From 522f2492c46175f2bfdb623c8a56ddd22b4327af Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sat, 10 Jun 2006 15:33:18 +0000 Subject: ext/wavpack/: Add wavpack encoder element (#343131). MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpack.c: (plugin_init): * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type), (gst_wavpack_enc_correction_mode_get_type), (gst_wavpack_enc_joint_stereo_mode_get_type), (gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_dispose), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_format_samples), (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block), (gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property), (gst_wavpack_enc_plugin_init): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/md5.c: * ext/wavpack/md5.h: Add wavpack encoder element (#343131). --- ext/wavpack/gstwavpackenc.c | 1002 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 1002 insertions(+) create mode 100644 ext/wavpack/gstwavpackenc.c (limited to 'ext/wavpack/gstwavpackenc.c') diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c new file mode 100644 index 00000000..a7d6f748 --- /dev/null +++ b/ext/wavpack/gstwavpackenc.c @@ -0,0 +1,1002 @@ +/* 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. + */ + +/* + * 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_dispose (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_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) FALSE") + ); + +static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) FALSE") + ); + +#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->dispose = GST_DEBUG_FUNCPTR (gst_wavpack_enc_dispose); + + /* 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. " + "This enables lossy encoding! (0 .. 2.0 == disabled, 2.0 .. 23.9 == bits/sample, 24.0 .. 9600 == kbit/second)", + 0.0, 9600.0, 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 * wavpack_enc, GstWavpackEncClass * gclass) +{ + GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpack_enc); + + /* setup sink pad, add handlers */ + wavpack_enc->sinkpad = + gst_pad_new_from_template (gst_element_class_get_pad_template (klass, + "sink"), "sink"); + gst_pad_set_setcaps_function (wavpack_enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps)); + gst_pad_set_chain_function (wavpack_enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain)); + gst_pad_set_event_function (wavpack_enc->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event)); + gst_element_add_pad (GST_ELEMENT (wavpack_enc), + GST_DEBUG_FUNCPTR (wavpack_enc->sinkpad)); + + /* setup src pad */ + wavpack_enc->srcpad = + gst_pad_new_from_template (gst_element_class_get_pad_template (klass, + "src"), "src"); + gst_element_add_pad (GST_ELEMENT (wavpack_enc), + GST_DEBUG_FUNCPTR (wavpack_enc->srcpad)); + + /* initialize object attributes */ + wavpack_enc->wp_config = NULL; + wavpack_enc->wp_context = NULL; + wavpack_enc->first_block = NULL; + wavpack_enc->first_block_size = 0; + wavpack_enc->md5_context = NULL; + wavpack_enc->samplerate = 0; + wavpack_enc->width = 0; + wavpack_enc->channels = 0; + + wavpack_enc->wv_id = (write_id *) g_malloc0 (sizeof (write_id)); + wavpack_enc->wv_id->correction = FALSE; + wavpack_enc->wv_id->wavpack_enc = wavpack_enc; + wavpack_enc->wvc_id = (write_id *) g_malloc0 (sizeof (write_id)); + wavpack_enc->wvc_id->correction = TRUE; + wavpack_enc->wvc_id->wavpack_enc = wavpack_enc; + + /* set default values of params */ + wavpack_enc->mode = 1; + wavpack_enc->bitrate = 0.0; + wavpack_enc->correction_mode = 0; + wavpack_enc->md5 = FALSE; + wavpack_enc->extra_processing = FALSE; + wavpack_enc->joint_stereo_mode = 0; +} + +static void +gst_wavpack_enc_dispose (GObject * object) +{ + GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (object); + + /* free the blockout helpers */ + g_free (wavpack_enc->wv_id); + g_free (wavpack_enc->wvc_id); + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static gboolean +gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) +{ + GstWavpackEnc *wavpack_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", &wavpack_enc->channels)) + || (!gst_structure_get_int (structure, "rate", &wavpack_enc->samplerate)) + || (!gst_structure_get_int (structure, "width", &wavpack_enc->width)) + || (!(gst_structure_get_int (structure, "depth", &depth)) + || depth != wavpack_enc->width)) { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL), + ("got invalid caps: %", GST_PTR_FORMAT, caps)); + gst_object_unref (wavpack_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, wavpack_enc->channels, + "rate", G_TYPE_INT, wavpack_enc->samplerate, + "width", G_TYPE_INT, wavpack_enc->width, + "framed", G_TYPE_BOOLEAN, TRUE, NULL); + + if (!gst_pad_set_caps (wavpack_enc->srcpad, caps)) { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL), + ("setting caps failed: %", GST_PTR_FORMAT, caps)); + gst_caps_unref (caps); + gst_object_unref (wavpack_enc); + return FALSE; + } + gst_pad_use_fixed_caps (wavpack_enc->srcpad); + + gst_caps_unref (caps); + gst_object_unref (wavpack_enc); + return TRUE; +} + +static void +gst_wavpack_enc_set_wp_config (GstWavpackEnc * wavpack_enc) +{ + wavpack_enc->wp_config = (WavpackConfig *) g_malloc0 (sizeof (WavpackConfig)); + /* set general stream informations in the WavpackConfig */ + wavpack_enc->wp_config->bytes_per_sample = (wavpack_enc->width + 7) >> 3; + wavpack_enc->wp_config->bits_per_sample = wavpack_enc->width; + wavpack_enc->wp_config->num_channels = wavpack_enc->channels; + + /* TODO: handle more than 2 channels correctly! */ + if (wavpack_enc->channels == 1) { + wavpack_enc->wp_config->channel_mask = 0x4; + } else if (wavpack_enc->channels == 2) { + wavpack_enc->wp_config->channel_mask = 0x2 | 0x1; + } + wavpack_enc->wp_config->sample_rate = wavpack_enc->samplerate; + + /* + * Set parameters in WavpackConfig + */ + + /* Encoding mode */ + switch (wavpack_enc->mode) { + case 0: + wavpack_enc->wp_config->flags |= CONFIG_FAST_FLAG; + break; + case 1: /* default */ + break; + case 2: + wavpack_enc->wp_config->flags |= CONFIG_HIGH_FLAG; + break; + } + + /* Bitrate, enables lossy mode */ + if (wavpack_enc->bitrate > 2.0) { + wavpack_enc->wp_config->flags |= CONFIG_HYBRID_FLAG; + wavpack_enc->wp_config->bitrate = wavpack_enc->bitrate; + if (wavpack_enc->bitrate >= 24.0) + wavpack_enc->wp_config->flags |= CONFIG_BITRATE_KBPS; + } + + /* Correction Mode, only in lossy mode */ + if (wavpack_enc->wp_config->flags & CONFIG_HYBRID_FLAG) { + if (wavpack_enc->correction_mode > 0) { + wavpack_enc->wvcsrcpad = + gst_pad_new_from_template (gst_element_class_get_pad_template + (GST_ELEMENT_GET_CLASS (wavpack_enc), "wvcsrc"), "wvcsrc"); + + /* try to add correction src pad, don't set correction mode on failure */ + if (gst_element_add_pad (GST_ELEMENT (wavpack_enc), + GST_DEBUG_FUNCPTR (wavpack_enc->wvcsrcpad))) { + GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction", + "framed", G_TYPE_BOOLEAN, FALSE, NULL); + + gst_element_no_more_pads (GST_ELEMENT (wavpack_enc)); + + if (!gst_pad_set_caps (wavpack_enc->wvcsrcpad, caps)) { + wavpack_enc->correction_mode = 0; + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL), + ("setting correction caps failed: %", GST_PTR_FORMAT, caps)); + } else { + gst_pad_use_fixed_caps (wavpack_enc->wvcsrcpad); + wavpack_enc->wp_config->flags |= CONFIG_CREATE_WVC; + if (wavpack_enc->correction_mode == 2) { + wavpack_enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC; + } + } + gst_caps_unref (caps); + } else { + wavpack_enc->correction_mode = 0; + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL), + ("add correction pad failed. no correction file will be created.")); + } + } + } else { + if (wavpack_enc->correction_mode > 0) { + wavpack_enc->correction_mode = 0; + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, SETTINGS, (NULL), + ("settings correction mode only has effect if a bitrate is provided.")); + } + } + + /* MD5, setup MD5 context */ + if ((wavpack_enc->md5) && !(wavpack_enc->md5_context)) { + wavpack_enc->wp_config->flags |= CONFIG_MD5_CHECKSUM; + wavpack_enc->md5_context = (MD5_CTX *) g_malloc0 (sizeof (MD5_CTX)); + MD5Init (wavpack_enc->md5_context); + } + + /* Extra encode processing */ + if (wavpack_enc->extra_processing) { + wavpack_enc->wp_config->flags |= CONFIG_EXTRA_MODE; + } + + /* Joint stereo mode */ + switch (wavpack_enc->joint_stereo_mode) { + case 0: /* default */ + break; + case 1: + wavpack_enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE; + wavpack_enc->wp_config->flags &= ~CONFIG_JOINT_STEREO; + break; + case 2: + wavpack_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 = (int32_t *) g_malloc0 (sizeof (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) +{ + write_id *wid = (write_id *) id; + GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (wid->wavpack_enc); + GstFlowReturn ret; + GstBuffer *buffer; + guchar *block = (guchar *) data; + + if (wid->correction == FALSE) { + /* we got something that should be pushed to the (non-correction) src pad */ + + /* try to allocate a buffer, compatible with the pad, fail otherwise */ + ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->srcpad, + GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (wavpack_enc->srcpad), + &buffer); + if (ret != GST_FLOW_OK) { + wavpack_enc->srcpad_last_return = ret; + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("Dropped one block (%d bytes) of encoded data while allocating buffer! Reason: '%s'\n", + count, gst_flow_get_name (ret))); + return FALSE; + } + + g_memmove (GST_BUFFER_DATA (buffer), block, count); + + if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p') + && (block[3] == 'k')) { + /* if it's a Wavpack block set buffer timestamp and duration, etc */ + WavpackHeader wph; + + GST_DEBUG ("got %d bytes of encoded wavpack data", count); + gst_wavpack_read_header (&wph, block); + + /* if it's the first wavpack block save it for later reference + * i.e. sample count correction and send a NEW_SEGMENT event */ + if (wph.block_index == 0) { + GstEvent *event = gst_event_new_new_segment (FALSE, + 1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0); + + gst_pad_push_event (wavpack_enc->srcpad, event); + wavpack_enc->first_block = g_malloc0 (count); + g_memmove (wavpack_enc->first_block, block, count); + wavpack_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, + wavpack_enc->samplerate); + GST_BUFFER_DURATION (buffer) = + gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, + wavpack_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 ("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 */ + ret = gst_pad_push (wavpack_enc->srcpad, buffer); + wavpack_enc->srcpad_last_return = ret; + if (ret == GST_FLOW_OK) { + return TRUE; + } else { + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("Dropped one block (%d bytes) of encoded data while pushing! Reason: '%s'\n", + count, gst_flow_get_name (ret))); + return FALSE; + } + } else if (wid->correction == TRUE) { + /* we got something that should be pushed to the correction src pad */ + + /* is the correction pad linked? */ + if (!gst_pad_is_linked (wavpack_enc->wvcsrcpad)) { + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("Dropped one block (%d bytes) of encoded correction data because of unlinked pad", + count)); + wavpack_enc->wvcsrcpad_last_return = GST_FLOW_NOT_LINKED; + return FALSE; + } + + /* try to allocate a buffer, compatible with the pad, fail otherwise */ + ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->wvcsrcpad, + GST_BUFFER_OFFSET_NONE, count, + GST_PAD_CAPS (wavpack_enc->wvcsrcpad), &buffer); + if (ret != GST_FLOW_OK) { + wavpack_enc->wvcsrcpad_last_return = ret; + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("Dropped one block (%d bytes) of encoded correction data while allocating buffer! Reason: '%s'\n", + count, gst_flow_get_name (ret))); + return FALSE; + } + + g_memmove (GST_BUFFER_DATA (buffer), block, count); + + if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p') + && (block[3] == 'k')) { + /* if it's a Wavpack block set buffer timestamp and duration, etc */ + WavpackHeader wph; + + GST_DEBUG ("got %d bytes of encoded wavpack correction data", count); + gst_wavpack_read_header (&wph, block); + + /* if it's the first wavpack block send a NEW_SEGMENT + * event */ + if (wph.block_index == 0) { + GstEvent *event = gst_event_new_new_segment (FALSE, + 1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0); + + gst_pad_push_event (wavpack_enc->wvcsrcpad, event); + } + + /* 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, + wavpack_enc->samplerate); + GST_BUFFER_DURATION (buffer) = + gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, + wavpack_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 ("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 */ + ret = gst_pad_push (wavpack_enc->wvcsrcpad, buffer); + wavpack_enc->wvcsrcpad_last_return = ret; + if (ret == GST_FLOW_OK) + return TRUE; + else { + GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("Dropped one block (%d bytes) of encoded correction data while pushing! Reason: '%s'\n", + count, gst_flow_get_name (ret))); + return FALSE; + } + } else { + /* (correction != TRUE) && (correction != FALSE), wtf? ignore this */ + g_assert_not_reached (); + return TRUE; + } +} + +static GstFlowReturn +gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) +{ + GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); + uint32_t sample_count = + GST_BUFFER_SIZE (buf) / ((wavpack_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 */ + wavpack_enc->srcpad_last_return = wavpack_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 (!wavpack_enc->wp_context) { + gint64 duration; + GstFormat fmt = GST_FORMAT_DEFAULT; + + /* create raw context */ + wavpack_enc->wp_context = + WavpackOpenFileOutput (gst_wavpack_enc_push_block, wavpack_enc->wv_id, + (wavpack_enc->correction_mode > 0) ? wavpack_enc->wvc_id : NULL); + if (!wavpack_enc->wp_context) { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL), + ("error creating Wavpack context")); + gst_object_unref (wavpack_enc); + gst_buffer_unref (buf); + return GST_FLOW_ERROR; + } + + /* set the WavpackConfig according to our parameters */ + gst_wavpack_enc_set_wp_config (wavpack_enc); + + /* try to get the duration (or an estimate) in samples from upstream */ + if (gst_pad_query_peer_duration (pad, &fmt, &duration)) { + switch (fmt) { + case GST_FORMAT_DEFAULT: + break; + case GST_FORMAT_TIME: + duration = + gst_util_uint64_scale (wavpack_enc->samplerate, + duration, GST_SECOND); + break; + default: + duration = 0; + break; + } + } else { + duration = 0; + } + + /* Wavpack doesn't support more than 2^32 samples unfortunately */ + if (duration > G_GINT64_CONSTANT (1) << 32) { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, SETTINGS, (NULL), + ("more than 2^32 samples are not supported")); + WavpackCloseFile (wavpack_enc->wp_context); + gst_object_unref (wavpack_enc); + gst_buffer_unref (buf); + return GST_FLOW_ERROR; + } + + /* set the configuration to the context now that we know everything + * and initialize the encoder */ + if (!WavpackSetConfiguration (wavpack_enc->wp_context, + wavpack_enc->wp_config, (uint32_t) duration) + || !WavpackPackInit (wavpack_enc->wp_context)) { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, SETTINGS, (NULL), + ("error setting up wavpack encoding context")); + WavpackCloseFile (wavpack_enc->wp_context); + gst_object_unref (wavpack_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 (wavpack_enc->md5) { + MD5Update (wavpack_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, + wavpack_enc->width); + + gst_buffer_unref (buf); + + /* encode and handle return values from encoding */ + if (WavpackPackSamples (wavpack_enc->wp_context, data, + sample_count / wavpack_enc->channels)) { + GST_DEBUG ("encoding samples successfull"); + ret = GST_FLOW_OK; + } else { + if ((wavpack_enc->srcpad_last_return == GST_FLOW_RESEND) || + (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { + ret = GST_FLOW_RESEND; + } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_OK) || + (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_OK)) { + ret = GST_FLOW_OK; + } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && + (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) { + ret = GST_FLOW_NOT_LINKED; + } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_WRONG_STATE) && + (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) { + ret = GST_FLOW_WRONG_STATE; + } else { + GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, ENCODE, (NULL), + ("encoding samples failed")); + ret = GST_FLOW_ERROR; + } + } + + g_free (data); + gst_object_unref (wavpack_enc); + return ret; +} + +static void +gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * wavpack_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 (wavpack_enc); + g_return_if_fail (wavpack_enc->first_block); + + /* update the sample count in the first block */ + WavpackUpdateNumSamples (wavpack_enc->wp_context, wavpack_enc->first_block); + + /* try to seek to the beginning of the output */ + ret = gst_pad_push_event (wavpack_enc->srcpad, event); + if (ret) { + /* try to rewrite the first block */ + ret = gst_wavpack_enc_push_block (wavpack_enc->wv_id, + wavpack_enc->first_block, wavpack_enc->first_block_size); + if (ret) { + GST_DEBUG ("rewriting of first block succeeded!"); + } else { + GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, WRITE, (NULL), + ("rewriting of first block failed while pushing!")); + } + } else { + GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL), + ("rewriting of first block failed. Seeking to first block failed!")); + } +} + +static gboolean +gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event) +{ + GstWavpackEnc *wavpack_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 (wavpack_enc->wp_context); + + /* write the MD5 sum if we have to write one */ + if ((wavpack_enc->md5) && (wavpack_enc->md5_context)) { + guchar md5_digest[16]; + + MD5Final (md5_digest, wavpack_enc->md5_context); + WavpackStoreMD5Sum (wavpack_enc->wp_context, md5_digest); + } + + /* Try to rewrite the first frame with the correct sample number if we + * had a wrong one at the start of encoding */ + if ((wavpack_enc->first_block) + && (WavpackGetNumSamples (wavpack_enc->wp_context) != + WavpackGetSampleIndex (wavpack_enc->wp_context))) + gst_wavpack_enc_rewrite_first_block (wavpack_enc); + + /* close the context if not already happened */ + if (wavpack_enc->wp_context) { + WavpackCloseFile (wavpack_enc->wp_context); + wavpack_enc->wp_context = NULL; + } + + ret = gst_pad_event_default (pad, event); + break; + case GST_EVENT_NEWSEGMENT: + if (wavpack_enc->wp_context) { + GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL), + ("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 (wavpack_enc); + return ret; +} + +static GstStateChangeReturn +gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition) +{ + GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; + GstWavpackEnc *wavpack_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 */ + wavpack_enc->srcpad_last_return = wavpack_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 (wavpack_enc->wp_context) { + WavpackCloseFile (wavpack_enc->wp_context); + wavpack_enc->wp_context = NULL; + } + if (wavpack_enc->wp_config) { + g_free (wavpack_enc->wp_config); + wavpack_enc->wp_config = NULL; + } + if (wavpack_enc->first_block) { + g_free (wavpack_enc->first_block); + wavpack_enc->first_block = NULL; + wavpack_enc->first_block_size = 0; + } + if (wavpack_enc->md5_context) { + g_free (wavpack_enc->md5_context); + wavpack_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 */ + wavpack_enc->srcpad_last_return = wavpack_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 *wavpack_enc = GST_WAVPACK_ENC (object); + + switch (prop_id) { + case ARG_MODE: + wavpack_enc->mode = g_value_get_enum (value); + break; + case ARG_BITRATE: + wavpack_enc->bitrate = g_value_get_double (value); + break; + case ARG_CORRECTION_MODE: + wavpack_enc->correction_mode = g_value_get_enum (value); + break; + case ARG_MD5: + wavpack_enc->md5 = g_value_get_boolean (value); + break; + case ARG_EXTRA_PROCESSING: + wavpack_enc->extra_processing = g_value_get_boolean (value); + break; + case ARG_JOINT_STEREO_MODE: + wavpack_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 *wavpack_enc = GST_WAVPACK_ENC (object); + + switch (prop_id) { + case ARG_MODE: + g_value_set_enum (value, wavpack_enc->mode); + break; + case ARG_BITRATE: + g_value_set_double (value, wavpack_enc->bitrate); + break; + case ARG_CORRECTION_MODE: + g_value_set_enum (value, wavpack_enc->correction_mode); + break; + case ARG_MD5: + g_value_set_boolean (value, wavpack_enc->md5); + break; + case ARG_EXTRA_PROCESSING: + g_value_set_boolean (value, wavpack_enc->extra_processing); + break; + case ARG_JOINT_STEREO_MODE: + g_value_set_enum (value, wavpack_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; +} -- cgit v1.2.1