1 /* GStreamer Wavpack encoder plugin
2  * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
3  *
4  * gstwavpackdec.c: Wavpack audio encoder
5  *
6  * This library is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Library General Public
8  * License as published by the Free Software Foundation; either
9  * version 2 of the License, or (at your option) any later version.
10  *
11  * This library is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
14  * Library General Public License for more details.
15  *
16  * You should have received a copy of the GNU Library General Public
17  * License along with this library; if not, write to the
18  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19  * Boston, MA 02110-1301, USA.
20  */
21 
22 /**
23  * SECTION:element-wavpackenc
24  *
25  * WavpackEnc encodes raw audio into a framed Wavpack stream.
26  * <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
27  * audio codec that features both lossless and lossy encoding.
28  *
29  * <refsect2>
30  * <title>Example launch line</title>
31  * |[
32  * gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
33  * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
34  * as the Wavpack encoder only accepts input with 32 bit width.
35  * |[
36  * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
37  * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
38  * lossless encoding (the file output will be fairly large).
39  * |[
40  * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
41  * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
42  * lossy encoding at a certain bitrate (the file will be fairly small).
43  * </refsect2>
44  */
45 
46 /*
47  * TODO: - add 32 bit float mode. CONFIG_FLOAT_DATA
48  */
49 
50 #include <string.h>
51 #include <gst/gst.h>
52 #include <glib/gprintf.h>
53 
54 #include <wavpack/wavpack.h>
55 #include "gstwavpackenc.h"
56 #include "gstwavpackcommon.h"
57 
58 static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
59 static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
60 static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
61     GstAudioInfo * info);
62 static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
63     GstBuffer * in_buf);
64 static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
65     GstEvent * event);
66 
67 static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
68 static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
69 
70 static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
71     const GValue * value, GParamSpec * pspec);
72 static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
73     GValue * value, GParamSpec * pspec);
74 
75 enum
76 {
77   ARG_0,
78   ARG_MODE,
79   ARG_BITRATE,
80   ARG_BITSPERSAMPLE,
81   ARG_CORRECTION_MODE,
82   ARG_MD5,
83   ARG_EXTRA_PROCESSING,
84   ARG_JOINT_STEREO_MODE
85 };
86 
87 GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
88 #define GST_CAT_DEFAULT gst_wavpack_enc_debug
89 
90 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
91     GST_PAD_SINK,
92     GST_PAD_ALWAYS,
93     GST_STATIC_CAPS ("audio/x-raw, "
94         "format = (string) " GST_AUDIO_NE (S32) ", "
95         "layout = (string) interleaved, "
96         "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
97     );
98 
99 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
100     GST_PAD_SRC,
101     GST_PAD_ALWAYS,
102     GST_STATIC_CAPS ("audio/x-wavpack, "
103         "depth = (int) [ 1, 32 ], "
104         "channels = (int) [ 1, 8 ], "
105         "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
106     );
107 
108 static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
109     GST_PAD_SRC,
110     GST_PAD_SOMETIMES,
111     GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
112     );
113 
114 enum
115 {
116   GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
117   GST_WAVPACK_ENC_MODE_FAST,
118   GST_WAVPACK_ENC_MODE_DEFAULT,
119   GST_WAVPACK_ENC_MODE_HIGH,
120   GST_WAVPACK_ENC_MODE_VERY_HIGH
121 };
122 
123 #define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
124 static GType
gst_wavpack_enc_mode_get_type(void)125 gst_wavpack_enc_mode_get_type (void)
126 {
127   static GType qtype = 0;
128 
129   if (qtype == 0) {
130     static const GEnumValue values[] = {
131 #if 0
132       /* Very Fast Compression is not supported yet, but will be supported
133        * in future wavpack versions */
134       {GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
135 #endif
136       {GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
137       {GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
138       {GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
139       {GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
140       {0, NULL, NULL}
141     };
142 
143     qtype = g_enum_register_static ("GstWavpackEncMode", values);
144   }
145   return qtype;
146 }
147 
148 enum
149 {
150   GST_WAVPACK_CORRECTION_MODE_OFF = 0,
151   GST_WAVPACK_CORRECTION_MODE_ON,
152   GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
153 };
154 
155 #define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
156 static GType
gst_wavpack_enc_correction_mode_get_type(void)157 gst_wavpack_enc_correction_mode_get_type (void)
158 {
159   static GType qtype = 0;
160 
161   if (qtype == 0) {
162     static const GEnumValue values[] = {
163       {GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
164       {GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
165       {GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
166           "Create optimized correction file", "optimized"},
167       {0, NULL, NULL}
168     };
169 
170     qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
171   }
172   return qtype;
173 }
174 
175 enum
176 {
177   GST_WAVPACK_JS_MODE_AUTO = 0,
178   GST_WAVPACK_JS_MODE_LEFT_RIGHT,
179   GST_WAVPACK_JS_MODE_MID_SIDE
180 };
181 
182 #define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
183 static GType
gst_wavpack_enc_joint_stereo_mode_get_type(void)184 gst_wavpack_enc_joint_stereo_mode_get_type (void)
185 {
186   static GType qtype = 0;
187 
188   if (qtype == 0) {
189     static const GEnumValue values[] = {
190       {GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
191       {GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
192       {GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
193       {0, NULL, NULL}
194     };
195 
196     qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
197   }
198   return qtype;
199 }
200 
201 #define gst_wavpack_enc_parent_class parent_class
202 G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
203 
204 static void
gst_wavpack_enc_class_init(GstWavpackEncClass * klass)205 gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
206 {
207   GObjectClass *gobject_class = (GObjectClass *) klass;
208   GstElementClass *element_class = (GstElementClass *) (klass);
209   GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
210 
211   /* add pad templates */
212   gst_element_class_add_static_pad_template (element_class, &sink_factory);
213   gst_element_class_add_static_pad_template (element_class, &src_factory);
214   gst_element_class_add_static_pad_template (element_class, &wvcsrc_factory);
215 
216   /* set element details */
217   gst_element_class_set_static_metadata (element_class, "Wavpack audio encoder",
218       "Codec/Encoder/Audio",
219       "Encodes audio with the Wavpack lossless/lossy audio codec",
220       "Sebastian Dröge <slomo@circular-chaos.org>");
221 
222   /* set property handlers */
223   gobject_class->set_property = gst_wavpack_enc_set_property;
224   gobject_class->get_property = gst_wavpack_enc_get_property;
225 
226   base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
227   base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
228   base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
229   base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
230   base_class->sink_event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
231 
232   /* install all properties */
233   g_object_class_install_property (gobject_class, ARG_MODE,
234       g_param_spec_enum ("mode", "Encoding mode",
235           "Speed versus compression tradeoff.",
236           GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
237           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
238   g_object_class_install_property (gobject_class, ARG_BITRATE,
239       g_param_spec_uint ("bitrate", "Bitrate",
240           "Try to encode with this average bitrate (bits/sec). "
241           "This enables lossy encoding, values smaller than 24000 disable it again.",
242           0, 9600000, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
243   g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
244       g_param_spec_double ("bits-per-sample", "Bits per sample",
245           "Try to encode with this amount of bits per sample. "
246           "This enables lossy encoding, values smaller than 2.0 disable it again.",
247           0.0, 24.0, 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
248   g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
249       g_param_spec_enum ("correction-mode", "Correction stream mode",
250           "Use this mode for the correction stream. Only works in lossy mode!",
251           GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
252           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
253   g_object_class_install_property (gobject_class, ARG_MD5,
254       g_param_spec_boolean ("md5", "MD5",
255           "Store MD5 hash of raw samples within the file.", FALSE,
256           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
257   g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
258       g_param_spec_uint ("extra-processing", "Extra processing",
259           "Use better but slower filters for better compression/quality.",
260           0, 6, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
261   g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
262       g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
263           "Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
264           GST_WAVPACK_JS_MODE_AUTO,
265           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
266 }
267 
268 static void
gst_wavpack_enc_reset(GstWavpackEnc * enc)269 gst_wavpack_enc_reset (GstWavpackEnc * enc)
270 {
271   /* close and free everything stream related if we already did something */
272   if (enc->wp_context) {
273     WavpackCloseFile (enc->wp_context);
274     enc->wp_context = NULL;
275   }
276   if (enc->wp_config) {
277     g_free (enc->wp_config);
278     enc->wp_config = NULL;
279   }
280   if (enc->first_block) {
281     g_free (enc->first_block);
282     enc->first_block = NULL;
283   }
284   enc->first_block_size = 0;
285   if (enc->md5_context) {
286     g_checksum_free (enc->md5_context);
287     enc->md5_context = NULL;
288   }
289   if (enc->pending_segment)
290     gst_event_unref (enc->pending_segment);
291   enc->pending_segment = NULL;
292 
293   if (enc->pending_buffer) {
294     gst_buffer_unref (enc->pending_buffer);
295     enc->pending_buffer = NULL;
296     enc->pending_offset = 0;
297   }
298 
299   /* reset the last returns to GST_FLOW_OK. This is only set to something else
300    * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
301    * so not valid anymore */
302   enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
303 
304   /* reset stream information */
305   enc->samplerate = 0;
306   enc->depth = 0;
307   enc->channels = 0;
308   enc->channel_mask = 0;
309   enc->need_channel_remap = FALSE;
310 
311   enc->timestamp_offset = GST_CLOCK_TIME_NONE;
312   enc->next_ts = GST_CLOCK_TIME_NONE;
313 }
314 
315 static void
gst_wavpack_enc_init(GstWavpackEnc * enc)316 gst_wavpack_enc_init (GstWavpackEnc * enc)
317 {
318   GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
319 
320   /* initialize object attributes */
321   enc->wp_config = NULL;
322   enc->wp_context = NULL;
323   enc->first_block = NULL;
324   enc->md5_context = NULL;
325   gst_wavpack_enc_reset (enc);
326 
327   enc->wv_id.correction = FALSE;
328   enc->wv_id.wavpack_enc = enc;
329   enc->wv_id.passthrough = FALSE;
330   enc->wvc_id.correction = TRUE;
331   enc->wvc_id.wavpack_enc = enc;
332   enc->wvc_id.passthrough = FALSE;
333 
334   /* set default values of params */
335   enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
336   enc->bitrate = 0;
337   enc->bps = 0.0;
338   enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
339   enc->md5 = FALSE;
340   enc->extra_processing = 0;
341   enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
342 
343   /* require perfect ts */
344   gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
345 
346   GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
347 }
348 
349 
350 static gboolean
gst_wavpack_enc_start(GstAudioEncoder * enc)351 gst_wavpack_enc_start (GstAudioEncoder * enc)
352 {
353   GST_DEBUG_OBJECT (enc, "start");
354 
355   return TRUE;
356 }
357 
358 static gboolean
gst_wavpack_enc_stop(GstAudioEncoder * enc)359 gst_wavpack_enc_stop (GstAudioEncoder * enc)
360 {
361   GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
362 
363   GST_DEBUG_OBJECT (enc, "stop");
364   gst_wavpack_enc_reset (wpenc);
365 
366   return TRUE;
367 }
368 
369 static gboolean
gst_wavpack_enc_set_format(GstAudioEncoder * benc,GstAudioInfo * info)370 gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
371 {
372   GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
373   GstAudioChannelPosition *pos;
374   GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
375   GstCaps *caps;
376   guint64 mask = 0;
377 
378   /* we may be configured again, but that change should have cleanup context */
379   g_assert (enc->wp_context == NULL);
380 
381   enc->channels = GST_AUDIO_INFO_CHANNELS (info);
382   enc->depth = GST_AUDIO_INFO_DEPTH (info);
383   enc->samplerate = GST_AUDIO_INFO_RATE (info);
384 
385   pos = info->position;
386   g_assert (pos);
387 
388   /* If one channel is NONE they'll be all undefined */
389   if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
390     goto invalid_channels;
391   }
392 
393   enc->channel_mask =
394       gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
395   enc->need_channel_remap =
396       gst_wavpack_set_channel_mapping (pos, enc->channels,
397       enc->channel_mapping);
398 
399   /* wavpack caps hold gst mask, not wavpack mask */
400   gst_audio_channel_positions_to_mask (opos, enc->channels, FALSE, &mask);
401 
402   /* set fixed src pad caps now that we know what we will get */
403   caps = gst_caps_new_simple ("audio/x-wavpack",
404       "channels", G_TYPE_INT, enc->channels,
405       "rate", G_TYPE_INT, enc->samplerate,
406       "depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
407 
408   if (mask)
409     gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
410 
411   if (!gst_audio_encoder_set_output_format (benc, caps))
412     goto setting_src_caps_failed;
413 
414   gst_caps_unref (caps);
415 
416   /* no special feedback to base class; should provide all available samples */
417 
418   return TRUE;
419 
420   /* ERRORS */
421 setting_src_caps_failed:
422   {
423     GST_DEBUG_OBJECT (enc,
424         "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
425     gst_caps_unref (caps);
426     return FALSE;
427   }
428 invalid_channels:
429   {
430     GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
431     return FALSE;
432   }
433 }
434 
435 static void
gst_wavpack_enc_set_wp_config(GstWavpackEnc * enc)436 gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
437 {
438   enc->wp_config = g_new0 (WavpackConfig, 1);
439   /* set general stream informations in the WavpackConfig */
440   enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
441   enc->wp_config->bits_per_sample = enc->depth;
442   enc->wp_config->num_channels = enc->channels;
443   enc->wp_config->channel_mask = enc->channel_mask;
444   enc->wp_config->sample_rate = enc->samplerate;
445 
446   /*
447    * Set parameters in WavpackConfig
448    */
449 
450   /* Encoding mode */
451   switch (enc->mode) {
452 #if 0
453     case GST_WAVPACK_ENC_MODE_VERY_FAST:
454       enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG;
455       enc->wp_config->flags |= CONFIG_FAST_FLAG;
456       break;
457 #endif
458     case GST_WAVPACK_ENC_MODE_FAST:
459       enc->wp_config->flags |= CONFIG_FAST_FLAG;
460       break;
461     case GST_WAVPACK_ENC_MODE_DEFAULT:
462       break;
463     case GST_WAVPACK_ENC_MODE_HIGH:
464       enc->wp_config->flags |= CONFIG_HIGH_FLAG;
465       break;
466     case GST_WAVPACK_ENC_MODE_VERY_HIGH:
467       enc->wp_config->flags |= CONFIG_HIGH_FLAG;
468       enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
469       break;
470   }
471 
472   /* Bitrate, enables lossy mode */
473   if (enc->bitrate) {
474     enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
475     enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
476     enc->wp_config->bitrate = enc->bitrate / 1000.0;
477   } else if (enc->bps) {
478     enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
479     enc->wp_config->bitrate = enc->bps;
480   }
481 
482   /* Correction Mode, only in lossy mode */
483   if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
484     if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
485       GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
486           "framed", G_TYPE_BOOLEAN, TRUE, NULL);
487 
488       enc->wvcsrcpad =
489           gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
490 
491       /* try to add correction src pad, don't set correction mode on failure */
492       GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
493           GST_PTR_FORMAT, caps);
494       if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
495         enc->correction_mode = 0;
496         GST_WARNING_OBJECT (enc, "setting correction caps failed");
497       } else {
498         gst_pad_use_fixed_caps (enc->wvcsrcpad);
499         gst_pad_set_active (enc->wvcsrcpad, TRUE);
500         gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad);
501         enc->wp_config->flags |= CONFIG_CREATE_WVC;
502         if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) {
503           enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
504         }
505       }
506       gst_caps_unref (caps);
507     }
508   } else {
509     if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
510       enc->correction_mode = 0;
511       GST_WARNING_OBJECT (enc, "setting correction mode only has "
512           "any effect if a bitrate is provided.");
513     }
514   }
515   gst_element_no_more_pads (GST_ELEMENT (enc));
516 
517   /* MD5, setup MD5 context */
518   if ((enc->md5) && !(enc->md5_context)) {
519     enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
520     enc->md5_context = g_checksum_new (G_CHECKSUM_MD5);
521   }
522 
523   /* Extra encode processing */
524   if (enc->extra_processing) {
525     enc->wp_config->flags |= CONFIG_EXTRA_MODE;
526     enc->wp_config->xmode = enc->extra_processing;
527   }
528 
529   /* Joint stereo mode */
530   switch (enc->joint_stereo_mode) {
531     case GST_WAVPACK_JS_MODE_AUTO:
532       break;
533     case GST_WAVPACK_JS_MODE_LEFT_RIGHT:
534       enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
535       enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
536       break;
537     case GST_WAVPACK_JS_MODE_MID_SIDE:
538       enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
539       break;
540   }
541 }
542 
543 static int
gst_wavpack_enc_push_block(void * id,void * data,int32_t count)544 gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
545 {
546   GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
547   GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
548   GstFlowReturn *flow;
549   GstBuffer *buffer;
550   GstPad *pad;
551   guchar *block = (guchar *) data;
552   gint samples = 0;
553 
554   pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
555   flow =
556       (wid->correction) ? &enc->
557       wvcsrcpad_last_return : &enc->srcpad_last_return;
558 
559   buffer = gst_buffer_new_and_alloc (count);
560   gst_buffer_fill (buffer, 0, data, count);
561 
562   if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
563     /* if it's a Wavpack block set buffer timestamp and duration, etc */
564     WavpackHeader wph;
565 
566     GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
567         count, (wid->correction) ? "correction " : "");
568 
569     gst_wavpack_read_header (&wph, block);
570 
571     /* Only set when pushing the first buffer again, in that case
572      * we don't want to delay the buffer or push newsegment events
573      */
574     if (!wid->passthrough) {
575       /* Only push complete blocks */
576       if (enc->pending_buffer == NULL) {
577         enc->pending_buffer = buffer;
578         enc->pending_offset = wph.block_index;
579       } else if (enc->pending_offset == wph.block_index) {
580         enc->pending_buffer = gst_buffer_append (enc->pending_buffer, buffer);
581       } else {
582         GST_ERROR ("Got incomplete block, dropping");
583         gst_buffer_unref (enc->pending_buffer);
584         enc->pending_buffer = buffer;
585         enc->pending_offset = wph.block_index;
586       }
587 
588       /* Is this the not-final block of multi-channel data? If so, just
589        * accumulate and return here. */
590       if (!(wph.flags & FINAL_BLOCK) && ((block[32] & ID_OPTIONAL_DATA) == 0))
591         return TRUE;
592 
593       buffer = enc->pending_buffer;
594       enc->pending_buffer = NULL;
595       enc->pending_offset = 0;
596 
597       /* only send segment on correction pad,
598        * regular pad is handled normally by baseclass */
599       if (wid->correction && enc->pending_segment) {
600         gst_pad_push_event (pad, enc->pending_segment);
601         enc->pending_segment = NULL;
602       }
603 
604       if (wph.block_index == 0) {
605         /* save header for later reference, so we can re-send it later on
606          * EOS with fixed up values for total sample count etc. */
607         if (enc->first_block == NULL && !wid->correction) {
608           GstMapInfo map;
609 
610           gst_buffer_map (buffer, &map, GST_MAP_READ);
611           enc->first_block = g_memdup (map.data, map.size);
612           enc->first_block_size = map.size;
613           gst_buffer_unmap (buffer, &map);
614         }
615       }
616     }
617     samples = wph.block_samples;
618 
619     /* decorate buffer */
620     /* NOTE: this will get overwritten by baseclass, but stay for those
621      * that are pushed directly
622      * FIXME: add setting to baseclass to avoid overwriting it ?? */
623     GST_BUFFER_OFFSET (buffer) = wph.block_index;
624     GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
625   } else {
626     /* if it's something else set no timestamp and duration on the buffer */
627     GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
628   }
629 
630   if (wid->correction || wid->passthrough) {
631     /* push the buffer and forward errors */
632     GST_DEBUG_OBJECT (enc, "pushing buffer with %" G_GSIZE_FORMAT " bytes",
633         gst_buffer_get_size (buffer));
634     *flow = gst_pad_push (pad, buffer);
635   } else {
636     GST_DEBUG_OBJECT (enc, "handing frame of %" G_GSIZE_FORMAT " bytes",
637         gst_buffer_get_size (buffer));
638     *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
639         samples);
640   }
641 
642   if (*flow != GST_FLOW_OK) {
643     GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
644         GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
645     return FALSE;
646   }
647 
648   return TRUE;
649 }
650 
651 static void
gst_wavpack_enc_fix_channel_order(GstWavpackEnc * enc,gint32 * data,gint nsamples)652 gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
653     gint nsamples)
654 {
655   gint i, j;
656   gint32 tmp[8];
657 
658   for (i = 0; i < nsamples / enc->channels; i++) {
659     for (j = 0; j < enc->channels; j++) {
660       tmp[enc->channel_mapping[j]] = data[j];
661     }
662     for (j = 0; j < enc->channels; j++) {
663       data[j] = tmp[j];
664     }
665     data += enc->channels;
666   }
667 }
668 
669 static GstFlowReturn
gst_wavpack_enc_handle_frame(GstAudioEncoder * benc,GstBuffer * buf)670 gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
671 {
672   GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
673   uint32_t sample_count;
674   GstFlowReturn ret;
675   GstMapInfo map;
676 
677   /* base class ensures configuration */
678   g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
679 
680   /* reset the last returns to GST_FLOW_OK. This is only set to something else
681    * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
682    * so not valid anymore */
683   enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
684 
685   if (G_UNLIKELY (!buf))
686     return gst_wavpack_enc_drain (enc);
687 
688   sample_count = gst_buffer_get_size (buf) / 4;
689   GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
690 
691   /* check if we already have a valid WavpackContext, otherwise make one */
692   if (!enc->wp_context) {
693     /* create raw context */
694     enc->wp_context =
695         WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
696         (enc->correction_mode > 0) ? &enc->wvc_id : NULL);
697     if (!enc->wp_context)
698       goto context_failed;
699 
700     /* set the WavpackConfig according to our parameters */
701     gst_wavpack_enc_set_wp_config (enc);
702 
703     /* set the configuration to the context now that we know everything
704      * and initialize the encoder */
705     if (!WavpackSetConfiguration (enc->wp_context,
706             enc->wp_config, (uint32_t) (-1))
707         || !WavpackPackInit (enc->wp_context)) {
708       WavpackCloseFile (enc->wp_context);
709       goto config_failed;
710     }
711     GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
712   }
713 
714   if (enc->need_channel_remap) {
715     buf = gst_buffer_make_writable (buf);
716     gst_buffer_map (buf, &map, GST_MAP_WRITE);
717     gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
718     gst_buffer_unmap (buf, &map);
719   }
720 
721   gst_buffer_map (buf, &map, GST_MAP_READ);
722 
723   /* if we want to append the MD5 sum to the stream update it here
724    * with the current raw samples */
725   if (enc->md5) {
726     g_checksum_update (enc->md5_context, map.data, map.size);
727   }
728 
729   /* encode and handle return values from encoding */
730   if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
731           sample_count / enc->channels)) {
732     GST_DEBUG_OBJECT (enc, "encoding samples successful");
733     gst_buffer_unmap (buf, &map);
734     ret = GST_FLOW_OK;
735   } else {
736     gst_buffer_unmap (buf, &map);
737     if ((enc->srcpad_last_return == GST_FLOW_OK) ||
738         (enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
739       ret = GST_FLOW_OK;
740     } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
741         (enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
742       ret = GST_FLOW_NOT_LINKED;
743     } else if ((enc->srcpad_last_return == GST_FLOW_FLUSHING) &&
744         (enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
745       ret = GST_FLOW_FLUSHING;
746     } else {
747       goto encoding_failed;
748     }
749   }
750 
751 exit:
752   return ret;
753 
754   /* ERRORS */
755 encoding_failed:
756   {
757     GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
758         ("encoding samples failed"));
759     ret = GST_FLOW_ERROR;
760     goto exit;
761   }
762 config_failed:
763   {
764     GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
765         ("error setting up wavpack encoding context"));
766     ret = GST_FLOW_ERROR;
767     goto exit;
768   }
769 context_failed:
770   {
771     GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
772         ("error creating Wavpack context"));
773     ret = GST_FLOW_ERROR;
774     goto exit;
775   }
776 }
777 
778 static void
gst_wavpack_enc_rewrite_first_block(GstWavpackEnc * enc)779 gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
780 {
781   GstSegment segment;
782   gboolean ret;
783   GstQuery *query;
784   gboolean seekable = FALSE;
785 
786   g_return_if_fail (enc);
787   g_return_if_fail (enc->first_block);
788 
789   /* update the sample count in the first block */
790   WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
791 
792   /* try to seek to the beginning of the output */
793   query = gst_query_new_seeking (GST_FORMAT_BYTES);
794   if (gst_pad_peer_query (GST_AUDIO_ENCODER_SRC_PAD (enc), query)) {
795     GstFormat format;
796 
797     gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
798     if (format != GST_FORMAT_BYTES)
799       seekable = FALSE;
800   } else {
801     GST_LOG_OBJECT (enc, "SEEKING query not handled");
802   }
803   gst_query_unref (query);
804 
805   if (!seekable) {
806     GST_DEBUG_OBJECT (enc, "downstream not seekable; not rewriting");
807     return;
808   }
809 
810   gst_segment_init (&segment, GST_FORMAT_BYTES);
811   ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
812       gst_event_new_segment (&segment));
813   if (ret) {
814     /* try to rewrite the first block */
815     GST_DEBUG_OBJECT (enc, "rewriting first block ...");
816     enc->wv_id.passthrough = TRUE;
817     ret = gst_wavpack_enc_push_block (&enc->wv_id,
818         enc->first_block, enc->first_block_size);
819     enc->wv_id.passthrough = FALSE;
820     g_free (enc->first_block);
821     enc->first_block = NULL;
822   } else {
823     GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
824         "Seeking to first block failed!");
825   }
826 }
827 
828 static GstFlowReturn
gst_wavpack_enc_drain(GstWavpackEnc * enc)829 gst_wavpack_enc_drain (GstWavpackEnc * enc)
830 {
831   if (!enc->wp_context)
832     return GST_FLOW_OK;
833 
834   GST_DEBUG_OBJECT (enc, "draining");
835 
836   /* Encode all remaining samples and flush them to the src pads */
837   WavpackFlushSamples (enc->wp_context);
838 
839   /* Drop all remaining data, this is no complete block otherwise
840    * it would've been pushed already */
841   if (enc->pending_buffer) {
842     gst_buffer_unref (enc->pending_buffer);
843     enc->pending_buffer = NULL;
844     enc->pending_offset = 0;
845   }
846 
847   /* write the MD5 sum if we have to write one */
848   if ((enc->md5) && (enc->md5_context)) {
849     guint8 md5_digest[16];
850     gsize digest_len = sizeof (md5_digest);
851 
852     g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
853     if (digest_len == sizeof (md5_digest)) {
854       WavpackStoreMD5Sum (enc->wp_context, md5_digest);
855       WavpackFlushSamples (enc->wp_context);
856     } else
857       GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
858   }
859 
860   /* Try to rewrite the first frame with the correct sample number */
861   if (enc->first_block)
862     gst_wavpack_enc_rewrite_first_block (enc);
863 
864   /* close the context if not already happened */
865   if (enc->wp_context) {
866     WavpackCloseFile (enc->wp_context);
867     enc->wp_context = NULL;
868   }
869 
870   return GST_FLOW_OK;
871 }
872 
873 static gboolean
gst_wavpack_enc_sink_event(GstAudioEncoder * benc,GstEvent * event)874 gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
875 {
876   GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
877 
878   GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
879       GST_EVENT_TYPE_NAME (event));
880 
881   switch (GST_EVENT_TYPE (event)) {
882     case GST_EVENT_SEGMENT:
883       if (enc->wp_context) {
884         GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
885             "already started");
886       }
887       /* peek and hold NEWSEGMENT events for sending on correction pad */
888       if (enc->pending_segment)
889         gst_event_unref (enc->pending_segment);
890       enc->pending_segment = gst_event_ref (event);
891       break;
892     default:
893       break;
894   }
895 
896   /* baseclass handles rest */
897   return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
898 }
899 
900 static void
gst_wavpack_enc_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)901 gst_wavpack_enc_set_property (GObject * object, guint prop_id,
902     const GValue * value, GParamSpec * pspec)
903 {
904   GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
905 
906   switch (prop_id) {
907     case ARG_MODE:
908       enc->mode = g_value_get_enum (value);
909       break;
910     case ARG_BITRATE:{
911       guint val = g_value_get_uint (value);
912 
913       if ((val >= 24000) && (val <= 9600000)) {
914         enc->bitrate = val;
915         enc->bps = 0.0;
916       } else {
917         enc->bitrate = 0;
918         enc->bps = 0.0;
919       }
920       break;
921     }
922     case ARG_BITSPERSAMPLE:{
923       gdouble val = g_value_get_double (value);
924 
925       if ((val >= 2.0) && (val <= 24.0)) {
926         enc->bps = val;
927         enc->bitrate = 0;
928       } else {
929         enc->bps = 0.0;
930         enc->bitrate = 0;
931       }
932       break;
933     }
934     case ARG_CORRECTION_MODE:
935       enc->correction_mode = g_value_get_enum (value);
936       break;
937     case ARG_MD5:
938       enc->md5 = g_value_get_boolean (value);
939       break;
940     case ARG_EXTRA_PROCESSING:
941       enc->extra_processing = g_value_get_uint (value);
942       break;
943     case ARG_JOINT_STEREO_MODE:
944       enc->joint_stereo_mode = g_value_get_enum (value);
945       break;
946     default:
947       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
948       break;
949   }
950 }
951 
952 static void
gst_wavpack_enc_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)953 gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value,
954     GParamSpec * pspec)
955 {
956   GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
957 
958   switch (prop_id) {
959     case ARG_MODE:
960       g_value_set_enum (value, enc->mode);
961       break;
962     case ARG_BITRATE:
963       if (enc->bps == 0.0) {
964         g_value_set_uint (value, enc->bitrate);
965       } else {
966         g_value_set_uint (value, 0);
967       }
968       break;
969     case ARG_BITSPERSAMPLE:
970       if (enc->bitrate == 0) {
971         g_value_set_double (value, enc->bps);
972       } else {
973         g_value_set_double (value, 0.0);
974       }
975       break;
976     case ARG_CORRECTION_MODE:
977       g_value_set_enum (value, enc->correction_mode);
978       break;
979     case ARG_MD5:
980       g_value_set_boolean (value, enc->md5);
981       break;
982     case ARG_EXTRA_PROCESSING:
983       g_value_set_uint (value, enc->extra_processing);
984       break;
985     case ARG_JOINT_STEREO_MODE:
986       g_value_set_enum (value, enc->joint_stereo_mode);
987       break;
988     default:
989       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
990       break;
991   }
992 }
993 
994 gboolean
gst_wavpack_enc_plugin_init(GstPlugin * plugin)995 gst_wavpack_enc_plugin_init (GstPlugin * plugin)
996 {
997   if (!gst_element_register (plugin, "wavpackenc",
998           GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
999     return FALSE;
1000 
1001   GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
1002       "Wavpack encoder");
1003 
1004   return TRUE;
1005 }
1006