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authorAlessandro Decina <alessandro@nnva.org>2008-07-08 17:41:55 +0000
committerMichael Smith <msmith@xiph.org>2008-07-08 17:41:55 +0000
commit50eafabdd5a1e73642122cc0febe539de2f32668 (patch)
tree2eb61b91406c785070bff4c77423a8b69cce8639 /sys/dshowdecwrapper
parent7399002e2bf8cdae51298ff427ded637655543e3 (diff)
downloadgstreamer-plugins-bad-50eafabdd5a1e73642122cc0febe539de2f32668.tar.gz
sys/dshowdecwrapper/: Add AAC, AC3 to handled codecs.
Original commit message from CVS: Based on patch by: Alessandro Decina <alessandro@nnva.org> * sys/dshowdecwrapper/gstdshowaudiodec.c: * sys/dshowdecwrapper/gstdshowdecwrapper.h: * sys/dshowdecwrapper/gstdshowvideodec.c: Add AAC, AC3 to handled codecs. Fix handling of flush events. Improve debug/error output. Fix a number of typos in comments and variable names.
Diffstat (limited to 'sys/dshowdecwrapper')
-rw-r--r--sys/dshowdecwrapper/gstdshowaudiodec.c77
-rw-r--r--sys/dshowdecwrapper/gstdshowdecwrapper.h2
-rw-r--r--sys/dshowdecwrapper/gstdshowvideodec.c4
3 files changed, 63 insertions, 20 deletions
diff --git a/sys/dshowdecwrapper/gstdshowaudiodec.c b/sys/dshowdecwrapper/gstdshowaudiodec.c
index b7be7ac6f..cd2179a98 100644
--- a/sys/dshowdecwrapper/gstdshowaudiodec.c
+++ b/sys/dshowdecwrapper/gstdshowaudiodec.c
@@ -79,9 +79,9 @@ 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 */
+/* global variable */
+static const long bitrates[2][3][16] = {
+ /* mpeg 1 */
{
/* one list per layer 1-3 */
{0, 32000, 48000, 56000, 64000, 80000, 96000, 112000, 128000, 144000,
@@ -91,7 +91,7 @@ const long bitrates[2][3][16] = {
{0, 8000, 16000, 24000, 32000, 40000, 48000, 56000, 64000, 80000, 96000,
112000, 128000, 144000, 160000, 0},
},
- /* version 1 */
+ /* mpeg 2 */
{
/* one list per layer 1-3 */
{0, 32000, 64000, 96000, 128000, 160000, 192000, 224000, 256000,
@@ -112,6 +112,9 @@ const long bitrates[2][3][16] = {
#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 }}
+#define GUID_MEDIASUBTYPE_AAC {0x000000FF, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xAA, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_AC3 {0x00002000, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xAA, 0x00, 0x38, 0x9b, 0x71 }}
+
static const CodecEntry audio_dec_codecs[] = {
{"dshowadec_wma1",
@@ -176,7 +179,7 @@ static const CodecEntry audio_dec_codecs[] = {
},
{"dshowadec_mpeg1",
"MPEG-1 Layer 1,2,3 Audio",
- "MPEG Layer-3 Decoder",
+ "MPEG Layer-3 Decoder, MP3 Decoder DMO",
0x00000055,
GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_MP3,
"audio/mpeg, "
@@ -189,13 +192,37 @@ static const CodecEntry audio_dec_codecs[] = {
"width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
"signed = (boolean) true, endianness = (int) "
G_STRINGIFY (G_LITTLE_ENDIAN)
- }
+ },
+ {"dshowadec_aac",
+ "AAC decoder",
+ "ffdshow",
+ 0x000000FF,
+ GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_AAC,
+ "audio/mpeg, mpegversion = { 2, 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_ac3",
+ "AC3 decoder",
+ NULL,
+ 0x00002000,
+ GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_AC3,
+ "audio/x-ac3",
+ 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.
+ * The problem is that gstreamer doesn't care about caps like layer when connecting pads.
* So I've only one element handling mpeg audio in the public codecs map and
- * when it's loaded for mp3, I'm releasing mpeg audio decoder and replace it by
+ * when it's loaded for mp3, I release the mpeg audio decoder and replace it by
* the one described in this private map.
*/
static const CodecEntry audio_mpeg_1_2[] = { "dshowadec_mpeg_1_2",
@@ -307,7 +334,6 @@ gst_dshowaudiodec_init (GstDshowAudioDec * adec,
adec->rate = 0;
adec->layer = 0;
adec->codec_data = NULL;
-
adec->last_ret = GST_FLOW_OK;
CoInitializeEx (NULL, COINIT_MULTITHREADED);
@@ -582,6 +608,7 @@ gst_dshowaudiodec_sink_event (GstPad * pad, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:{
gst_dshowaudiodec_flush (adec);
+ adec->last_ret = GST_FLOW_OK;
ret = gst_pad_event_default (pad, event);
break;
}
@@ -650,6 +677,7 @@ gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
IPin *output_pin = NULL, *input_pin = NULL;
IGstDshowInterface *gstdshowinterface = NULL;
CodecEntry *codec_entry = klass->entry;
+ char err_buf[1024];
if (adec->layer != 0) {
if (adec->layer == 1 || adec->layer == 2) {
@@ -663,10 +691,10 @@ gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
codec_entry->input_subtype,
codec_entry->output_majortype,
codec_entry->output_subtype,
- codec_entry->prefered_filter_substring, &adec->decfilter);
+ codec_entry->preferred_filter_substring, &adec->decfilter);
IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder");
} else {
- /* mp3 don't need to negociate with MPEG1WAVEFORMAT */
+ /* mp3 doesn't need to negotiate with MPEG1WAVEFORMAT */
adec->layer = 0;
}
}
@@ -812,8 +840,13 @@ gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin,
NULL);
if (hres != S_OK) {
+ if (!FormatMessage (FORMAT_MESSAGE_FROM_SYSTEM,
+ 0, hres, 0, err_buf, 1024, NULL))
+ err_buf[0] = 0;
+
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
- ("Can't connect fakesrc with decoder (error=%d)", hres), (NULL));
+ ("Can't connect fakesrc with decoder (error=%d %s)", hres, err_buf),
+ (NULL));
goto end;
}
@@ -858,7 +891,7 @@ gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
IGstDshowInterface_gst_set_media_type (gstdshowinterface, &output_mediatype);
IGstDshowInterface_gst_set_buffer_callback (gstdshowinterface,
- gst_dshowaudiodec_push_buffer, (byte *) adec);
+ (byte *) gst_dshowaudiodec_push_buffer, (byte *) adec);
IGstDshowInterface_Release (gstdshowinterface);
gstdshowinterface = NULL;
@@ -895,15 +928,25 @@ gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin,
NULL);
if (hres != S_OK) {
+ if (!FormatMessage (FORMAT_MESSAGE_FROM_SYSTEM,
+ 0, hres, 0, err_buf, 1024, NULL))
+ err_buf[0] = 0;
+
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
- ("Can't connect decoder with fakesink (error=%d)", hres), (NULL));
+ ("Can't connect decoder with fakesink (error=%d %s)", hres, err_buf),
+ (NULL));
goto end;
}
hres = IMediaFilter_Run (adec->mediafilter, -1);
if (hres != S_OK) {
+ if (!FormatMessage (FORMAT_MESSAGE_FROM_SYSTEM,
+ 0, hres, 0, err_buf, 1024, NULL))
+ err_buf[0] = 0;
+
GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
- ("Can't run the directshow graph (error=%d)", hres), (NULL));
+ ("Can't run the directshow graph (error=%d %s)", hres, err_buf),
+ (NULL));
goto end;
}
@@ -1017,7 +1060,7 @@ gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * adec)
klass->entry->input_subtype,
klass->entry->output_majortype,
klass->entry->output_subtype,
- klass->entry->prefered_filter_substring, &adec->decfilter)) {
+ klass->entry->preferred_filter_substring, &adec->decfilter)) {
GST_ELEMENT_ERROR (adec, STREAM, FAILED,
("Can't create an instance of the decoder filter"), (NULL));
goto error;
@@ -1153,7 +1196,7 @@ dshow_adec_register (GstPlugin * plugin)
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)) {
+ audio_dec_codecs[i].preferred_filter_substring, NULL)) {
GST_CAT_DEBUG (dshowaudiodec_debug, "Registering %s",
audio_dec_codecs[i].element_name);
diff --git a/sys/dshowdecwrapper/gstdshowdecwrapper.h b/sys/dshowdecwrapper/gstdshowdecwrapper.h
index f912e18e7..3e7042c86 100644
--- a/sys/dshowdecwrapper/gstdshowdecwrapper.h
+++ b/sys/dshowdecwrapper/gstdshowdecwrapper.h
@@ -58,7 +58,7 @@
typedef struct _CodecEntry {
gchar *element_name;
gchar *element_longname;
- gchar *prefered_filter_substring;
+ gchar *preferred_filter_substring;
gint32 format;
GUID input_majortype;
GUID input_subtype;
diff --git a/sys/dshowdecwrapper/gstdshowvideodec.c b/sys/dshowdecwrapper/gstdshowvideodec.c
index c31bb9454..812b32c44 100644
--- a/sys/dshowdecwrapper/gstdshowvideodec.c
+++ b/sys/dshowdecwrapper/gstdshowvideodec.c
@@ -978,7 +978,7 @@ gst_dshowvideodec_create_graph_and_filters (GstDshowVideoDec * vdec)
klass->entry->input_subtype,
klass->entry->output_majortype,
klass->entry->output_subtype,
- klass->entry->prefered_filter_substring, &vdec->decfilter)) {
+ klass->entry->preferred_filter_substring, &vdec->decfilter)) {
GST_ELEMENT_ERROR (vdec, STREAM, FAILED, ("Can't create an instance "
"of the decoder filter"), (NULL));
goto error;
@@ -1111,7 +1111,7 @@ dshow_vdec_register (GstPlugin * plugin)
video_dec_codecs[i].input_subtype,
video_dec_codecs[i].output_majortype,
video_dec_codecs[i].output_subtype,
- video_dec_codecs[i].prefered_filter_substring, NULL)) {
+ video_dec_codecs[i].preferred_filter_substring, NULL)) {
GST_CAT_DEBUG (dshowvideodec_debug, "Registering %s",
video_dec_codecs[i].element_name);