/* 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. * |[ * 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 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_details_simple (element_class, "Wavpack audio encoder", "Codec/Encoder/Audio", "Encodes audio with the Wavpack lossless/lossy audio codec", "Sebastian Dröge "); /* 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->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, &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_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), 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_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; /* 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 %d bytes", gst_buffer_get_size (buffer)); *flow = gst_pad_push (pad, buffer); } else { GST_DEBUG_OBJECT (enc, "handing frame of %d 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; 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 */ 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)->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; }