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gstreamer/sys/dshowdecwrapper/gstdshowaudiodec.cpp

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/*
* GStreamer DirectShow codecs wrapper
* Copyright <2006, 2007, 2008, 2009, 2010> Fluendo <support@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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdshowaudiodec.h"
#include <mmreg.h>
#include <dmoreg.h>
#include <wmcodecdsp.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (dshowaudiodec_debug);
#define GST_CAT_DEFAULT dshowaudiodec_debug
#define gst_dshowaudiodec_parent_class parent_class
G_DEFINE_TYPE(GstDshowAudioDec, gst_dshowaudiodec, GST_TYPE_ELEMENT)
static void gst_dshowaudiodec_finalize (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, GstObject *parent, GstBuffer * buffer);
static gboolean gst_dshowaudiodec_sink_event (GstPad * pad, GstObject *parent, 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;
guint64 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 */
out_buf = gst_buffer_new_and_alloc(size);
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;
if (gst_buffer_fill(out_buf, 0, pBuffer, size) != size) {
gst_buffer_unref (out_buf);
GST_WARNING_OBJECT (mDec, "unable to fill output buffer");
goto done;
}
/* 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_copy_region (out_buf, GST_BUFFER_COPY_ALL,
start_offset, stop_offset - start_offset);
if (subbuf) {
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;
}
int AudioFakeSink::GetBufferSize()
{
IMemAllocator *allocator = NULL;
if (m_pInputPin) {
allocator = m_pInputPin->Allocator();
if(allocator) {
ALLOCATOR_PROPERTIES props;
allocator->GetProperties(&props);
return props.cbBuffer;
}
}
return 0;
}
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);
const AudioCodecEntry *tmp;
gpointer qdata;
gchar *longname, *description;
qdata = g_type_get_qdata (G_OBJECT_CLASS_TYPE (klass), DSHOW_CODEC_QDATA);
/* element details */
tmp = audiodec_class->entry = (AudioCodecEntry *) qdata;
longname = g_strdup_printf ("DirectShow %s Decoder Wrapper",
tmp->element_longname);
description = g_strdup_printf ("DirectShow %s Decoder Wrapper",
tmp->element_longname);
gst_element_class_set_metadata(element_class, longname, "Codec/Decoder/Audio", description,
"Sebastien Moutte <sebastien@moutte.net>");
g_free (longname);
g_free (description);
sinkcaps = gst_caps_from_string (tmp->sinkcaps);
srccaps = gst_caps_from_string (
"audio/x-raw,"
"format = (string)" GST_AUDIO_FORMATS_ALL ","
"rate = (int)[1, MAX],"
"channels = (int)[1, MAX],"
"layout = (string)interleaved");
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);
if (sinkcaps)
gst_caps_unref(sinkcaps);
if (srccaps)
gst_caps_unref(srccaps);
}
static void
gst_dshowaudiodec_class_init (GstDshowAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->finalize = gst_dshowaudiodec_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dshowaudiodec_change_state);
parent_class = (GstElementClass *) g_type_class_peek_parent (klass);
}
static void
gst_dshowaudiodec_com_thread (GstDshowAudioDec * adec)
{
HRESULT res;
g_mutex_lock (&adec->com_init_lock);
/* Initialize COM with a MTA for this process. This thread will
* be the first one to enter the apartement and the last one to leave
* it, unitializing COM properly */
res = CoInitializeEx (0, COINIT_MULTITHREADED);
if (res == S_FALSE)
GST_WARNING_OBJECT (adec, "COM has been already initialized in the same process");
else if (res == RPC_E_CHANGED_MODE)
GST_WARNING_OBJECT (adec, "The concurrency model of COM has changed.");
else
GST_INFO_OBJECT (adec, "COM intialized succesfully");
adec->comInitialized = TRUE;
/* Signal other threads waiting on this condition that COM was initialized */
g_cond_signal (&adec->com_initialized);
g_mutex_unlock (&adec->com_init_lock);
/* Wait until the unitialize condition is met to leave the COM apartement */
g_mutex_lock (&adec->com_deinit_lock);
g_cond_wait (&adec->com_uninitialize, &adec->com_deinit_lock);
CoUninitialize ();
GST_INFO_OBJECT (adec, "COM unintialized succesfully");
adec->comInitialized = FALSE;
g_cond_signal (&adec->com_uninitialized);
g_mutex_unlock (&adec->com_deinit_lock);
}
static void
gst_dshowaudiodec_init (GstDshowAudioDec * adec)
{
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_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;
g_mutex_init(&adec->com_init_lock);
g_mutex_init(&adec->com_deinit_lock);
g_cond_init(&adec->com_initialized);
g_cond_init(&adec->com_uninitialize);
g_cond_init(&adec->com_uninitialized);
g_mutex_lock (&adec->com_init_lock);
/* create the COM initialization thread */
g_thread_new ("COM init thread", (GThreadFunc)gst_dshowaudiodec_com_thread,
adec);
/* wait until the COM thread signals that COM has been initialized */
g_cond_wait (&adec->com_initialized, &adec->com_init_lock);
g_mutex_unlock (&adec->com_init_lock);
}
static void
gst_dshowaudiodec_finalize (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;
}
/* signal the COM thread that it sould uninitialize COM */
if (adec->comInitialized) {
g_mutex_lock (&adec->com_deinit_lock);
g_cond_signal (&adec->com_uninitialize);
g_cond_wait (&adec->com_uninitialized, &adec->com_deinit_lock);
g_mutex_unlock (&adec->com_deinit_lock);
}
g_mutex_clear (&adec->com_init_lock);
g_mutex_clear (&adec->com_deinit_lock);
g_cond_clear (&adec->com_initialized);
g_cond_clear (&adec->com_uninitialize);
g_cond_clear (&adec->com_uninitialized);
G_OBJECT_CLASS (parent_class)->finalize (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, GstObject *parent, GstBuffer *buffer)
{
GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
GstMapInfo map;
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_FLUSHING;
goto beach;
}
if (adec->last_ret != GST_FLOW_OK) {
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_get_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 */
gst_buffer_map(buffer, &map, GST_MAP_READ);
adec->fakesrc->GetOutputPin()->PushBuffer (
map.data, GST_BUFFER_TIMESTAMP (buffer),
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer),
map.size, (bool)discont);
gst_buffer_unmap(buffer, &map);
beach:
gst_buffer_unref (buffer);
gst_object_unref (adec);
return adec->last_ret;
}
static gboolean
gst_dshowaudiodec_sink_event (GstPad * pad, GstObject *parent, GstEvent * event)
{
gboolean ret = TRUE;
GstDshowAudioDec *adec = (GstDshowAudioDec *) parent;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:{
GstCaps *caps;
gst_event_parse_caps(event, &caps);
ret = gst_dshowaudiodec_sink_setcaps(pad, caps);
break;
}
case GST_EVENT_FLUSH_STOP:{
gst_dshowaudiodec_flush (adec);
ret = gst_pad_event_default (pad, parent, event);
break;
}
case GST_EVENT_SEGMENT:{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
"received new segment from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT,
GST_TIME_ARGS (segment->start), GST_TIME_ARGS (segment->stop));
/* save the new segment in our local current segment */
gst_segment_copy_into(segment, adec->segment);
ret = gst_pad_event_default (pad, parent, event);
break;
}
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
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_get_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;
}
else {
format = (WAVEFORMATEX *)g_malloc0 (size);
if (adec->codec_data) { /* Codec data is appended after our header */
gsize codec_size = gst_buffer_get_size(adec->codec_data);
gst_buffer_extract(adec->codec_data, 0, ((guchar *) format) + sizeof (WAVEFORMATEX),
codec_size);
format->cbSize = codec_size;
}
}
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)
{
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 = NULL;
AM_MEDIA_TYPE *output_mediatype = NULL;
AM_MEDIA_TYPE *input_mediatype = NULL;
IPinPtr output_pin = NULL;
IPinPtr input_pin = NULL;
const AudioCodecEntry *codec_entry = klass->entry;
IBaseFilterPtr srcfilter;
IBaseFilterPtr sinkfilter;
GstAudioInfo audio_info;
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);
gst_audio_info_init(&audio_info);
gst_audio_info_set_format(&audio_info,
gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, adec->depth, adec->depth),
adec->rate, adec->channels, NULL);
outcaps = gst_audio_info_to_caps(&audio_info);
if (!gst_pad_set_caps (adec->srcpad, outcaps)) {
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
("Failed to negotiate output"), (NULL));
goto end;
}
/* 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 (outcaps)
gst_caps_unref(outcaps);
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)
{
IPinPtr output_pin;
IEnumMediaTypesPtr 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);
IBaseFilterPtr srcfilter;
IBaseFilterPtr 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.CreateInstance (
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) {
IBaseFilterPtr 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) {
IBaseFilterPtr 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;
IBaseFilterPtr 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);
type = g_type_register_static (GST_TYPE_ELEMENT,
audio_dec_codecs[i].element_name, &info, (GTypeFlags)0);
g_type_set_qdata (type, DSHOW_CODEC_QDATA, (gpointer) (audio_dec_codecs + i));
if (!gst_element_register (plugin, audio_dec_codecs[i].element_name,
GST_RANK_MARGINAL, 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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