mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-30 21:51:09 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/sys/dshowdecwrapper/gstdshowaudiodec.c
Sebastian Dröge 32c43f884e sys/dshowdecwrapper/: Chain up dispose methods properly. Fixes bug #515970. 
Original commit message from CVS:
* sys/dshowdecwrapper/gstdshowaudiodec.c:
(gst_dshowaudiodec_dispose):
* sys/dshowdecwrapper/gstdshowvideodec.c:
(gst_dshowvideodec_dispose):
Chain up dispose methods properly. Fixes bug #515970.
2008-02-12 11:08:27 +00:00

1170 lines
39 KiB
C
Raw Blame History

/*
* 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);
/* gobal variable */
const long bitrates[2][3][16] = {
/* version 0 */
{
/* 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},
},
/* version 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},
}
};
#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 don'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'm releasing 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);
/* 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;
CoInitializeEx (NULL, COINIT_MULTITHREADED);
}
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;
}
CoUninitialize ();
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)
{
GstFlowReturn ret = GST_FLOW_OK;
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)) {
return GST_FLOW_ERROR;
}
}
if (!adec->gstdshowsrcfilter) {
/* we are not setup */
ret = GST_FLOW_WRONG_STATE;
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 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 */
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_SIZE (out_buf) = size;
GST_BUFFER_TIMESTAMP (out_buf) = buf_start;
GST_BUFFER_DURATION (out_buf) = buf_stop - buf_start;
memcpy (GST_BUFFER_DATA (out_buf), buffer, size);
/* 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)));
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;
}
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->prefered_filter_substring, &adec->decfilter);
IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder");
} else {
/* mp3 don't need to negociate 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;
if (layer == 1) {
samples = 384;
} else {
if (version == 0) {
samples = 576;
} else {
samples = 1152;
}
}
mpeg1_format->wfx.nBlockAlign = (WORD) samples;
mpeg1_format->wfx.nSamplesPerSec = adec->rate;
mpeg1_format->dwHeadBitrate = 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:
gst_object_unref (adec);
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->prefered_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;
GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0,
"Directshow filter audio decoder");
CoInitializeEx (NULL, COINIT_MULTITHREADED);
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].prefered_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);
}
}
CoUninitialize ();
return TRUE;
}
Reference in a new issue View git blame Copy permalink