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gstreamer/sys/dshowdecwrapper/gstdshowaudiodec.cpp
Michael Smith d8d1d23471 dshowdecwrapper: Make audio decoders have SECONDARY rank. 
These wrappers are not ideal, and for many of these formats there are
better gstreamer elements available. So, make the rank SECONDARY. In
particular, the mp3 decoder on winXP doesn't work very well.
2009-02-10 12:26:25 -08:00

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/*
* 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>
#include <dmoreg.h>
#include <wmcodecdsp.h>
GST_DEBUG_CATEGORY_STATIC (dshowaudiodec_debug);
#define GST_CAT_DEFAULT dshowaudiodec_debug
GST_BOILERPLATE (GstDshowAudioDec, gst_dshowaudiodec, GstElement,
GST_TYPE_ELEMENT);
static const AudioCodecEntry *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);
/* 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, GstCaps *caps);
/* All the GUIDs we want are generated from the FOURCC like this */
#define GUID_MEDIASUBTYPE_FROM_FOURCC(fourcc) \
{ fourcc , 0x0000, 0x0010, \
{ 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
/* WMA we should always use the DMO */
static PreferredFilter preferred_wma_filters[] = {
{&CLSID_CWMADecMediaObject, &DMOCATEGORY_AUDIO_DECODER},
{0}
};
/* Prefer the Vista (DMO) decoder if present, otherwise the XP
* decoder (not a DMO), otherwise fallback to highest-merit */
static const GUID CLSID_XP_MP3_DECODER = {0x38BE3000, 0xDBF4, 0x11D0,
{0x86,0x0E,0x00,0xA0,0x24,0xCF,0xEF,0x6D}};
static PreferredFilter preferred_mp3_filters[] = {
{&CLSID_CMP3DecMediaObject, &DMOCATEGORY_AUDIO_DECODER},
{&CLSID_XP_MP3_DECODER},
{0}
};
/* MPEG 1/2: use the MPEG Audio Decoder filter */
static const GUID CLSID_WINDOWS_MPEG_AUDIO_DECODER =
{0x4A2286E0, 0x7BEF, 0x11CE,
{0x9B, 0xD9, 0x00, 0x00, 0xE2, 0x02, 0x59, 0x9C}};
static PreferredFilter preferred_mpegaudio_filters[] = {
{&CLSID_WINDOWS_MPEG_AUDIO_DECODER},
{0}
};
static const AudioCodecEntry audio_dec_codecs[] = {
{"dshowadec_wma1", "Windows Media Audio 7",
WAVE_FORMAT_MSAUDIO1,
"audio/x-wma, wmaversion = (int) 1",
preferred_wma_filters},
{"dshowadec_wma2", "Windows Media Audio 8",
WAVE_FORMAT_WMAUDIO2,
"audio/x-wma, wmaversion = (int) 2",
preferred_wma_filters},
{"dshowadec_wma3", "Windows Media Audio 9 Professional",
WAVE_FORMAT_WMAUDIO3,
"audio/x-wma, wmaversion = (int) 3",
preferred_wma_filters},
{"dshowadec_wma4", "Windows Media Audio 9 Lossless",
WAVE_FORMAT_WMAUDIO_LOSSLESS,
"audio/x-wma, wmaversion = (int) 4",
preferred_wma_filters},
{"dshowadec_wms", "Windows Media Audio Voice v9",
WAVE_FORMAT_WMAVOICE9,
"audio/x-wms",
preferred_wma_filters},
{"dshowadec_mp3", "MPEG Layer 3 Audio",
WAVE_FORMAT_MPEGLAYER3,
"audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int)3, "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ], "
"parsed= (boolean) true",
preferred_mp3_filters},
{"dshowadec_mpeg_1_2", "MPEG Layer 1,2 Audio",
WAVE_FORMAT_MPEG,
"audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 2 ], "
"rate = (int) [ 8000, 48000 ], "
"channels = (int) [ 1, 2 ], "
"parsed= (boolean) true",
preferred_mpegaudio_filters},
};
HRESULT AudioFakeSink::DoRenderSample(IMediaSample *pMediaSample)
{
GstBuffer *out_buf = NULL;
gboolean in_seg = FALSE;
GstClockTime buf_start, buf_stop;
gint64 clip_start = 0, clip_stop = 0;
guint start_offset = 0, stop_offset;
GstClockTime duration;
if(pMediaSample)
{
BYTE *pBuffer = NULL;
LONGLONG lStart = 0, lStop = 0;
long size = pMediaSample->GetActualDataLength();
pMediaSample->GetPointer(&pBuffer);
pMediaSample->GetTime(&lStart, &lStop);
if (!GST_CLOCK_TIME_IS_VALID (mDec->timestamp)) {
// Convert REFERENCE_TIME to GST_CLOCK_TIME
mDec->timestamp = (GstClockTime)lStart * 100;
}
duration = (lStop - lStart) * 100;
buf_start = mDec->timestamp;
buf_stop = mDec->timestamp + duration;
/* save stop position to start next buffer with it */
mDec->timestamp = buf_stop;
/* check if this buffer is in our current segment */
in_seg = gst_segment_clip (mDec->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_DEBUG_OBJECT (mDec,
"buffer is out of segment, start %" GST_TIME_FORMAT " stop %"
GST_TIME_FORMAT, GST_TIME_ARGS (buf_start), GST_TIME_ARGS (buf_stop));
goto done;
}
/* buffer is entirely or partially in-segment, so allocate a
* GstBuffer for output, and clip if required */
/* allocate a new buffer for raw audio */
mDec->last_ret = gst_pad_alloc_buffer (mDec->srcpad,
GST_BUFFER_OFFSET_NONE,
size,
GST_PAD_CAPS (mDec->srcpad), &out_buf);
if (!out_buf) {
GST_WARNING_OBJECT (mDec, "cannot allocate a new GstBuffer");
goto done;
}
/* set buffer properties */
GST_BUFFER_TIMESTAMP (out_buf) = buf_start;
GST_BUFFER_DURATION (out_buf) = duration;
memcpy (GST_BUFFER_DATA (out_buf), pBuffer,
MIN ((unsigned int)size, GST_BUFFER_SIZE (out_buf)));
/* we have to remove some heading samples */
if ((GstClockTime) clip_start > buf_start) {
start_offset = (guint)gst_util_uint64_scale_int (clip_start - buf_start,
mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
}
else
start_offset = 0;
/* we have to remove some trailing samples */
if ((GstClockTime) clip_stop < buf_stop) {
stop_offset = (guint)gst_util_uint64_scale_int (buf_stop - clip_stop,
mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
}
else
stop_offset = size;
/* 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 (mDec->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 */
mDec->timestamp = clip_stop;
GST_DEBUG_OBJECT (mDec,
"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)));
mDec->last_ret = gst_pad_push (mDec->srcpad, out_buf);
}
done:
return S_OK;
}
HRESULT AudioFakeSink::CheckMediaType(const CMediaType *pmt)
{
if(pmt != NULL)
{
/* The Vista MP3 decoder (and possibly others?) outputs an
* AM_MEDIA_TYPE with the wrong cbFormat. So, rather than using
* CMediaType.operator==, we implement a sufficient check ourselves.
* I think this is a bug in the MP3 decoder.
*/
if (IsEqualGUID (pmt->majortype, m_MediaType.majortype) &&
IsEqualGUID (pmt->subtype, m_MediaType.subtype) &&
IsEqualGUID (pmt->formattype, m_MediaType.formattype))
{
/* Types are the same at the top-level. Now, we need to compare
* the format blocks.
* We special case WAVEFORMATEX to not check that
* pmt->cbFormat == m_MediaType.cbFormat, though the actual format
* blocks must still be the same.
*/
if (pmt->formattype == FORMAT_WaveFormatEx) {
if (pmt->cbFormat >= sizeof (WAVEFORMATEX) &&
m_MediaType.cbFormat >= sizeof (WAVEFORMATEX))
{
WAVEFORMATEX *wf1 = (WAVEFORMATEX *)pmt->pbFormat;
WAVEFORMATEX *wf2 = (WAVEFORMATEX *)m_MediaType.pbFormat;
if (wf1->cbSize == wf2->cbSize &&
memcmp (wf1, wf2, sizeof(WAVEFORMATEX) + wf1->cbSize) == 0)
return S_OK;
}
}
else {
if (pmt->cbFormat == m_MediaType.cbFormat &&
pmt->cbFormat == 0 ||
(pmt->pbFormat != NULL && m_MediaType.pbFormat != NULL &&
memcmp (pmt->pbFormat, m_MediaType.pbFormat, pmt->cbFormat) == 0))
return S_OK;
}
}
}
return S_FALSE;
}
static void
gst_dshowaudiodec_base_init (gpointer klass)
{
GstDshowAudioDecClass *audiodec_class = (GstDshowAudioDecClass *)klass;
GstPadTemplate *src, *sink;
GstCaps *srccaps, *sinkcaps;
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstElementDetails details;
audiodec_class->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);
srccaps = gst_caps_from_string (
"audio/x-raw-int,"
"width = (int)[1, 32],"
"depth = (int)[1, 32],"
"rate = (int)[1, MAX],"
"channels = (int)[1, MAX],"
"signed = (boolean)true,"
"endianness = (int)" G_STRINGIFY(G_LITTLE_ENDIAN));
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 = (GstElementClass *)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->fakesrc = NULL;
adec->fakesink = NULL;
adec->decfilter = 0;
adec->filtergraph = 0;
adec->mediafilter = 0;
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));
ret = gst_dshowaudiodec_setup_graph (adec, caps);
end:
gst_object_unref (adec);
return ret;
}
static GstFlowReturn
gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer)
{
GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
bool discont = FALSE;
if (!adec->setup) {
/* we are not set up */
GST_WARNING_OBJECT (adec, "Decoder not set up, failing");
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);
discont = TRUE;
}
/* push the buffer to the directshow decoder */
adec->fakesrc->GetOutputPin()->PushBuffer (
GST_BUFFER_DATA (buffer), GST_BUFFER_TIMESTAMP (buffer),
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer),
GST_BUFFER_SIZE (buffer), (bool)discont);
beach:
gst_buffer_unref (buffer);
gst_object_unref (adec);
return adec->last_ret;
}
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->fakesrc)
return FALSE;
/* flush dshow decoder and reset timestamp */
adec->fakesrc->GetOutputPin()->Flush();
adec->timestamp = GST_CLOCK_TIME_NONE;
adec->last_ret = GST_FLOW_OK;
return TRUE;
}
static AM_MEDIA_TYPE *
dshowaudiodec_set_input_format (GstDshowAudioDec *adec, GstCaps *caps)
{
AM_MEDIA_TYPE *mediatype;
WAVEFORMATEX *format;
GstDshowAudioDecClass *klass =
(GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
const AudioCodecEntry *codec_entry = klass->entry;
int size;
mediatype = (AM_MEDIA_TYPE *)g_malloc0 (sizeof(AM_MEDIA_TYPE));
mediatype->majortype = MEDIATYPE_Audio;
GUID subtype = GUID_MEDIASUBTYPE_FROM_FOURCC (0x00000000);
subtype.Data1 = codec_entry->format;
mediatype->subtype = subtype;
mediatype->bFixedSizeSamples = TRUE;
mediatype->bTemporalCompression = FALSE;
if (adec->block_align)
mediatype->lSampleSize = adec->block_align;
else
mediatype->lSampleSize = 8192; /* need to evaluate it dynamically */
mediatype->formattype = FORMAT_WaveFormatEx;
/* We need this special behaviour for layers 1 and 2 (layer 3 uses a different
* decoder which doesn't need this */
if (adec->layer == 1 || adec->layer == 2) {
MPEG1WAVEFORMAT *mpeg1_format;
int samples, version;
GstStructure *structure = gst_caps_get_structure (caps, 0);
size = sizeof (MPEG1WAVEFORMAT);
format = (WAVEFORMATEX *)g_malloc0 (size);
format->cbSize = sizeof (MPEG1WAVEFORMAT) - sizeof (WAVEFORMATEX);
format->wFormatTag = WAVE_FORMAT_MPEG;
mpeg1_format = (MPEG1WAVEFORMAT *) format;
mpeg1_format->wfx.nChannels = adec->channels;
if (adec->channels == 2)
mpeg1_format->fwHeadMode = ACM_MPEG_STEREO;
else
mpeg1_format->fwHeadMode = ACM_MPEG_SINGLECHANNEL;
mpeg1_format->fwHeadModeExt = 0;
mpeg1_format->wHeadEmphasis = 0;
mpeg1_format->fwHeadFlags = 0;
switch (adec->layer) {
case 1:
mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER3;
break;
case 2:
mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER2;
break;
case 3:
mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER1;
break;
};
gst_structure_get_int (structure, "mpegaudioversion", &version);
if (adec->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 = 128000; /* This doesn't seem to matter */
mpeg1_format->wfx.nAvgBytesPerSec = mpeg1_format->dwHeadBitrate / 8;
}
else
{
size = sizeof (WAVEFORMATEX) +
(adec->codec_data ? GST_BUFFER_SIZE (adec->codec_data) : 0);
if (adec->layer == 3) {
MPEGLAYER3WAVEFORMAT *mp3format;
/* The WinXP mp3 decoder doesn't actually check the size of this structure,
* but requires that this be allocated and filled out (or we get obscure
* random crashes)
*/
size = sizeof (MPEGLAYER3WAVEFORMAT);
mp3format = (MPEGLAYER3WAVEFORMAT *)g_malloc0 (size);
format = (WAVEFORMATEX *)mp3format;
format->cbSize = MPEGLAYER3_WFX_EXTRA_BYTES;
mp3format->wID = MPEGLAYER3_ID_MPEG;
mp3format->fdwFlags = MPEGLAYER3_FLAG_PADDING_ISO; /* No idea what this means for a decoder */
/* The XP decoder divides by nBlockSize, so we must set this to a
non-zero value, but it doesn't matter what - this is meaningless
for VBR mp3 anyway */
mp3format->nBlockSize = 1;
mp3format->nFramesPerBlock = 1;
mp3format->nCodecDelay = 0;
/* The XP decoder also has problems with some MP3 frames. If it tries
* to decode one, then forever after it outputs silence.
* If we recognise such a frame, just skip decoding it.
*/
if (adec->decoder_is_xp_mp3)
adec->check_mp3_frames = TRUE;
}
else {
format = (WAVEFORMATEX *)g_malloc0 (size);
if (adec->codec_data) { /* Codec data is appended after our header */
memcpy (((guchar *) format) + sizeof (WAVEFORMATEX),
GST_BUFFER_DATA (adec->codec_data),
GST_BUFFER_SIZE (adec->codec_data));
format->cbSize = GST_BUFFER_SIZE (adec->codec_data);
}
}
format->wFormatTag = codec_entry->format;
format->nChannels = adec->channels;
format->nSamplesPerSec = adec->rate;
format->nAvgBytesPerSec = adec->bitrate / 8;
format->nBlockAlign = adec->block_align;
format->wBitsPerSample = adec->depth;
}
mediatype->cbFormat = size;
mediatype->pbFormat = (BYTE *) format;
return mediatype;
}
static AM_MEDIA_TYPE *
dshowaudiodec_set_output_format (GstDshowAudioDec *adec)
{
AM_MEDIA_TYPE *mediatype;
WAVEFORMATEX *format;
GstDshowAudioDecClass *klass =
(GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
const AudioCodecEntry *codec_entry = klass->entry;
if (!gst_dshowaudiodec_get_filter_settings (adec)) {
return NULL;
}
format = (WAVEFORMATEX *)g_malloc0(sizeof (WAVEFORMATEX));
format->wFormatTag = WAVE_FORMAT_PCM;
format->wBitsPerSample = adec->depth;
format->nChannels = adec->channels;
format->nBlockAlign = adec->channels * (adec->depth / 8);
format->nSamplesPerSec = adec->rate;
format->nAvgBytesPerSec = format->nBlockAlign * adec->rate;
mediatype = (AM_MEDIA_TYPE *)g_malloc0(sizeof (AM_MEDIA_TYPE));
mediatype->majortype = MEDIATYPE_Audio;
GUID subtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
mediatype->subtype = subtype;
mediatype->bFixedSizeSamples = TRUE;
mediatype->bTemporalCompression = FALSE;
mediatype->lSampleSize = format->nBlockAlign;
mediatype->formattype = FORMAT_WaveFormatEx;
mediatype->cbFormat = sizeof (WAVEFORMATEX);
mediatype->pbFormat = (BYTE *)format;
return mediatype;
}
static void
dshowadec_free_mediatype (AM_MEDIA_TYPE *mediatype)
{
if (mediatype->pbFormat)
g_free (mediatype->pbFormat);
g_free (mediatype);
}
static gboolean
gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec, GstCaps *caps)
{
gboolean ret = FALSE;
GstDshowAudioDecClass *klass =
(GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
HRESULT hres;
GstCaps *outcaps;
AM_MEDIA_TYPE *output_mediatype = NULL;
AM_MEDIA_TYPE *input_mediatype = NULL;
CComPtr<IPin> output_pin;
CComPtr<IPin> input_pin;
const AudioCodecEntry *codec_entry = klass->entry;
CComQIPtr<IBaseFilter> srcfilter;
CComQIPtr<IBaseFilter> sinkfilter;
input_mediatype = dshowaudiodec_set_input_format (adec, caps);
adec->fakesrc->GetOutputPin()->SetMediaType (input_mediatype);
srcfilter = adec->fakesrc;
/* connect our fake source to decoder */
output_pin = gst_dshow_get_pin_from_filter (srcfilter, PINDIR_OUTPUT);
if (!output_pin) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't get output pin from our directshow fakesrc filter"), (NULL));
goto end;
}
input_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_INPUT);
if (!input_pin) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't get input pin from decoder filter"), (NULL));
goto end;
}
hres = adec->filtergraph->ConnectDirect (output_pin, input_pin,
NULL);
if (hres != S_OK) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't connect fakesrc with decoder (error=%x)", hres), (NULL));
goto end;
}
output_mediatype = dshowaudiodec_set_output_format (adec);
if (!output_mediatype) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't get audio output format from decoder"), (NULL));
goto end;
}
adec->fakesink->SetMediaType(output_mediatype);
outcaps = gst_caps_new_simple ("audio/x-raw-int",
"width", G_TYPE_INT, adec->depth,
"depth", G_TYPE_INT, adec->depth,
"rate", G_TYPE_INT, adec->rate,
"channels", G_TYPE_INT, adec->channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
NULL);
if (!gst_pad_set_caps (adec->srcpad, outcaps)) {
gst_caps_unref (outcaps);
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Failed to negotiate output"), (NULL));
goto end;
}
gst_caps_unref (outcaps);
/* connect the decoder to our fake sink */
output_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT);
if (!output_pin) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't get output pin from our decoder filter"), (NULL));
goto end;
}
sinkfilter = adec->fakesink;
input_pin = gst_dshow_get_pin_from_filter (sinkfilter, PINDIR_INPUT);
if (!input_pin) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't get input pin from our directshow fakesink filter"), (NULL));
goto end;
}
hres = adec->filtergraph->ConnectDirect(output_pin, input_pin, NULL);
if (hres != S_OK) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't connect decoder with fakesink (error=%x)", hres), (NULL));
goto end;
}
hres = adec->mediafilter->Run (-1);
if (hres != S_OK) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Can't run the directshow graph (error=%x)", hres), (NULL));
goto end;
}
ret = TRUE;
adec->setup = TRUE;
end:
if (input_mediatype)
dshowadec_free_mediatype (input_mediatype);
if (output_mediatype)
dshowadec_free_mediatype (output_mediatype);
return ret;
}
static gboolean
gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec)
{
CComPtr<IPin> output_pin;
CComPtr<IEnumMediaTypes> enum_mediatypes;
HRESULT hres;
ULONG fetched;
BOOL ret = FALSE;
if (adec->decfilter == 0)
return FALSE;
output_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT);
if (!output_pin) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("failed getting ouput pin from the decoder"), (NULL));
return FALSE;
}
hres = output_pin->EnumMediaTypes (&enum_mediatypes);
if (hres == S_OK && enum_mediatypes) {
AM_MEDIA_TYPE *mediatype = NULL;
enum_mediatypes->Reset();
while (!ret && enum_mediatypes->Next(1, &mediatype, &fetched) == S_OK)
{
if (IsEqualGUID (mediatype->subtype, MEDIASUBTYPE_PCM) &&
IsEqualGUID (mediatype->formattype, FORMAT_WaveFormatEx))
{
WAVEFORMATEX *audio_info = (WAVEFORMATEX *) mediatype->pbFormat;
adec->channels = audio_info->nChannels;
adec->depth = audio_info->wBitsPerSample;
adec->rate = audio_info->nSamplesPerSec;
ret = TRUE;
}
DeleteMediaType (mediatype);
}
}
return ret;
}
static gboolean
gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * adec)
{
HRESULT hres;
GstDshowAudioDecClass *klass =
(GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
CComQIPtr<IBaseFilter> srcfilter;
CComQIPtr<IBaseFilter> sinkfilter;
GUID insubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (klass->entry->format);
GUID outsubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
/* create the filter graph manager object */
hres = adec->filtergraph.CoCreateInstance (
CLSID_FilterGraph, NULL, CLSCTX_INPROC);
if (FAILED (hres)) {
GST_ELEMENT_ERROR (adec, STREAM, FAILED,
("Can't create an instance of the directshow graph manager (error=%d)",
hres), (NULL));
goto error;
}
hres = adec->filtergraph->QueryInterface (&adec->mediafilter);
if (FAILED (hres)) {
GST_WARNING_OBJECT (adec, "Can't QI filtergraph to mediafilter");
goto error;
}
/* create fake src filter */
adec->fakesrc = new FakeSrc();
/* Created with a refcount of zero, so increment that */
adec->fakesrc->AddRef();
/* create decoder filter */
adec->decfilter = gst_dshow_find_filter (MEDIATYPE_Audio,
insubtype,
MEDIATYPE_Audio,
outsubtype,
klass->entry->preferred_filters);
if (adec->decfilter == NULL) {
GST_ELEMENT_ERROR (adec, STREAM, FAILED,
("Can't create an instance of the decoder filter"), (NULL));
goto error;
}
/* create fake sink filter */
adec->fakesink = new AudioFakeSink(adec);
/* Created with a refcount of zero, so increment that */
adec->fakesink->AddRef();
/* add filters to the graph */
srcfilter = adec->fakesrc;
hres = adec->filtergraph->AddFilter (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 = adec->filtergraph->AddFilter(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;
}
sinkfilter = adec->fakesink;
hres = adec->filtergraph->AddFilter(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->fakesrc) {
adec->fakesrc->Release();
adec->fakesrc = NULL;
}
if (adec->fakesink) {
adec->fakesink->Release();
adec->fakesink = NULL;
}
adec->decfilter = 0;
adec->mediafilter = 0;
adec->filtergraph = 0;
return FALSE;
}
static gboolean
gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec * adec)
{
if (adec->mediafilter) {
adec->mediafilter->Stop();
}
if (adec->fakesrc) {
if (adec->filtergraph) {
CComQIPtr<IBaseFilter> filter = adec->fakesrc;
adec->filtergraph->RemoveFilter(filter);
}
adec->fakesrc->Release();
adec->fakesrc = NULL;
}
if (adec->decfilter) {
if (adec->filtergraph)
adec->filtergraph->RemoveFilter(adec->decfilter);
adec->decfilter = 0;
}
if (adec->fakesink) {
if (adec->filtergraph) {
CComQIPtr<IBaseFilter> filter = adec->fakesink;
adec->filtergraph->RemoveFilter(filter);
}
adec->fakesink->Release();
adec->fakesink = NULL;
}
adec->mediafilter = 0;
adec->filtergraph = 0;
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 (AudioCodecEntry); i++) {
GType type;
CComPtr<IBaseFilter> filter;
GUID insubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (audio_dec_codecs[i].format);
GUID outsubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
filter = gst_dshow_find_filter (MEDIATYPE_Audio,
insubtype,
MEDIATYPE_Audio,
outsubtype,
audio_dec_codecs[i].preferred_filters);
if (filter)
{
GST_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, (GTypeFlags)0);
if (!gst_element_register (plugin, audio_dec_codecs[i].element_name,
GST_RANK_SECONDARY, type)) {
return FALSE;
}
GST_CAT_DEBUG (dshowaudiodec_debug, "Registered %s",
audio_dec_codecs[i].element_name);
}
else {
GST_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;
}
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