bilboed gst-ffmpeg: gst-ffmpeg/ gst-ffmpeg/ext/ffmpeg/
[email protected] Wed, 17 Dec 2008 04:05:27 -0800 (PST)
| Newsgroups | gmane.comp.video.gstreamer.cvs |
|---|---|
| Message-ID | <[email protected]> |
CVS Root: /cvs/gstreamer
Module: gst-ffmpeg
Changes by: bilboed
Date: Wed Dec 17 2008 12:05:27 UTC
Log message:
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new),
(gst_ffmpeg_codecid_to_caps), (gst_ffmpeg_smpfmt_to_caps),
(gst_ffmpeg_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt),
(gst_ffmpeg_caps_to_codecid), (av_smp_format_depth):
* ext/ffmpeg/gstffmpegcodecmap.h:
Add mapping for EAC3 and QCELP audio codecs.
Add conversion functions for all available audo SampleFormat.
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open),
(gst_ffmpegdec_setcaps), (gst_ffmpegdec_negotiate),
(clip_audio_buffer), (gst_ffmpegdec_audio_frame):
Remove assumptions that we can only handle stereo 16bit signed integer
audio, and store the depth locally.
Modified files:
. : ChangeLog
ext/ffmpeg : gstffmpegcodecmap.c gstffmpegcodecmap.h
gstffmpegdec.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-ffmpeg/ChangeLog.diff?r1=1.537&r2=1.538
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegcodecmap.c.diff?r1=1.174&r2=1.175
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegcodecmap.h.diff?r1=1.26&r2=1.27
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegdec.c.diff?r1=1.216&r2=1.217
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-ffmpeg/ChangeLog,v
retrieving revision 1.537
retrieving revision 1.538
diff -u -d -r1.537 -r1.538
--- ChangeLog 16 Dec 2008 16:21:27 -0000 1.537
+++ ChangeLog 17 Dec 2008 12:05:11 -0000 1.538
@@ -1,3 +1,18 @@
+2008-12-17 Edward Hervey <[email protected]>
+
+ * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new),
+ (gst_ffmpeg_codecid_to_caps), (gst_ffmpeg_smpfmt_to_caps),
+ (gst_ffmpeg_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt),
+ (gst_ffmpeg_caps_to_codecid), (av_smp_format_depth):
+ * ext/ffmpeg/gstffmpegcodecmap.h:
+ Add mapping for EAC3 and QCELP audio codecs.
+ Add conversion functions for all available audo SampleFormat.
+ * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open),
+ (gst_ffmpegdec_setcaps), (gst_ffmpegdec_negotiate),
+ (clip_audio_buffer), (gst_ffmpegdec_audio_frame):
+ Remove assumptions that we can only handle stereo 16bit signed integer
+ audio, and store the depth locally.
2008-12-16 Stefan Kost <[email protected]>
* configure.ac:
Index: gstffmpegcodecmap.c
RCS file: /cvs/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegcodecmap.c,v
retrieving revision 1.174
retrieving revision 1.175
diff -u -d -r1.174 -r1.175
--- gstffmpegcodecmap.c 13 Dec 2008 08:01:45 -0000 1.174
+++ gstffmpegcodecmap.c 17 Dec 2008 12:05:12 -0000 1.175
@@ -322,6 +322,7 @@
rates = l_rates;
break;
}
+ case CODEC_ID_EAC3:
case CODEC_ID_AC3:
{
const static gint l_rates[] = { 48000, 44100, 32000 };
@@ -361,6 +362,7 @@
* they support, we whitelist them here. */
switch (codec_id) {
case CODEC_ID_AC3:
+ case CODEC_ID_EAC3:
case CODEC_ID_AAC:
case CODEC_ID_DTS:
maxchannels = 6;
@@ -560,6 +562,11 @@
caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-ac3", NULL);
break;
+ case CODEC_ID_EAC3:
+ /* FIXME: bitrate */
+ caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-eac3", NULL);
+ break;
case CODEC_ID_ATRAC3:
caps = gst_ff_aud_caps_new (context, codec_id, "audio/atrac3", NULL);
@@ -986,6 +993,10 @@
gst_ff_aud_caps_new (context, codec_id, "audio/x-truespeech", NULL);
+ case CODEC_ID_QCELP:
+ caps = gst_ff_aud_caps_new (context, codec_id, "audio/qcelp", NULL);
case CODEC_ID_WS_VQA:
case CODEC_ID_IDCIN:
case CODEC_ID_8BPS:
@@ -1529,6 +1540,7 @@
GstCaps *caps = NULL;
int bpp = 0;
+ gboolean integer = TRUE;
gboolean signedness = FALSE;
switch (sample_fmt) {
@@ -1537,16 +1549,36 @@
bpp = 16;
+ case SAMPLE_FMT_S32:
+ signedness = TRUE;
+ bpp = 32;
+ case SAMPLE_FMT_FLT:
+ integer = FALSE;
+ case SAMPLE_FMT_DBL:
+ bpp = 64;
default:
/* .. */
}
if (bpp) {
- caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-raw-int",
- "signed", G_TYPE_BOOLEAN, signedness,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "width", G_TYPE_INT, bpp, "depth", G_TYPE_INT, bpp, NULL);
+ if (integer) {
+ caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-raw-int",
+ "signed", G_TYPE_BOOLEAN, signedness,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "width", G_TYPE_INT, bpp, "depth", G_TYPE_INT, bpp, NULL);
+ } else {
+ caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-raw-float",
+ "width", G_TYPE_INT, bpp, NULL);
+ }
if (caps != NULL) {
@@ -1603,7 +1635,7 @@
ctx.channels = -1;
caps = gst_caps_new_empty ();
- for (i = 0; i <= SAMPLE_FMT_S16; i++) {
+ for (i = 0; i <= SAMPLE_FMT_DBL; i++) {
temp = gst_ffmpeg_smpfmt_to_caps (i, encode ? &ctx : NULL, codec_id);
if (temp != NULL) {
gst_caps_append (caps, temp);
@@ -1646,13 +1678,29 @@
if (!raw)
return;
- if (gst_structure_get_int (structure, "width", &width) &&
- gst_structure_get_int (structure, "depth", &depth) &&
- gst_structure_get_boolean (structure, "signed", &signedness) &&
- gst_structure_get_int (structure, "endianness", &endianness)) {
- if (width == 16 && depth == 16 &&
- endianness == G_BYTE_ORDER && signedness == TRUE) {
- context->sample_fmt = SAMPLE_FMT_S16;
+ if (!strcmp (gst_structure_get_name (structure), "audio/x-raw-float")) {
+ /* FLOAT */
+ if (gst_structure_get_int (structure, "width", &width) &&
+ gst_structure_get_int (structure, "endianness", &endianness)) {
+ if (endianness == G_BYTE_ORDER) {
+ if (width == 32)
+ context->sample_fmt = SAMPLE_FMT_FLT;
+ else if (width == 64)
+ context->sample_fmt = SAMPLE_FMT_DBL;
+ }
+ } else {
+ /* INT */
+ gst_structure_get_int (structure, "depth", &depth) &&
+ gst_structure_get_boolean (structure, "signed", &signedness) &&
+ gst_structure_get_int (structure, "endianness", &endianness)) {
+ if ((endianness == G_BYTE_ORDER) && (signedness == TRUE)) {
+ if ((width == 16) && (depth == 16))
+ context->sample_fmt = SAMPLE_FMT_S16;
+ else if ((width == 32) && (depth == 32))
+ context->sample_fmt = SAMPLE_FMT_S32;
}
}
@@ -2550,6 +2598,9 @@
} else if (!strcmp (mimetype, "audio/x-ac3")) {
id = CODEC_ID_AC3;
audio = TRUE;
+ } else if (!strcmp (mimetype, "audio/x-eac3")) {
+ id = CODEC_ID_EAC3;
+ audio = TRUE;
} else if (!strcmp (mimetype, "audio/atrac3")) {
id = CODEC_ID_ATRAC3;
@@ -2821,6 +2872,9 @@
} else if (!strcmp (mimetype, "audio/AMR-WB")) {
id = CODEC_ID_AMR_WB;
+ } else if (!strcmp (mimetype, "audio/qcelp")) {
+ id = CODEC_ID_QCELP;
} else if (!strcmp (mimetype, "video/x-h264")) {
id = CODEC_ID_H264;
video = TRUE;
@@ -3236,3 +3290,28 @@
return 0;
+gint
+av_smp_format_depth (enum SampleFormat smp_fmt)
+{
+ gint depth = -1;
+ switch (smp_fmt) {
+ case SAMPLE_FMT_U8:
+ depth = 1;
+ case SAMPLE_FMT_S16:
+ depth = 2;
+ depth = 4;
+ depth = 8;
+ default:
+ GST_ERROR ("UNHANDLED SAMPLE FORMAT !");
+ }
+ return depth;
+}
Index: gstffmpegcodecmap.h
RCS file: /cvs/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegcodecmap.h,v
retrieving revision 1.26
retrieving revision 1.27
diff -u -d -r1.26 -r1.27
--- gstffmpegcodecmap.h 13 Dec 2008 07:55:40 -0000 1.26
+++ gstffmpegcodecmap.h 17 Dec 2008 12:05:12 -0000 1.27
@@ -163,4 +163,7 @@
void
gst_ffmpeg_init_pix_fmt_info();
+av_smp_format_depth(enum SampleFormat smp_fmt);
#endif /* __GST_FFMPEG_CODECMAP_H__ */
Index: gstffmpegdec.c
RCS file: /cvs/gstreamer/gst-ffmpeg/ext/ffmpeg/gstffmpegdec.c,v
retrieving revision 1.216
retrieving revision 1.217
diff -u -d -r1.216 -r1.217
--- gstffmpegdec.c 3 Dec 2008 17:43:45 -0000 1.216
+++ gstffmpegdec.c 17 Dec 2008 12:05:12 -0000 1.217
@@ -67,6 +67,7 @@
{
gint channels;
gint samplerate;
+ gint depth;
} audio;
} format;
gboolean waiting_for_key;
@@ -587,6 +588,7 @@
case CODEC_TYPE_AUDIO:
ffmpegdec->format.audio.samplerate = 0;
ffmpegdec->format.audio.channels = 0;
+ ffmpegdec->format.audio.depth = 0;
@@ -707,8 +709,7 @@
GST_DEBUG_OBJECT (ffmpegdec, "enabled direct rendering");
ffmpegdec->current_dr = TRUE;
}
- }
- else {
GST_DEBUG_OBJECT (ffmpegdec, "direct rendering not supported");
@@ -1075,16 +1076,22 @@
ffmpegdec->format.video.pix_fmt = ffmpegdec->context->pix_fmt;
+ {
+ gint depth = av_smp_format_depth (ffmpegdec->context->sample_fmt);
if (ffmpegdec->format.audio.samplerate ==
ffmpegdec->context->sample_rate &&
- ffmpegdec->format.audio.channels == ffmpegdec->context->channels)
+ ffmpegdec->format.audio.channels == ffmpegdec->context->channels &&
+ ffmpegdec->format.audio.depth == depth)
return TRUE;
GST_DEBUG_OBJECT (ffmpegdec,
- "Renegotiating audio from %dHz@%dchannels to %dHz@%dchannels",
+ "Renegotiating audio from %dHz@%dchannels (%d) to %dHz@%dchannels (%d)",
ffmpegdec->format.audio.samplerate, ffmpegdec->format.audio.channels,
- ffmpegdec->context->sample_rate, ffmpegdec->context->channels);
+ ffmpegdec->format.audio.depth,
+ ffmpegdec->context->sample_rate, ffmpegdec->context->channels, depth);
ffmpegdec->format.audio.samplerate = ffmpegdec->context->sample_rate;
ffmpegdec->format.audio.channels = ffmpegdec->context->channels;
+ ffmpegdec->format.audio.depth = depth;
@@ -1738,7 +1745,7 @@
/* bring clipped time to bytes */
diff =
gst_util_uint64_scale_int (diff, dec->format.audio.samplerate,
- GST_SECOND) * (2 * dec->format.audio.channels);
+ GST_SECOND) * (dec->format.audio.depth * dec->format.audio.channels);
GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %"
G_GINT64_FORMAT " bytes", GST_TIME_ARGS (ctime), diff);
@@ -1750,7 +1757,7 @@
GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %"
G_GINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
@@ -1824,7 +1831,8 @@
* 1) calculate based on number of samples
*/
in_duration = gst_util_uint64_scale_int (have_data, GST_SECOND,
- 2 * ffmpegdec->context->channels * ffmpegdec->context->sample_rate);
+ ffmpegdec->format.audio.depth * ffmpegdec->format.audio.channels *
+ ffmpegdec->format.audio.samplerate);
GST_DEBUG_OBJECT (ffmpegdec,
"Buffer created. Size:%d , timestamp:%" GST_TIME_FORMAT " , duration:%"
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