buffer_make_writable (buf); gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf), sample_count); } /* if we want to append the MD5 sum to the stream update it here * with the current raw samples */ if (enc->md5) { g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); } /* encode and handle return values from encoding */ if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf), sample_count / enc->channels)) { GST_DEBUG ("encoding samples successful"); ret = GST_FLOW_OK; } else { if ((enc->srcpad_last_return == GST_FLOW_RESEND) || (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { ret = GST_FLOW_RESEND; } else if ((enc->srcpad_last_return == GST_FLOW_OK) || (enc->wvcsrcpad_last_return == GST_FLOW_OK)) { ret = GST_FLOW_OK; } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && (enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) { ret = GST_FLOW_NOT_LINKED; } else if ((enc->srcpad_last_return == GST_FLOW_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; } } gst_buffer_unref (buf); gst_object_unref (enc); return ret; } static void gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) { GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0); gboolean ret; g_return_if_fail (enc); g_return_if_fail (enc->first_block); /* update the sample count in the first block */ WavpackUpdateNumSamples (enc->wp_context, enc->first_block); /* try to seek to the beginning of the output */ ret = gst_pad_push_event (enc->srcpad, event); if (ret) { /* try to rewrite the first block */ GST_DEBUG_OBJECT (enc, "rewriting first block ..."); enc->wv_id.passthrough = TRUE; ret = gst_wavpack_enc_push_block (&enc->wv_id, enc->first_block, enc->first_block_size); enc->wv_id.passthrough = FALSE; } else { GST_WARNING_OBJECT (enc, "rewriting of first block failed. " "Seeking to first block failed!"); } } static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); gboolean ret = TRUE; GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* Encode all remaining samples and flush them to the src pads */ WavpackFlushSamples (enc->wp_context); /* Drop all remaining data, this is no complete block otherwise * it would've been pushed already */ if (enc->pending_buffer) { gst_object_unref (enc->pending_buffer); enc->pending_buffer = NULL; enc->pending_offset = 0; } /* write the MD5 sum if we have to write one */ if ((enc->md5) && (enc->md5_context)) { guint8 md5_digest[16]; gsize digest_len = sizeof (md5_digest); g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len); if (digest_len == sizeof (md5_digest)) WavpackStoreMD5Sum (enc->wp_context, md5_digest); else GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed"); } /* Try to rewrite the first frame with the correct sample number */ if (enc->first_block) gst_wavpack_enc_rewrite_first_block (enc); /* close the context if not already happened */ if (enc->wp_context) { WavpackCloseFile (enc->wp_context); enc->wp_context = NULL; } ret = gst_pad_event_default (pad, event); break; case GST_EVENT_NEWSEGMENT: if (enc->wp_context) { GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding " "already started"); } /* drop NEWSEGMENT events, we create our own when pushing * the first buffer to the pads */ gst_event_unref (event); ret = TRUE; break; default: ret = gst_pad_event_default (pad, event); break; } gst_object_unref (enc); return ret; } static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstWavpackEnc *enc = GST_WAVPACK_ENC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK * as they're only set to something else in WavpackPackSamples() or more * specific gst_wavpack_enc_push_block() and nothing happened there yet */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; break; case GST_STATE_CHANGE_READY_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_wavpack_enc_reset (enc); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static void gst_wavpack_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWavpackEnc *enc = GST_WAVPACK_ENC (object); switch (prop_id) { case ARG_MODE: enc->mode = g_value_get_enum (value); break; case ARG_BITRATE:{ guint val = g_value_get_uint (value); if ((val >= 24000) && (val <= 9600000)) { enc->bitrate = val; enc->bps = 0.0; } else { enc->bitrate = 0; enc->bps = 0.0; } break; } case ARG_BITSPERSAMPLE:{ gdouble val = g_value_get_double (value); if ((val >= 2.0) && (val <= 24.0)) { enc->bps = val; enc->bitrate = 0; } else { enc->bps = 0.0; enc->bitrate = 0; } break; } case ARG_CORRECTION_MODE: enc->correction_mode = g_value_get_enum (value); break; case ARG_MD5: enc->md5 = g_value_get_boolean (value); break; case ARG_EXTRA_PROCESSING: enc->extra_processing = g_value_get_uint (value); break; case ARG_JOINT_STEREO_MODE: enc->joint_stereo_mode = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWavpackEnc *enc = GST_WAVPACK_ENC (object); switch (prop_id) { case ARG_MODE: g_value_set_enum (value, enc->mode); break; case ARG_BITRATE: if (enc->bps == 0.0) { g_value_set_uint (value, enc->bitrate); } else { g_value_set_uint (value, 0); } break; case ARG_BITSPERSAMPLE: if (enc->bitrate == 0) { g_value_set_double (value, enc->bps); } else { g_value_set_double (value, 0.0); } break; case ARG_CORRECTION_MODE: g_value_set_enum (value, enc->correction_mode); break; case ARG_MD5: g_value_set_boolean (value, enc->md5); break; case ARG_EXTRA_PROCESSING: g_value_set_uint (value, enc->extra_processing); break; case ARG_JOINT_STEREO_MODE: g_value_set_enum (value, enc->joint_stereo_mode); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } gboolean gst_wavpack_enc_plugin_init (GstPlugin * plugin) { if (!gst_element_register (plugin, "wavpackenc", GST_RANK_NONE, GST_TYPE_WAVPACK_ENC)) return FALSE; GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0, "Wavpack encoder"); return TRUE; }