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)
{
/* 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
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
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, &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"))) {
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));
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;
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;
}
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);
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;
+ }
}
}
}
}
+ 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);
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;
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;
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);
};
/* 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 */
projects / platform / upstream / gstreamer.git / commitdiff