ext/ffmpeg/gstffmpegcodecmap.*: Add mapping for EAC3 and QCELP audio codecs.

Original commit message from CVS:
* 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.

4 files changed, 125 insertions, 20 deletions

diff --git a/ChangeLog b/ChangeLog
index 38614b3..f6c28f1 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,18 @@
+2008-12-17  Edward Hervey  <edward.hervey@collabora.co.uk>
+
+	* 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  <ensonic@users.sf.net>
 
 	* configure.ac:
diff --git a/ext/ffmpeg/gstffmpegcodecmap.c b/ext/ffmpeg/gstffmpegcodecmap.c
index 1d896df..75b7b77 100644
--- a/ext/ffmpeg/gstffmpegcodecmap.c
+++ b/ext/ffmpeg/gstffmpegcodecmap.c
@@ -322,6 +322,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
           rates = l_rates;
           break;
         }
+        case CODEC_ID_EAC3:
         case CODEC_ID_AC3:
         {
           const static gint l_rates[] = { 48000, 44100, 32000 };
@@ -361,6 +362,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
      * 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 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
       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);
       break;
@@ -986,6 +993,10 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
           gst_ff_aud_caps_new (context, codec_id, "audio/x-truespeech", NULL);
       break;
 
+    case CODEC_ID_QCELP:
+      caps = gst_ff_aud_caps_new (context, codec_id, "audio/qcelp", NULL);
+      break;
+
     case CODEC_ID_WS_VQA:
     case CODEC_ID_IDCIN:
     case CODEC_ID_8BPS:
@@ -1529,6 +1540,7 @@ gst_ffmpeg_smpfmt_to_caps (enum SampleFormat sample_fmt,
   GstCaps *caps = NULL;
 
   int bpp = 0;
+  gboolean integer = TRUE;
   gboolean signedness = FALSE;
 
   switch (sample_fmt) {
@@ -1537,16 +1549,36 @@ gst_ffmpeg_smpfmt_to_caps (enum SampleFormat sample_fmt,
       bpp = 16;
       break;
 
+    case SAMPLE_FMT_S32:
+      signedness = TRUE;
+      bpp = 32;
+      break;
+
+    case SAMPLE_FMT_FLT:
+      integer = FALSE;
+      bpp = 32;
+      break;
+
+    case SAMPLE_FMT_DBL:
+      integer = FALSE;
+      bpp = 64;
+      break;
     default:
       /* .. */
       break;
   }
 
   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",
+          "endianness", G_TYPE_INT, G_BYTE_ORDER,
+          "width", G_TYPE_INT, bpp, NULL);
+    }
   }
 
   if (caps != NULL) {
@@ -1603,7 +1635,7 @@ gst_ffmpeg_codectype_to_caps (enum CodecType codec_type,
 
         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 @@ gst_ffmpeg_caps_to_smpfmt (const GstCaps * caps,
   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 */
+    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 ((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 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
   } 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;
     audio = TRUE;
@@ -2821,6 +2872,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
   } else if (!strcmp (mimetype, "audio/AMR-WB")) {
     id = CODEC_ID_AMR_WB;
     audio = TRUE;
+  } else if (!strcmp (mimetype, "audio/qcelp")) {
+    id = CODEC_ID_QCELP;
+    audio = TRUE;
   } else if (!strcmp (mimetype, "video/x-h264")) {
     id = CODEC_ID_H264;
     video = TRUE;
@@ -3236,3 +3290,28 @@ gst_ffmpeg_avpicture_fill (AVPicture * picture,
 
   return 0;
 }
+
+gint
+av_smp_format_depth (enum SampleFormat smp_fmt)
+{
+  gint depth = -1;
+  switch (smp_fmt) {
+    case SAMPLE_FMT_U8:
+      depth = 1;
+      break;
+    case SAMPLE_FMT_S16:
+      depth = 2;
+      break;
+    case SAMPLE_FMT_S32:
+    case SAMPLE_FMT_FLT:
+      depth = 4;
+      break;
+    case SAMPLE_FMT_DBL:
+      depth = 8;
+      break;
+    default:
+      GST_ERROR ("UNHANDLED SAMPLE FORMAT !");
+      break;
+  }
+  return depth;
+}
diff --git a/ext/ffmpeg/gstffmpegcodecmap.h b/ext/ffmpeg/gstffmpegcodecmap.h
index cf98216..46c5171 100644
--- a/ext/ffmpeg/gstffmpegcodecmap.h
+++ b/ext/ffmpeg/gstffmpegcodecmap.h
@@ -163,4 +163,7 @@ gst_ffmpeg_time_gst_to_ff (guint64 time, AVRational base)
 void 
 gst_ffmpeg_init_pix_fmt_info();
 
+gint
+av_smp_format_depth(enum SampleFormat smp_fmt);
+
 #endif /* __GST_FFMPEG_CODECMAP_H__ */
diff --git a/ext/ffmpeg/gstffmpegdec.c b/ext/ffmpeg/gstffmpegdec.c
index a70136b..7a14d2e 100644
--- a/ext/ffmpeg/gstffmpegdec.c
+++ b/ext/ffmpeg/gstffmpegdec.c
@@ -67,6 +67,7 @@ struct _GstFFMpegDec
     {
       gint channels;
       gint samplerate;
+      gint depth;
     } audio;
   } format;
   gboolean waiting_for_key;
@@ -587,6 +588,7 @@ gst_ffmpegdec_open (GstFFMpegDec * ffmpegdec)
     case CODEC_TYPE_AUDIO:
       ffmpegdec->format.audio.samplerate = 0;
       ffmpegdec->format.audio.channels = 0;
+      ffmpegdec->format.audio.depth = 0;
       break;
     default:
       break;
@@ -707,8 +709,7 @@ gst_ffmpegdec_setcaps (GstPad * pad, GstCaps * caps)
         GST_DEBUG_OBJECT (ffmpegdec, "enabled direct rendering");
         ffmpegdec->current_dr = TRUE;
       }
-    }
-    else {
+    } else {
       GST_DEBUG_OBJECT (ffmpegdec, "direct rendering not supported");
     }
   }
@@ -1075,16 +1076,22 @@ gst_ffmpegdec_negotiate (GstFFMpegDec * ffmpegdec)
       ffmpegdec->format.video.pix_fmt = ffmpegdec->context->pix_fmt;
       break;
     case CODEC_TYPE_AUDIO:
+    {
+      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;
+    }
       break;
     default:
       break;
@@ -1738,7 +1745,7 @@ clip_audio_buffer (GstFFMpegDec * dec, GstBuffer * buf, GstClockTime in_ts,
     /* 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 @@ clip_audio_buffer (GstFFMpegDec * dec, GstBuffer * buf, GstClockTime in_ts,
     /* 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 stop to %" GST_TIME_FORMAT " %"
         G_GINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff);
@@ -1824,7 +1831,8 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
      *  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:%"