diff options
Diffstat (limited to 'sys/dshowdecwrapper/gstdshowaudiodec.c')
| -rw-r--r-- | sys/dshowdecwrapper/gstdshowaudiodec.c | 1189 |
1 files changed, 0 insertions, 1189 deletions
diff --git a/sys/dshowdecwrapper/gstdshowaudiodec.c b/sys/dshowdecwrapper/gstdshowaudiodec.c deleted file mode 100644 index 574ce1c6..00000000 --- a/sys/dshowdecwrapper/gstdshowaudiodec.c +++ /dev/null @@ -1,1189 +0,0 @@ -/* - * GStreamer DirectShow codecs wrapper - * Copyright <2006, 2007, 2008> Fluendo <gstreamer@fluendo.com> - * Copyright <2006, 2007, 2008> Pioneers of the Inevitable <songbird@songbirdnest.com> - * Copyright <2007,2008> Sebastien Moutte <sebastien@moutte.net> - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE - * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING - * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER - * DEALINGS IN THE SOFTWARE. - * - * Alternatively, the contents of this file may be used under the - * GNU Lesser General Public License Version 2.1 (the "LGPL"), in - * which case the following provisions apply instead of the ones - * mentioned above: - * - * 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. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstdshowaudiodec.h" -#include <mmreg.h> - -GST_DEBUG_CATEGORY_STATIC (dshowaudiodec_debug); -#define GST_CAT_DEFAULT dshowaudiodec_debug - -GST_BOILERPLATE (GstDshowAudioDec, gst_dshowaudiodec, GstElement, - GST_TYPE_ELEMENT); -static const CodecEntry *tmp; - -static void gst_dshowaudiodec_dispose (GObject * object); -static GstStateChangeReturn gst_dshowaudiodec_change_state - (GstElement * element, GstStateChange transition); - -/* sink pad overwrites */ -static gboolean gst_dshowaudiodec_sink_setcaps (GstPad * pad, GstCaps * caps); -static GstFlowReturn gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer); -static gboolean gst_dshowaudiodec_sink_event (GstPad * pad, GstEvent * event); - -/* callback used by directshow to push buffers */ -static gboolean gst_dshowaudiodec_push_buffer (byte * buffer, long size, - byte * src_object, UINT64 start, UINT64 stop); - -/* utils */ -static gboolean gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * - adec); -static gboolean gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec * - adec); -static gboolean gst_dshowaudiodec_flush (GstDshowAudioDec * adec); -static gboolean gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec); -static gboolean gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec); - -static const long mpeg_bitrates[2][3][16] = { - /* mpeg 1 */ - { - /* one list per layer 1-3 */ - {0, 32000, 64000, 96000, 128000, 160000, 192000, 224000, 256000, - 288000, 320000, 352000, 384000, 416000, 448000, 0}, - {0, 32000, 48000, 56000, 64000, 80000, 96000, 112000, 128000, - 160000, 192000, 224000, 256000, 320000, 384000, 0}, - {0, 32000, 40000, 48000, 56000, 64000, 80000, 96000, 112000, - 128000, 160000, 192000, 224000, 256000, 320000, 0}, - }, - /* mpeg 2 */ - { - /* one list per layer 1-3 */ - {0, 32000, 48000, 56000, 64000, 80000, 96000, 112000, 128000, 144000, - 160000, 176000, 192000, 224000, 256000, 0}, - {0, 8000, 16000, 24000, 32000, 40000, 48000, 56000, 64000, 80000, 96000, - 112000, 128000, 144000, 160000, 0}, - {0, 8000, 16000, 24000, 32000, 40000, 48000, 56000, 64000, 80000, 96000, - 112000, 128000, 144000, 160000, 0}, - } -}; - -#define GUID_MEDIATYPE_AUDIO {0x73647561, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_PCM {0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_WMAV1 {0x00000160, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_WMAV2 {0x00000161, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_WMAV3 {0x00000162, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_WMAV4 {0x00000163, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_WMS {0x0000000a, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_MP3 {0x00000055, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }} -#define GUID_MEDIASUBTYPE_MPEG1AudioPayload {0x00000050, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xAA, 0x00, 0x38, 0x9b, 0x71 }} - -static const CodecEntry audio_dec_codecs[] = { - {"dshowadec_wma1", - "Windows Media Audio 7", - "DMO", - 0x00000160, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV1, - "audio/x-wma, wmaversion = (int) 1", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - }, - {"dshowadec_wma2", - "Windows Media Audio 8", - "DMO", - 0x00000161, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV2, - "audio/x-wma, wmaversion = (int) 2", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - }, - {"dshowadec_wma3", - "Windows Media Audio 9 Professional", - "DMO", - 0x00000162, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV3, - "audio/x-wma, wmaversion = (int) 3", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - }, - {"dshowadec_wma4", - "Windows Media Audio 9 Lossless", - "DMO", - 0x00000163, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV4, - "audio/x-wma, wmaversion = (int) 4", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - }, - {"dshowadec_wms", - "Windows Media Audio Voice v9", - "DMO", - 0x0000000a, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMS, - "audio/x-wms", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - }, - {"dshowadec_mpeg1", - "MPEG-1 Layer 1,2,3 Audio", - "MPEG Layer-3 Decoder", - 0x00000055, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_MP3, - "audio/mpeg, " - "mpegversion = (int) 1, " - "layer = (int) { 1 , 2, 3 }, " - "rate = (int) [ 8000, 48000 ], " - "channels = (int) [ 1, 2 ], " "parsed= (boolean) true", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) - } -}; - -/* Private map used when dshowadec_mpeg is loaded with layer=1 or 2. - * The problem is that gstreamer doesn't care about caps like layer when connecting pads. - * So I've only one element handling mpeg audio in the public codecs map and - * when it's loaded for mp3, I release the mpeg audio decoder and replace it by - * the one described in this private map. -*/ -static const CodecEntry audio_mpeg_1_2[] = { "dshowadec_mpeg_1_2", - "MPEG-1 Layer 1,2 Audio", - "MPEG Audio Decoder", - 0x00000050, - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_MPEG1AudioPayload, - "audio/mpeg, " - "mpegversion = (int) 1, " - "layer = (int) [ 1, 2 ], " - "rate = (int) [ 8000, 48000 ], " - "channels = (int) [ 1, 2 ], " "parsed= (boolean) true", - GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM, - "audio/x-raw-int, " - "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, " - "signed = (boolean) true, endianness = (int) " - G_STRINGIFY (G_LITTLE_ENDIAN) -}; - -static void -gst_dshowaudiodec_base_init (GstDshowAudioDecClass * klass) -{ - GstPadTemplate *src, *sink; - GstCaps *srccaps, *sinkcaps; - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - GstElementDetails details; - - klass->entry = tmp; - details.longname = g_strdup_printf ("DirectShow %s Decoder Wrapper", - tmp->element_longname); - details.klass = g_strdup ("Codec/Decoder/Audio"); - details.description = g_strdup_printf ("DirectShow %s Decoder Wrapper", - tmp->element_longname); - details.author = "Sebastien Moutte <sebastien@moutte.net>"; - gst_element_class_set_details (element_class, &details); - g_free (details.longname); - g_free (details.klass); - g_free (details.description); - - sinkcaps = gst_caps_from_string (tmp->sinkcaps); - gst_caps_set_simple (sinkcaps, - "block_align", GST_TYPE_INT_RANGE, 0, G_MAXINT, - "bitrate", GST_TYPE_INT_RANGE, 0, G_MAXINT, NULL); - - srccaps = gst_caps_from_string (tmp->srccaps); - - sink = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sinkcaps); - src = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, srccaps); - - /* register */ - gst_element_class_add_pad_template (element_class, src); - gst_element_class_add_pad_template (element_class, sink); -} - -static void -gst_dshowaudiodec_class_init (GstDshowAudioDecClass * klass) -{ - GObjectClass *gobject_class = G_OBJECT_CLASS (klass); - GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); - - gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_dshowaudiodec_dispose); - - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_dshowaudiodec_change_state); - - if (!parent_class) - parent_class = g_type_class_ref (GST_TYPE_ELEMENT); - - if (!dshowaudiodec_debug) { - GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0, - "Directshow filter audio decoder"); - } -} - -static void -gst_dshowaudiodec_init (GstDshowAudioDec * adec, - GstDshowAudioDecClass * adec_class) -{ - GstElementClass *element_class = GST_ELEMENT_GET_CLASS (adec); - HRESULT hr; - - /* setup pads */ - adec->sinkpad = - gst_pad_new_from_template (gst_element_class_get_pad_template - (element_class, "sink"), "sink"); - - gst_pad_set_setcaps_function (adec->sinkpad, gst_dshowaudiodec_sink_setcaps); - gst_pad_set_event_function (adec->sinkpad, gst_dshowaudiodec_sink_event); - gst_pad_set_chain_function (adec->sinkpad, gst_dshowaudiodec_chain); - gst_element_add_pad (GST_ELEMENT (adec), adec->sinkpad); - - adec->srcpad = - gst_pad_new_from_template (gst_element_class_get_pad_template - (element_class, "src"), "src"); - gst_element_add_pad (GST_ELEMENT (adec), adec->srcpad); - - adec->srcfilter = NULL; - adec->gstdshowsrcfilter = NULL; - adec->decfilter = NULL; - adec->sinkfilter = NULL; - adec->filtergraph = NULL; - adec->mediafilter = NULL; - adec->timestamp = GST_CLOCK_TIME_NONE; - adec->segment = gst_segment_new (); - adec->setup = FALSE; - adec->depth = 0; - adec->bitrate = 0; - adec->block_align = 0; - adec->channels = 0; - adec->rate = 0; - adec->layer = 0; - adec->codec_data = NULL; - - adec->last_ret = GST_FLOW_OK; - - hr = CoInitialize (0); - if (SUCCEEDED (hr)) { - adec->comInitialized = TRUE; - } -} - -static void -gst_dshowaudiodec_dispose (GObject * object) -{ - GstDshowAudioDec *adec = (GstDshowAudioDec *) (object); - - if (adec->segment) { - gst_segment_free (adec->segment); - adec->segment = NULL; - } - - if (adec->codec_data) { - gst_buffer_unref (adec->codec_data); - adec->codec_data = NULL; - } - - if (adec->comInitialized) { - CoUninitialize (); - adec->comInitialized = FALSE; - } - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - - -static GstStateChangeReturn -gst_dshowaudiodec_change_state (GstElement * element, GstStateChange transition) -{ - GstDshowAudioDec *adec = (GstDshowAudioDec *) (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - if (!gst_dshowaudiodec_create_graph_and_filters (adec)) - return GST_STATE_CHANGE_FAILURE; - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - break; - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - adec->depth = 0; - adec->bitrate = 0; - adec->block_align = 0; - adec->channels = 0; - adec->rate = 0; - adec->layer = 0; - if (adec->codec_data) { - gst_buffer_unref (adec->codec_data); - adec->codec_data = NULL; - } - break; - case GST_STATE_CHANGE_READY_TO_NULL: - if (!gst_dshowaudiodec_destroy_graph_and_filters (adec)) - return GST_STATE_CHANGE_FAILURE; - break; - default: - break; - } - - return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); -} - -static gboolean -gst_dshowaudiodec_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - gboolean ret = FALSE; - GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad); - GstStructure *s = gst_caps_get_structure (caps, 0); - const GValue *v = NULL; - - adec->timestamp = GST_CLOCK_TIME_NONE; - - /* read data, only rate and channels are needed */ - if (!gst_structure_get_int (s, "rate", &adec->rate) || - !gst_structure_get_int (s, "channels", &adec->channels)) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("error getting audio specs from caps"), (NULL)); - goto end; - } - - gst_structure_get_int (s, "depth", &adec->depth); - gst_structure_get_int (s, "bitrate", &adec->bitrate); - gst_structure_get_int (s, "block_align", &adec->block_align); - gst_structure_get_int (s, "layer", &adec->layer); - - if (adec->codec_data) { - gst_buffer_unref (adec->codec_data); - adec->codec_data = NULL; - } - - if ((v = gst_structure_get_value (s, "codec_data"))) - adec->codec_data = gst_buffer_ref (gst_value_get_buffer (v)); - - if (adec->layer != 1 && adec->layer != 2) { - /* setup dshow graph for all formats except for - * MPEG-1 layer 1 and 2 for which we need negociate - * in _chain function. - */ - ret = gst_dshowaudiodec_setup_graph (adec); - } - - ret = TRUE; -end: - gst_object_unref (adec); - - return ret; -} - -static GstFlowReturn -gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer) -{ - GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad); - gboolean discount = FALSE; - - if (!adec->setup) { - if (adec->layer != 0) { - if (adec->codec_data) { - gst_buffer_unref (adec->codec_data); - adec->codec_data = NULL; - } - /* extract the 3 bytes of MPEG-1 audio frame header */ - adec->codec_data = gst_buffer_create_sub (buffer, 1, 3); - } - - /* setup dshow graph */ - if (!gst_dshowaudiodec_setup_graph (adec)) { - adec->last_ret = GST_FLOW_ERROR; - goto beach; - } - } - - if (!adec->gstdshowsrcfilter) { - /* we are not setup */ - adec->last_ret = GST_FLOW_WRONG_STATE; - goto beach; - } - - if (GST_FLOW_IS_FATAL (adec->last_ret)) { - GST_DEBUG_OBJECT (adec, "last decoding iteration generated a fatal error " - "%s", gst_flow_get_name (adec->last_ret)); - goto beach; - } - - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, "chain (size %d)=> pts %" - GST_TIME_FORMAT " stop %" GST_TIME_FORMAT, - GST_BUFFER_SIZE (buffer), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer) + - GST_BUFFER_DURATION (buffer))); - - /* if the incoming buffer has discont flag set => flush decoder data */ - if (buffer && GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) { - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, - "this buffer has a DISCONT flag (%" GST_TIME_FORMAT "), flushing", - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); - gst_dshowaudiodec_flush (adec); - discount = TRUE; - } - - /* push the buffer to the directshow decoder */ - IGstDshowInterface_gst_push_buffer (adec->gstdshowsrcfilter, - GST_BUFFER_DATA (buffer), GST_BUFFER_TIMESTAMP (buffer), - GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer), - GST_BUFFER_SIZE (buffer), discount); - -beach: - gst_buffer_unref (buffer); - gst_object_unref (adec); - return adec->last_ret; -} - -static gboolean -gst_dshowaudiodec_push_buffer (byte * buffer, long size, byte * src_object, - UINT64 dshow_start, UINT64 dshow_stop) -{ - GstDshowAudioDec *adec = (GstDshowAudioDec *) src_object; - GstBuffer *out_buf = NULL; - gboolean in_seg = FALSE; - gint64 buf_start, buf_stop; - gint64 clip_start = 0, clip_stop = 0; - size_t start_offset = 0, stop_offset = size; - - if (!GST_CLOCK_TIME_IS_VALID (adec->timestamp)) { - adec->timestamp = dshow_start; - } - - buf_start = adec->timestamp; - buf_stop = adec->timestamp + (dshow_stop - dshow_start); - - /* save stop position to start next buffer with it */ - adec->timestamp = buf_stop; - - /* check if this buffer is in our current segment */ - in_seg = gst_segment_clip (adec->segment, GST_FORMAT_TIME, - buf_start, buf_stop, &clip_start, &clip_stop); - - /* if the buffer is out of segment do not push it downstream */ - if (!in_seg) { - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, - "buffer is out of segment, start %" GST_TIME_FORMAT " stop %" - GST_TIME_FORMAT, GST_TIME_ARGS (buf_start), GST_TIME_ARGS (buf_stop)); - return FALSE; - } - - /* buffer is in our segment allocate a new out buffer and clip it if needed */ - - /* allocate a new buffer for raw audio */ - adec->last_ret = gst_pad_alloc_buffer (adec->srcpad, GST_BUFFER_OFFSET_NONE, - size, GST_PAD_CAPS (adec->srcpad), &out_buf); - if (!out_buf) { - GST_CAT_ERROR_OBJECT (dshowaudiodec_debug, adec, - "can't not allocate a new GstBuffer"); - return FALSE; - } - - /* set buffer properties */ - GST_BUFFER_TIMESTAMP (out_buf) = buf_start; - GST_BUFFER_DURATION (out_buf) = buf_stop - buf_start; - memcpy (GST_BUFFER_DATA (out_buf), buffer, - MIN (size, GST_BUFFER_SIZE (out_buf))); - - /* we have to remove some heading samples */ - if (clip_start > buf_start) { - start_offset = (size_t) gst_util_uint64_scale_int (clip_start - buf_start, - adec->rate, GST_SECOND) * adec->depth / 8 * adec->channels; - } - /* we have to remove some trailing samples */ - if (clip_stop < buf_stop) { - stop_offset = (size_t) gst_util_uint64_scale_int (buf_stop - clip_stop, - adec->rate, GST_SECOND) * adec->depth / 8 * adec->channels; - } - - /* truncating */ - if ((start_offset != 0) || (stop_offset != (size_t) size)) { - GstBuffer *subbuf = gst_buffer_create_sub (out_buf, start_offset, - stop_offset - start_offset); - - if (subbuf) { - gst_buffer_set_caps (subbuf, GST_PAD_CAPS (adec->srcpad)); - gst_buffer_unref (out_buf); - out_buf = subbuf; - } - } - - GST_BUFFER_TIMESTAMP (out_buf) = clip_start; - GST_BUFFER_DURATION (out_buf) = clip_stop - clip_start; - - /* replace the saved stop position by the clipped one */ - adec->timestamp = clip_stop; - - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, - "push_buffer (size %d)=> pts %" GST_TIME_FORMAT " stop %" GST_TIME_FORMAT - " duration %" GST_TIME_FORMAT, size, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf) + - GST_BUFFER_DURATION (out_buf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (out_buf))); - - adec->last_ret = gst_pad_push (adec->srcpad, out_buf); - - return TRUE; -} - -static gboolean -gst_dshowaudiodec_sink_event (GstPad * pad, GstEvent * event) -{ - gboolean ret = TRUE; - GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_STOP:{ - gst_dshowaudiodec_flush (adec); - ret = gst_pad_event_default (pad, event); - break; - } - case GST_EVENT_NEWSEGMENT: - { - GstFormat format; - gdouble rate; - gint64 start, stop, time; - gboolean update; - - gst_event_parse_new_segment (event, &update, &rate, &format, &start, - &stop, &time); - - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, - "received new segment from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT, - GST_TIME_ARGS (start), GST_TIME_ARGS (stop)); - - if (update) { - GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, - "closing current segment flushing.."); - gst_dshowaudiodec_flush (adec); - } - - /* save the new segment in our local current segment */ - gst_segment_set_newsegment (adec->segment, update, rate, format, start, - stop, time); - - ret = gst_pad_event_default (pad, event); - break; - } - default: - ret = gst_pad_event_default (pad, event); - break; - } - - gst_object_unref (adec); - - return ret; -} - -static gboolean -gst_dshowaudiodec_flush (GstDshowAudioDec * adec) -{ - if (!adec->gstdshowsrcfilter) - return FALSE; - - /* flush dshow decoder and reset timestamp */ - IGstDshowInterface_gst_flush (adec->gstdshowsrcfilter); - adec->timestamp = GST_CLOCK_TIME_NONE; - - return TRUE; -} - - -static gboolean -gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec) -{ - gboolean ret = FALSE; - GstDshowAudioDecClass *klass = - (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec); - HRESULT hres; - gint size = 0; - GstCaps *out; - AM_MEDIA_TYPE output_mediatype, input_mediatype; - WAVEFORMATEX *input_format = NULL, output_format; - IPin *output_pin = NULL, *input_pin = NULL; - IGstDshowInterface *gstdshowinterface = NULL; - CodecEntry *codec_entry = klass->entry; - - if (adec->layer != 0) { - if (adec->layer == 1 || adec->layer == 2) { - /* for MPEG-1 layer 1 or 2 we have to release the current - * MP3 decoder and create an instance of MPEG Audio Decoder - */ - IBaseFilter_Release (adec->decfilter); - adec->decfilter = NULL; - codec_entry = audio_mpeg_1_2; - gst_dshow_find_filter (codec_entry->input_majortype, - codec_entry->input_subtype, - codec_entry->output_majortype, - codec_entry->output_subtype, - codec_entry->preferred_filter_substring, &adec->decfilter); - IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder"); - } else { - /* mp3 doesn't need to negotiate with MPEG1WAVEFORMAT */ - adec->layer = 0; - } - } - - /* set mediatype on fakesrc filter output pin */ - memset (&input_mediatype, 0, sizeof (AM_MEDIA_TYPE)); - input_mediatype.majortype = codec_entry->input_majortype; - input_mediatype.subtype = codec_entry->input_subtype; - input_mediatype.bFixedSizeSamples = TRUE; - input_mediatype.bTemporalCompression = FALSE; - if (adec->block_align) - input_mediatype.lSampleSize = adec->block_align; - else - input_mediatype.lSampleSize = 8192; /* need to evaluate it dynamically */ - input_mediatype.formattype = FORMAT_WaveFormatEx; - - if (adec->layer != 0) { - MPEG1WAVEFORMAT *mpeg1_format; - BYTE b1, b2, b3; - gint samples, version, layer; - - size = sizeof (MPEG1WAVEFORMAT); - input_format = g_malloc0 (size); - input_format->cbSize = sizeof (MPEG1WAVEFORMAT) - sizeof (WAVEFORMATEX); - mpeg1_format = (MPEG1WAVEFORMAT *) input_format; - - /* initialize header bytes */ - b1 = *GST_BUFFER_DATA (adec->codec_data); - b2 = *(GST_BUFFER_DATA (adec->codec_data) + 1); - b3 = *(GST_BUFFER_DATA (adec->codec_data) + 2); - - /* fill MPEG1WAVEFORMAT using header */ - input_format->wFormatTag = WAVE_FORMAT_MPEG; - mpeg1_format->wfx.nChannels = 2; - switch (b3 >> 6) { - case 0x00: - mpeg1_format->fwHeadMode = ACM_MPEG_STEREO; - break; - case 0x01: - mpeg1_format->fwHeadMode = ACM_MPEG_JOINTSTEREO; - break; - case 0x02: - mpeg1_format->fwHeadMode = ACM_MPEG_DUALCHANNEL; - break; - case 0x03: - mpeg1_format->fwHeadMode = ACM_MPEG_SINGLECHANNEL; - mpeg1_format->wfx.nChannels = 1; - break; - } - - mpeg1_format->fwHeadModeExt = (WORD) (1 << (b3 >> 4)); - mpeg1_format->wHeadEmphasis = (WORD) ((b3 & 0x03) + 1); - mpeg1_format->fwHeadFlags = (WORD) (((b2 & 1) ? ACM_MPEG_PRIVATEBIT : 0) + - ((b3 & 8) ? ACM_MPEG_COPYRIGHT : 0) + - ((b3 & 4) ? ACM_MPEG_ORIGINALHOME : 0) + - ((b1 & 1) ? ACM_MPEG_PROTECTIONBIT : 0) + ACM_MPEG_ID_MPEG1); - - layer = (b1 >> 1) & 3; - switch (layer) { - case 1: - mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER3; - layer = 3; - break; - case 2: - mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER2; - break; - case 3: - mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER1; - layer = 1; - break; - }; - - version = ((b1 >> 3) & 1) ? 0 : 1; - if (layer == 1) { - samples = 384; - } else { - if (version == 1) { - samples = 576; - } else { - samples = 1152; - } - } - mpeg1_format->wfx.nBlockAlign = (WORD) samples; - mpeg1_format->wfx.nSamplesPerSec = adec->rate; - mpeg1_format->dwHeadBitrate = mpeg_bitrates[version][layer - 1][b2 >> 4]; - mpeg1_format->wfx.nAvgBytesPerSec = mpeg1_format->dwHeadBitrate / 8; - } else { - size = sizeof (WAVEFORMATEX) + - (adec->codec_data ? GST_BUFFER_SIZE (adec->codec_data) : 0); - input_format = g_malloc0 (size); - if (adec->codec_data) { /* Codec data is appended after our header */ - memcpy (((guchar *) input_format) + sizeof (WAVEFORMATEX), - GST_BUFFER_DATA (adec->codec_data), - GST_BUFFER_SIZE (adec->codec_data)); - input_format->cbSize = GST_BUFFER_SIZE (adec->codec_data); - } - - input_format->wFormatTag = codec_entry->format; - input_format->nChannels = adec->channels; - input_format->nSamplesPerSec = adec->rate; - input_format->nAvgBytesPerSec = adec->bitrate / 8; - input_format->nBlockAlign = adec->block_align; - input_format->wBitsPerSample = adec->depth; - } - - input_mediatype.cbFormat = size; - input_mediatype.pbFormat = (BYTE *) input_format; - - hres = IBaseFilter_QueryInterface (adec->srcfilter, &IID_IGstDshowInterface, - (void **) &gstdshowinterface); - if (hres != S_OK || !gstdshowinterface) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get IGstDshowInterface interface from dshow fakesrc filter (error=%d)", - hres), (NULL)); - goto end; - } - - /* save a reference to IGstDshowInterface to use it processing functions */ - if (!adec->gstdshowsrcfilter) { - adec->gstdshowsrcfilter = gstdshowinterface; - IBaseFilter_AddRef (adec->gstdshowsrcfilter); - } - - IGstDshowInterface_gst_set_media_type (gstdshowinterface, &input_mediatype); - IGstDshowInterface_Release (gstdshowinterface); - gstdshowinterface = NULL; - - /* connect our fake source to decoder */ - gst_dshow_get_pin_from_filter (adec->srcfilter, PINDIR_OUTPUT, &output_pin); - if (!output_pin) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get output pin from our directshow fakesrc filter"), (NULL)); - goto end; - } - gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_INPUT, &input_pin); - if (!input_pin) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get input pin from decoder filter"), (NULL)); - goto end; - } - - hres = - IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin, - NULL); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't connect fakesrc with decoder (error=%d)", hres), (NULL)); - goto end; - } - - IPin_Release (input_pin); - IPin_Release (output_pin); - input_pin = NULL; - output_pin = NULL; - - if (!gst_dshowaudiodec_get_filter_settings (adec)) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get audio depth from decoder"), (NULL)); - goto end; - } - - /* set mediatype on fake sink input pin */ - memset (&output_format, 0, sizeof (WAVEFORMATEX)); - output_format.wFormatTag = WAVE_FORMAT_PCM; - output_format.wBitsPerSample = adec->depth; - output_format.nChannels = adec->channels; - output_format.nBlockAlign = adec->channels * (adec->depth / 8); - output_format.nSamplesPerSec = adec->rate; - output_format.nAvgBytesPerSec = output_format.nBlockAlign * adec->rate; - - memset (&output_mediatype, 0, sizeof (AM_MEDIA_TYPE)); - output_mediatype.majortype = codec_entry->output_majortype; - output_mediatype.subtype = codec_entry->output_subtype; - output_mediatype.bFixedSizeSamples = TRUE; - output_mediatype.bTemporalCompression = FALSE; - output_mediatype.lSampleSize = output_format.nBlockAlign; - output_mediatype.formattype = FORMAT_WaveFormatEx; - output_mediatype.cbFormat = sizeof (WAVEFORMATEX); - output_mediatype.pbFormat = (char *) &output_format; - - hres = IBaseFilter_QueryInterface (adec->sinkfilter, &IID_IGstDshowInterface, - (void **) &gstdshowinterface); - if (hres != S_OK || !gstdshowinterface) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get IGstDshowInterface interface from dshow fakesink filter (error=%d)", - hres), (NULL)); - goto end; - } - - IGstDshowInterface_gst_set_media_type (gstdshowinterface, &output_mediatype); - IGstDshowInterface_gst_set_buffer_callback (gstdshowinterface, - gst_dshowaudiodec_push_buffer, (byte *) adec); - IGstDshowInterface_Release (gstdshowinterface); - gstdshowinterface = NULL; - - /* negotiate output */ - out = gst_caps_from_string (codec_entry->srccaps); - gst_caps_set_simple (out, - "width", G_TYPE_INT, adec->depth, - "depth", G_TYPE_INT, adec->depth, - "rate", G_TYPE_INT, adec->rate, - "channels", G_TYPE_INT, adec->channels, NULL); - if (!gst_pad_set_caps (adec->srcpad, out)) { - gst_caps_unref (out); - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Failed to negotiate output"), (NULL)); - goto end; - } - gst_caps_unref (out); - - /* connect the decoder to our fake sink */ - gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT, &output_pin); - if (!output_pin) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get output pin from our decoder filter"), (NULL)); - goto end; - } - gst_dshow_get_pin_from_filter (adec->sinkfilter, PINDIR_INPUT, &input_pin); - if (!input_pin) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't get input pin from our directshow fakesink filter"), (NULL)); - goto end; - } - - hres = - IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin, - NULL); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't connect decoder with fakesink (error=%d)", hres), (NULL)); - goto end; - } - - hres = IMediaFilter_Run (adec->mediafilter, -1); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("Can't run the directshow graph (error=%d)", hres), (NULL)); - goto end; - } - - ret = TRUE; - adec->setup = TRUE; -end: - if (input_format) - g_free (input_format); - if (gstdshowinterface) - IGstDshowInterface_Release (gstdshowinterface); - if (input_pin) - IPin_Release (input_pin); - if (output_pin) - IPin_Release (output_pin); - - return ret; -} - -static gboolean -gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec) -{ - IPin *output_pin = NULL; - IEnumMediaTypes *enum_mediatypes = NULL; - HRESULT hres; - ULONG fetched; - BOOL ret = FALSE; - - if (!adec->decfilter) - return FALSE; - - if (!gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT, - &output_pin)) { - GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION, - ("failed getting ouput pin from the decoder"), (NULL)); - return FALSE; - } - - hres = IPin_EnumMediaTypes (output_pin, &enum_mediatypes); - if (hres == S_OK && enum_mediatypes) { - AM_MEDIA_TYPE *mediatype = NULL; - - IEnumMediaTypes_Reset (enum_mediatypes); - while (hres = - IEnumMoniker_Next (enum_mediatypes, 1, &mediatype, &fetched), - hres == S_OK) { - RPC_STATUS rpcstatus; - - if ((UuidCompare (&mediatype->subtype, &MEDIASUBTYPE_PCM, &rpcstatus) == 0 - && rpcstatus == RPC_S_OK) && - (UuidCompare (&mediatype->formattype, &FORMAT_WaveFormatEx, - &rpcstatus) == 0 && rpcstatus == RPC_S_OK)) { - WAVEFORMATEX *audio_info = (WAVEFORMATEX *) mediatype->pbFormat; - - adec->channels = audio_info->nChannels; - adec->depth = audio_info->wBitsPerSample; - adec->rate = audio_info->nSamplesPerSec; - ret = TRUE; - } - gst_dshow_free_mediatype (mediatype); - if (ret) - break; - } - IEnumMediaTypes_Release (enum_mediatypes); - } - if (output_pin) { - IPin_Release (output_pin); - } - - return ret; -} - -static gboolean -gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * adec) -{ - BOOL ret = FALSE; - HRESULT hres = S_FALSE; - GstDshowAudioDecClass *klass = - (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec); - - /* create the filter graph manager object */ - hres = CoCreateInstance (&CLSID_FilterGraph, NULL, CLSCTX_INPROC, - &IID_IFilterGraph, (LPVOID *) & adec->filtergraph); - if (hres != S_OK || !adec->filtergraph) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't create an instance of the directshow graph manager (error=%d)", - hres), (NULL)); - goto error; - } - - hres = IFilterGraph_QueryInterface (adec->filtergraph, &IID_IMediaFilter, - (void **) &adec->mediafilter); - if (hres != S_OK || !adec->mediafilter) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't get IMediacontrol interface from the graph manager (error=%d)", - hres), (NULL)); - goto error; - } - - /* create fake src filter */ - hres = CoCreateInstance (&CLSID_DshowFakeSrc, NULL, CLSCTX_INPROC, - &IID_IBaseFilter, (LPVOID *) & adec->srcfilter); - if (hres != S_OK || !adec->srcfilter) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't create an instance of the directshow fakesrc (error=%d)", hres), - (NULL)); - goto error; - } - - /* create decoder filter */ - if (!gst_dshow_find_filter (klass->entry->input_majortype, - klass->entry->input_subtype, - klass->entry->output_majortype, - klass->entry->output_subtype, - klass->entry->preferred_filter_substring, &adec->decfilter)) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't create an instance of the decoder filter"), (NULL)); - goto error; - } - - /* create fake sink filter */ - hres = CoCreateInstance (&CLSID_DshowFakeSink, NULL, CLSCTX_INPROC, - &IID_IBaseFilter, (LPVOID *) & adec->sinkfilter); - if (hres != S_OK || !adec->sinkfilter) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't create an instance of the directshow fakesink (error=%d)", - hres), (NULL)); - goto error; - } - - /* add filters to the graph */ - hres = IFilterGraph_AddFilter (adec->filtergraph, adec->srcfilter, L"src"); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't add fakesrc filter to the graph (error=%d)", hres), (NULL)); - goto error; - } - - hres = - IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder"); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't add decoder filter to the graph (error=%d)", hres), (NULL)); - goto error; - } - - hres = IFilterGraph_AddFilter (adec->filtergraph, adec->sinkfilter, L"sink"); - if (hres != S_OK) { - GST_ELEMENT_ERROR (adec, STREAM, FAILED, - ("Can't add fakesink filter to the graph (error=%d)", hres), (NULL)); - goto error; - } - - return TRUE; - -error: - if (adec->srcfilter) { - IBaseFilter_Release (adec->srcfilter); - adec->srcfilter = NULL; - } - if (adec->decfilter) { - IBaseFilter_Release (adec->decfilter); - adec->decfilter = NULL; - } - if (adec->sinkfilter) { - IBaseFilter_Release (adec->sinkfilter); - adec->sinkfilter = NULL; - } - if (adec->mediafilter) { - IMediaFilter_Release (adec->mediafilter); - adec->mediafilter = NULL; - } - if (adec->filtergraph) { - IFilterGraph_Release (adec->filtergraph); - adec->filtergraph = NULL; - } - - return FALSE; -} - -static gboolean -gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec * adec) -{ - if (adec->mediafilter) { - IMediaFilter_Stop (adec->mediafilter); - } - - if (adec->gstdshowsrcfilter) { - IGstDshowInterface_Release (adec->gstdshowsrcfilter); - adec->gstdshowsrcfilter = NULL; - } - if (adec->srcfilter) { - if (adec->filtergraph) - IFilterGraph_RemoveFilter (adec->filtergraph, adec->srcfilter); - IBaseFilter_Release (adec->srcfilter); - adec->srcfilter = NULL; - } - if (adec->decfilter) { - if (adec->filtergraph) - IFilterGraph_RemoveFilter (adec->filtergraph, adec->decfilter); - IBaseFilter_Release (adec->decfilter); - adec->decfilter = NULL; - } - if (adec->sinkfilter) { - if (adec->filtergraph) - IFilterGraph_RemoveFilter (adec->filtergraph, adec->sinkfilter); - IBaseFilter_Release (adec->sinkfilter); - adec->sinkfilter = NULL; - } - if (adec->mediafilter) { - IMediaFilter_Release (adec->mediafilter); - adec->mediafilter = NULL; - } - if (adec->filtergraph) { - IFilterGraph_Release (adec->filtergraph); - adec->filtergraph = NULL; - } - - adec->setup = FALSE; - - return TRUE; -} - -gboolean -dshow_adec_register (GstPlugin * plugin) -{ - GTypeInfo info = { - sizeof (GstDshowAudioDecClass), - (GBaseInitFunc) gst_dshowaudiodec_base_init, - NULL, - (GClassInitFunc) gst_dshowaudiodec_class_init, - NULL, - NULL, - sizeof (GstDshowAudioDec), - 0, - (GInstanceInitFunc) gst_dshowaudiodec_init, - }; - gint i; - HRESULT hr; - - GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0, - "Directshow filter audio decoder"); - - hr = CoInitialize (0); - for (i = 0; i < sizeof (audio_dec_codecs) / sizeof (CodecEntry); i++) { - GType type; - - if (gst_dshow_find_filter (audio_dec_codecs[i].input_majortype, - audio_dec_codecs[i].input_subtype, - audio_dec_codecs[i].output_majortype, - audio_dec_codecs[i].output_subtype, - audio_dec_codecs[i].preferred_filter_substring, NULL)) { - - GST_CAT_DEBUG (dshowaudiodec_debug, "Registering %s", - audio_dec_codecs[i].element_name); - - tmp = &audio_dec_codecs[i]; - type = - g_type_register_static (GST_TYPE_ELEMENT, - audio_dec_codecs[i].element_name, &info, 0); - if (!gst_element_register (plugin, audio_dec_codecs[i].element_name, - GST_RANK_PRIMARY, type)) { - return FALSE; - } - GST_CAT_DEBUG (dshowaudiodec_debug, "Registered %s", - audio_dec_codecs[i].element_name); - } else { - GST_CAT_DEBUG (dshowaudiodec_debug, - "Element %s not registered (the format is not supported by the system)", - audio_dec_codecs[i].element_name); - } - } - - if (SUCCEEDED (hr)) - CoUninitialize (); - - return TRUE; -} |