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banshee r4780 - in trunk/banshee: . build/osx/targets

Author: eoinh
Date: Thu Nov  6 07:30:56 2008
New Revision: 4780
URL: http://svn.gnome.org/viewvc/banshee?rev=4780&view=rev

Log:
2008-11-05  Eoin Hennessy <eoin randomrules org>

	* build/osx/targets/gst-plugins-bad-qtwrapper-fixes.patch:
	Added a patch containing a number of fixes for qtwrapper issues in 
	gst-plugins-bad 0.10.9 backported from current cvs. Makes playback on 
	OSX considerably more stable.




Added:
   trunk/banshee/build/osx/targets/gst-plugins-bad-qtwrapper-fixes.patch
Modified:
   trunk/banshee/ChangeLog

Added: trunk/banshee/build/osx/targets/gst-plugins-bad-qtwrapper-fixes.patch
==============================================================================
--- (empty file)
+++ trunk/banshee/build/osx/targets/gst-plugins-bad-qtwrapper-fixes.patch	Thu Nov  6 07:30:56 2008
@@ -0,0 +1,417 @@
+diff --git sys/qtwrapper/Makefile.am sys/qtwrapper/Makefile.am
+index db08647..35a8337 100644
+--- sys/qtwrapper/Makefile.am
++++ sys/qtwrapper/Makefile.am
+@@ -14,7 +14,7 @@ libgstqtwrapper_la_SOURCES = \
+ libgstqtwrapper_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
+ libgstqtwrapper_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
+ libgstqtwrapper_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -Wl,-framework,CoreAudio,-framework,AudioToolbox,-framework,Carbon,-framework,QuickTime,-framework,QuartzCore
+-
++libgstqtwrapper_la_LIBTOOLFLAGS = --tag=disable-static
+ 
+ # headers we need but don't want installed
+ noinst_HEADERS = 		\
+diff --git sys/qtwrapper/audiodecoders.c sys/qtwrapper/audiodecoders.c
+index ed5c69c..1816a02 100644
+--- sys/qtwrapper/audiodecoders.c
++++ sys/qtwrapper/audiodecoders.c
+@@ -210,6 +210,49 @@ fill_indesc_generic (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc,
+   qtwrapper->indesc.mChannelsPerFrame = channels;
+ }
+ 
++static void
++fill_indesc_alac (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc,
++    gint rate, gint channels)
++{
++  clear_AudioStreamBasicDescription (&qtwrapper->indesc);
++  qtwrapper->indesc.mSampleRate = rate;
++  qtwrapper->indesc.mFormatID = fourcc;
++  qtwrapper->indesc.mChannelsPerFrame = channels;
++
++  // This has to be set, but the particular value doesn't seem to matter much
++  qtwrapper->indesc.mFramesPerPacket = 4096;
++}
++
++static gpointer
++make_alac_magic_cookie (GstBuffer * codec_data, gsize * len)
++{
++  guint8 *res;
++
++  if (GST_BUFFER_SIZE (codec_data) < 4)
++    return NULL;
++
++  *len = 20 + GST_BUFFER_SIZE (codec_data);
++  res = g_malloc0 (*len);
++
++  /* 12 first bytes are 'frma' (format) atom with 'alac' value */
++  GST_WRITE_UINT32_BE (res, 0xc);       /* Atom length: 12 bytes */
++  GST_WRITE_UINT32_LE (res + 4, QT_MAKE_FOURCC_BE ('f', 'r', 'm', 'a'));
++  GST_WRITE_UINT32_LE (res + 8, QT_MAKE_FOURCC_BE ('a', 'l', 'a', 'c'));
++
++  /* Write the codec_data, but with the first four bytes reversed (different
++     endianness). This is the 'alac' atom. */
++  GST_WRITE_UINT32_BE (res + 12,
++      GST_READ_UINT32_LE (GST_BUFFER_DATA (codec_data)));
++  memcpy (res + 16, GST_BUFFER_DATA (codec_data) + 4,
++      GST_BUFFER_SIZE (codec_data) - 4);
++
++  /* Terminator atom */
++  GST_WRITE_UINT32_BE (res + 12 + GST_BUFFER_SIZE (codec_data), 8);
++  GST_WRITE_UINT32_BE (res + 12 + GST_BUFFER_SIZE (codec_data) + 4, 0);
++
++  return res;
++}
++
+ static gpointer
+ make_samr_magic_cookie (GstBuffer * codec_data, gsize * len)
+ {
+@@ -234,7 +277,7 @@ make_samr_magic_cookie (GstBuffer * codec_data, gsize * len)
+   /* yes... we need to replace 'damr' by 'samr'. Blame Apple ! */
+   GST_WRITE_UINT8 (res + 26, 's');
+ 
+-  /* padding 8 bytes */
++  /* Terminator atom */
+   GST_WRITE_UINT32_BE (res + 40, 8);
+ 
+ #if DEBUG_DUMP
+@@ -259,6 +302,27 @@ write_len (guint8 * buf, int val)
+   return 4;
+ }
+ 
++static void
++aac_parse_codec_data (GstBuffer * codec_data, guint * channels)
++{
++  guint8 *data = GST_BUFFER_DATA (codec_data);
++  int codec_channels;
++
++  if (GST_BUFFER_SIZE (codec_data) < 2) {
++    GST_WARNING ("Cannot parse codec_data for channel count");
++    return;
++  }
++
++  codec_channels = (data[1] & 0x7f) >> 3;
++
++  if (*channels != codec_channels) {
++    GST_INFO ("Overwriting channels %d with %d", *channels, codec_channels);
++    *channels = codec_channels;
++  } else {
++    GST_INFO ("Retaining channel count %d", codec_channels);
++  }
++}
++
+ /* The AAC decoder requires the entire mpeg4 audio elementary stream 
+  * descriptor, which is the body (except the 4-byte version field) of
+  * the quicktime 'esds' atom. However, qtdemux only passes through the 
+@@ -360,6 +424,16 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+     codec_data = GST_BUFFER_CAST (gst_value_get_mini_object (value));
+   }
+ 
++  oclass = (QTWrapperAudioDecoderClass *) (G_OBJECT_GET_CLASS (qtwrapper));
++
++  if (codec_data
++      && oclass->componentSubType == QT_MAKE_FOURCC_LE ('m', 'p', '4', 'a')) {
++    /* QuickTime/iTunes creates AAC files with the wrong channel count in the header,
++       so parse that out of the codec data if we can.
++     */
++    aac_parse_codec_data (codec_data, (guint *) & channels);
++  }
++
+   /* If the quicktime demuxer gives us a full esds atom, use that instead of 
+    * the codec_data */
+   if ((value = gst_structure_get_value (s, "quicktime_esds"))) {
+@@ -375,7 +449,6 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+ 
+   GST_INFO_OBJECT (qtwrapper, "rate:%d, channels:%d", rate, channels);
+ 
+-  oclass = (QTWrapperAudioDecoderClass *) (G_OBJECT_GET_CLASS (qtwrapper));
+   GST_INFO_OBJECT (qtwrapper, "componentSubType is %" GST_FOURCC_FORMAT,
+       QT_FOURCC_ARGS (oclass->componentSubType));
+ 
+@@ -391,6 +464,9 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+       fill_indesc_samr (qtwrapper, oclass->componentSubType, channels);
+       rate = 8000;
+       break;
++    case QT_MAKE_FOURCC_LE ('a', 'l', 'a', 'c'):
++      fill_indesc_alac (qtwrapper, oclass->componentSubType, rate, channels);
++      break;
+     default:
+       fill_indesc_generic (qtwrapper, oclass->componentSubType, rate, channels);
+       break;
+@@ -433,6 +509,10 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+   if (status) {
+     GST_WARNING_OBJECT (qtwrapper,
+         "Error setting input description on SCAudio: %ld", status);
++
++    GST_ELEMENT_ERROR (qtwrapper, STREAM, NOT_IMPLEMENTED,
++        ("A QuickTime error occurred trying to decode this stream"),
++        ("QuickTime returned error status %lx", status));
+     goto beach;
+   }
+ 
+@@ -450,6 +530,9 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+       case QT_MAKE_FOURCC_LE ('s', 'a', 'm', 'r'):
+         magiccookie = make_samr_magic_cookie (codec_data, &len);
+         break;
++      case QT_MAKE_FOURCC_LE ('a', 'l', 'a', 'c'):
++        magiccookie = make_alac_magic_cookie (codec_data, &len);
++        break;
+       case QT_MAKE_FOURCC_LE ('m', 'p', '4', 'a'):
+         if (!have_esds) {
+           magiccookie = make_aac_magic_cookie (codec_data, &len);
+@@ -462,19 +545,22 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+         break;
+     }
+ 
+-    GST_LOG_OBJECT (qtwrapper, "Setting magic cookie %p of size %"
+-        G_GSIZE_FORMAT, magiccookie, len);
++    if (magiccookie) {
++      GST_LOG_OBJECT (qtwrapper, "Setting magic cookie %p of size %"
++          G_GSIZE_FORMAT, magiccookie, len);
+ 
+ #if DEBUG_DUMP
+-    gst_util_dump_mem (magiccookie, len);
++      gst_util_dump_mem (magiccookie, len);
+ #endif
+ 
+-    status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
+-        kQTSCAudioPropertyID_InputMagicCookie, len, magiccookie);
+-    if (status) {
+-      GST_WARNING_OBJECT (qtwrapper, "Error setting extra codec data: %ld",
+-          status);
+-      goto beach;
++      status =
++          QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
++          kQTSCAudioPropertyID_InputMagicCookie, len, magiccookie);
++      if (status) {
++        GST_WARNING_OBJECT (qtwrapper, "Error setting extra codec data: %ld",
++            status);
++        goto beach;
++      }
+     }
+   }
+ 
+@@ -507,6 +593,25 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+     }
+   }
+ 
++  qtwrapper->outdesc.mSampleRate = 0;   /* Use recommended; we read this out later */
++  qtwrapper->outdesc.mFormatID = kAudioFormatLinearPCM;
++  qtwrapper->outdesc.mFormatFlags = kAudioFormatFlagIsFloat;
++  qtwrapper->outdesc.mBytesPerPacket = 0;
++  qtwrapper->outdesc.mFramesPerPacket = 0;
++  qtwrapper->outdesc.mBytesPerFrame = 4 * channels;
++  qtwrapper->outdesc.mChannelsPerFrame = channels;
++  qtwrapper->outdesc.mBitsPerChannel = 32;
++  qtwrapper->outdesc.mReserved = 0;
++
++  status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
++      kQTSCAudioPropertyID_BasicDescription,
++      sizeof (qtwrapper->outdesc), &qtwrapper->outdesc);
++  if (status) {
++    GST_WARNING_OBJECT (qtwrapper, "Error setting output description: %ld",
++        status);
++    goto beach;
++  }
++
+   status = QTGetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
+       kQTSCAudioPropertyID_BasicDescription,
+       sizeof (qtwrapper->outdesc), &qtwrapper->outdesc, NULL);
+@@ -518,9 +623,9 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+     goto beach;
+   }
+ 
+-  if (qtwrapper->outdesc.mFormatID != kAudioFormatLinearPCM ||
+-      (qtwrapper->outdesc.mFormatFlags & kAudioFormatFlagIsFloat) !=
+-      kAudioFormatFlagIsFloat) {
++  if (qtwrapper->outdesc.mFormatID != kAudioFormatLinearPCM     /*||
++                                                                   (qtwrapper->outdesc.mFormatFlags & kAudioFormatFlagIsFloat) !=
++                                                                   kAudioFormatFlagIsFloat */ ) {
+     GST_WARNING_OBJECT (qtwrapper, "Output is not floating point PCM");
+     ret = FALSE;
+     goto beach;
+@@ -531,21 +636,21 @@ open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
+   GST_DEBUG_OBJECT (qtwrapper, "Output is %d Hz, %d channels",
+       qtwrapper->samplerate, qtwrapper->channels);
+ 
+-  /* Create output bufferlist, big enough for 50ms of audio */
++  /* Create output bufferlist, big enough for 200ms of audio */
+   GST_DEBUG_OBJECT (qtwrapper, "Allocating bufferlist for %d channels",
+       channels);
+   qtwrapper->bufferlist =
+       AllocateAudioBufferList (channels,
+-      qtwrapper->samplerate * qtwrapper->channels * 4 / 20);
++      qtwrapper->samplerate / 5 * qtwrapper->channels * 4);
+ 
+-  /* TODO: Figure out how the output format is determined, can we pick this? */
+-  /* Create output caps */
++  /* Create output caps matching the format the component is giving us */
+   *othercaps = gst_caps_new_simple ("audio/x-raw-float",
+       "endianness", G_TYPE_INT, G_BYTE_ORDER,
+       "signed", G_TYPE_BOOLEAN, TRUE,
+       "width", G_TYPE_INT, 32,
+       "depth", G_TYPE_INT, 32,
+-      "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, channels, NULL);
++      "rate", G_TYPE_INT, qtwrapper->samplerate, "channels", G_TYPE_INT,
++      qtwrapper->channels, NULL);
+ 
+   ret = TRUE;
+ 
+@@ -671,70 +776,76 @@ qtwrapper_audio_decoder_chain (GstPad * pad, GstBuffer * buf)
+ 
+   qtwrapper->input_buffer = buf;
+ 
+-  GST_LOG_OBJECT (qtwrapper, "Calling FillBuffer(outsamples:%d , outdata:%p)",
+-      outsamples, qtwrapper->bufferlist->mBuffers[0].mData);
+-
+-  /* Ask SCAudio to give us data ! */
+-  status = SCAudioFillBuffer (qtwrapper->adec,
+-      (SCAudioInputDataProc) process_buffer_cb,
+-      qtwrapper, (UInt32 *) & outsamples, qtwrapper->bufferlist, NULL);
+-
+-  /* TODO: What's this '42' crap?? It does seem to be needed, though. */
+-  if ((status != noErr) && (status != 42)) {
+-    if (status < 0)
+-      GST_WARNING_OBJECT (qtwrapper,
+-          "Error in SCAudioFillBuffer() : %d", (gint32) status);
+-    else
+-      GST_WARNING_OBJECT (qtwrapper,
+-          "Error in SCAudioFillBuffer() : %" GST_FOURCC_FORMAT,
+-          QT_FOURCC_ARGS (status));
+-    ret = GST_FLOW_ERROR;
+-    goto beach;
+-  }
++  do {
++    GST_LOG_OBJECT (qtwrapper,
++        "Calling SCAudioFillBuffer(outsamples:%d , outdata:%p)", outsamples,
++        qtwrapper->bufferlist->mBuffers[0].mData);
++
++    /* Ask SCAudio to give us data ! */
++    status = SCAudioFillBuffer (qtwrapper->adec,
++        (SCAudioInputDataProc) process_buffer_cb,
++        qtwrapper, (UInt32 *) & outsamples, qtwrapper->bufferlist, NULL);
++
++    /* TODO: What's this '42' crap?? It does seem to be needed, though. */
++    if ((status != noErr) && (status != 42)) {
++      if (status < 0)
++        GST_WARNING_OBJECT (qtwrapper,
++            "Error in SCAudioFillBuffer() : %d", (gint32) status);
++      else
++        GST_WARNING_OBJECT (qtwrapper,
++            "Error in SCAudioFillBuffer() : %" GST_FOURCC_FORMAT,
++            QT_FOURCC_ARGS (status));
++      ret = GST_FLOW_ERROR;
++      goto beach;
++    }
+ 
+-  realbytes = qtwrapper->bufferlist->mBuffers[0].mDataByteSize;
++    realbytes = qtwrapper->bufferlist->mBuffers[0].mDataByteSize;
+ 
+-  GST_LOG_OBJECT (qtwrapper, "We now have %d samples [%d bytes]",
+-      outsamples, realbytes);
++    GST_LOG_OBJECT (qtwrapper, "We now have %d samples [%d bytes]",
++        outsamples, realbytes);
+ 
+-  qtwrapper->bufferlist->mBuffers[0].mDataByteSize = savedbytes;
++    qtwrapper->bufferlist->mBuffers[0].mDataByteSize = savedbytes;
+ 
+-  if (!outsamples)
+-    goto beach;
++    if (!outsamples)
++      goto beach;
+ 
+-  /* 4. Create buffer and copy data in it */
+-  ret = gst_pad_alloc_buffer (qtwrapper->srcpad, qtwrapper->cur_offset,
+-      realbytes, GST_PAD_CAPS (qtwrapper->srcpad), &outbuf);
+-  if (ret != GST_FLOW_OK)
+-    goto beach;
++    /* 4. Create buffer and copy data in it */
++    ret = gst_pad_alloc_buffer (qtwrapper->srcpad, qtwrapper->cur_offset,
++        realbytes, GST_PAD_CAPS (qtwrapper->srcpad), &outbuf);
++    if (ret != GST_FLOW_OK)
++      goto beach;
+ 
+-  /* copy data from bufferlist to output buffer */
+-  g_memmove (GST_BUFFER_DATA (outbuf),
+-      qtwrapper->bufferlist->mBuffers[0].mData, realbytes);
++    /* copy data from bufferlist to output buffer */
++    g_memmove (GST_BUFFER_DATA (outbuf),
++        qtwrapper->bufferlist->mBuffers[0].mData, realbytes);
+ 
+-  /* 5. calculate timestamp and duration */
+-  GST_BUFFER_TIMESTAMP (outbuf) =
+-      qtwrapper->initial_time + gst_util_uint64_scale_int (GST_SECOND,
+-      (gint) qtwrapper->cur_offset, qtwrapper->samplerate);
+-  GST_BUFFER_SIZE (outbuf) = realbytes;
+-  GST_BUFFER_DURATION (outbuf) =
+-      gst_util_uint64_scale_int (GST_SECOND,
+-      realbytes / (qtwrapper->channels * 4), qtwrapper->samplerate);
++    /* 5. calculate timestamp and duration */
++    GST_BUFFER_TIMESTAMP (outbuf) =
++        qtwrapper->initial_time + gst_util_uint64_scale_int (GST_SECOND,
++        (gint) qtwrapper->cur_offset, qtwrapper->samplerate);
++    GST_BUFFER_SIZE (outbuf) = realbytes;
++    GST_BUFFER_DURATION (outbuf) =
++        gst_util_uint64_scale_int (GST_SECOND,
++        realbytes / (qtwrapper->channels * 4), qtwrapper->samplerate);
+ 
+-  GST_LOG_OBJECT (qtwrapper,
+-      "timestamp:%" GST_TIME_FORMAT ", duration:%" GST_TIME_FORMAT
+-      "offset:%lld, offset_end:%lld",
+-      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+-      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+-      GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
++    GST_LOG_OBJECT (qtwrapper,
++        "timestamp:%" GST_TIME_FORMAT ", duration:%" GST_TIME_FORMAT
++        "offset:%lld, offset_end:%lld",
++        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
++        GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
++        GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
+ 
+-  qtwrapper->cur_offset += outsamples;
++    qtwrapper->cur_offset += outsamples;
+ 
+-  /* 6. push buffer downstream */
++    /* 6. push buffer downstream */
+ 
+-  ret = gst_pad_push (qtwrapper->srcpad, outbuf);
+-  if (ret != GST_FLOW_OK)
+-    goto beach;
++    ret = gst_pad_push (qtwrapper->srcpad, outbuf);
++    if (ret != GST_FLOW_OK)
++      goto beach;
++
++    GST_DEBUG_OBJECT (qtwrapper,
++        "Read %d bytes, could have read up to %d bytes", realbytes, savedbytes);
++  } while (realbytes == savedbytes);
+ 
+ beach:
+   gst_buffer_unref (buf);
+@@ -875,8 +986,6 @@ qtwrapper_audio_decoders_register (GstPlugin * plugin)
+   };
+ 
+   /* Find all SoundDecompressors ! */
+-  fprintf (stderr, "There are %ld decompressors available\n",
+-      CountComponents (&desc));
+   GST_DEBUG ("There are %ld decompressors available", CountComponents (&desc));
+ 
+   /* loop over SoundDecompressors */
+diff --git sys/qtwrapper/qtutils.c sys/qtwrapper/qtutils.c
+index b698dc8..9b71bd0 100644
+--- sys/qtwrapper/qtutils.c
++++ sys/qtwrapper/qtutils.c
+@@ -42,7 +42,12 @@
+  * Boston, MA 02111-1307, USA.
+  */
+ 
++#ifdef HAVE_CONFIG_H
++#include "config.h"
++#endif
++
+ #include <string.h>
++#include <glib.h>
+ 
+ #include "qtutils.h"
+ 
+-- 
+1.5.4
+
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