Commit 96da5200 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵
Browse files

gst/matroska/: Fix demuxing of WavPack files. Muxing is still broken.

Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_add_wvpk_header),
(gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Fix demuxing of WavPack files. Muxing is still broken.
parent 7afcb806
2008-06-19 Sebastian Dröge <slomo@circular-chaos.org>
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_add_wvpk_header),
(gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Fix demuxing of WavPack files. Muxing is still broken.
2008-06-19 Sebastian Dröge <slomo@circular-chaos.org>
 
* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
......
......@@ -3315,61 +3315,148 @@ gst_matroska_demux_add_wvpk_header (GstElement * element,
{
GstMatroskaDemux *demux = GST_MATROSKA_DEMUX (element);
GstBuffer *newbuf;
GstMatroskaTrackAudioContext *audiocontext =
(GstMatroskaTrackAudioContext *) stream;
GstBuffer *newbuf = NULL;
guint8 *data;
guint newlen;
GstFlowReturn ret, cret;
GstFlowReturn ret, cret = GST_FLOW_OK;
/* we need to reconstruct the header of the wavpack block */
Wavpack4Header wvh;
/* FIXME: broken for > 2 channels and hybrid files
http://www.matroska.org/technical/specs/codecid/wavpack.html */
wvh.ck_id[0] = 'w';
wvh.ck_id[1] = 'v';
wvh.ck_id[2] = 'p';
wvh.ck_id[3] = 'k';
/* -20 because ck_size is the size of the wavpack block -8
* and lace_size is the size of the wavpack block + 12
* (the three guint32 of the header that already are in the buffer) */
wvh.ck_size = GST_BUFFER_SIZE (*buf) + sizeof (Wavpack4Header) - 20;
wvh.version = GST_READ_UINT16_LE (stream->codec_priv);
wvh.track_no = 0;
wvh.index_no = 0;
wvh.total_samples = -1;
wvh.block_index = 0;
wvh.block_index = audiocontext->wvpk_block_index;
/* block_samples, flags and crc are already in the buffer */
newlen = GST_BUFFER_SIZE (*buf) + sizeof (Wavpack4Header) - 12;
ret = gst_pad_alloc_buffer_and_set_caps (stream->pad, GST_BUFFER_OFFSET_NONE,
newlen, stream->caps, &newbuf);
cret = gst_matroska_demux_combine_flows (demux, stream, ret);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (demux, "pad_alloc failed %s, combined %s",
gst_flow_get_name (ret), gst_flow_get_name (cret));
return cret;
}
if (audiocontext->channels <= 2) {
guint32 block_samples;
data = GST_BUFFER_DATA (newbuf);
data[0] = 'w';
data[1] = 'v';
data[2] = 'p';
data[3] = 'k';
GST_WRITE_UINT32_LE (data + 4, wvh.ck_size);
GST_WRITE_UINT16_LE (data + 8, wvh.version);
GST_WRITE_UINT8 (data + 10, wvh.track_no);
GST_WRITE_UINT8 (data + 11, wvh.index_no);
GST_WRITE_UINT32_LE (data + 12, wvh.total_samples);
GST_WRITE_UINT32_LE (data + 16, wvh.block_index);
g_memmove (data + 20, GST_BUFFER_DATA (*buf), GST_BUFFER_SIZE (*buf));
gst_buffer_copy_metadata (newbuf, *buf,
GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS);
gst_buffer_unref (*buf);
*buf = newbuf;
block_samples = GST_READ_UINT32_LE (GST_BUFFER_DATA (*buf));
/* we need to reconstruct the header of the wavpack block */
/* -20 because ck_size is the size of the wavpack block -8
* and lace_size is the size of the wavpack block + 12
* (the three guint32 of the header that already are in the buffer) */
wvh.ck_size = GST_BUFFER_SIZE (*buf) + sizeof (Wavpack4Header) - 20;
/* block_samples, flags and crc are already in the buffer */
newlen = GST_BUFFER_SIZE (*buf) + sizeof (Wavpack4Header) - 12;
ret =
gst_pad_alloc_buffer_and_set_caps (stream->pad, GST_BUFFER_OFFSET_NONE,
newlen, stream->caps, &newbuf);
cret = gst_matroska_demux_combine_flows (demux, stream, ret);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (demux, "pad_alloc failed %s, combined %s",
gst_flow_get_name (ret), gst_flow_get_name (cret));
return cret;
}
data = GST_BUFFER_DATA (newbuf);
data[0] = 'w';
data[1] = 'v';
data[2] = 'p';
data[3] = 'k';
GST_WRITE_UINT32_LE (data + 4, wvh.ck_size);
GST_WRITE_UINT16_LE (data + 8, wvh.version);
GST_WRITE_UINT8 (data + 10, wvh.track_no);
GST_WRITE_UINT8 (data + 11, wvh.index_no);
GST_WRITE_UINT32_LE (data + 12, wvh.total_samples);
GST_WRITE_UINT32_LE (data + 16, wvh.block_index);
g_memmove (data + 20, GST_BUFFER_DATA (*buf), GST_BUFFER_SIZE (*buf));
gst_buffer_copy_metadata (newbuf, *buf,
GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS);
gst_buffer_unref (*buf);
*buf = newbuf;
audiocontext->wvpk_block_index += block_samples;
} else {
guint8 *outdata;
guint outpos = 0;
guint size;
guint32 block_samples, flags, crc, blocksize;
data = GST_BUFFER_DATA (*buf);
size = GST_BUFFER_SIZE (*buf);
if (size < 4) {
GST_ERROR_OBJECT (demux, "Too small wavpack buffer");
return GST_FLOW_ERROR;
}
block_samples = GST_READ_UINT32_LE (data);
data += 4;
size -= 4;
while (size > 12) {
flags = GST_READ_UINT32_LE (data);
data += 4;
size -= 4;
crc = GST_READ_UINT32_LE (data);
data += 4;
size -= 4;
blocksize = GST_READ_UINT32_LE (data);
data += 4;
size -= 4;
if (blocksize == 0 || size < blocksize)
break;
if (newbuf == NULL) {
newbuf = gst_buffer_new_and_alloc (sizeof (Wavpack4Header) + blocksize);
gst_buffer_set_caps (newbuf, stream->caps);
gst_buffer_copy_metadata (newbuf, *buf,
GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS);
outpos = 0;
outdata = GST_BUFFER_DATA (newbuf);
} else {
GST_BUFFER_SIZE (newbuf) += sizeof (Wavpack4Header) + blocksize;
GST_BUFFER_DATA (newbuf) =
g_realloc (GST_BUFFER_DATA (newbuf), GST_BUFFER_SIZE (newbuf));
GST_BUFFER_MALLOCDATA (newbuf) = GST_BUFFER_DATA (newbuf);
outdata = GST_BUFFER_DATA (newbuf);
}
outdata[outpos] = 'w';
outdata[outpos + 1] = 'v';
outdata[outpos + 2] = 'p';
outdata[outpos + 3] = 'k';
outpos += 4;
GST_WRITE_UINT32_LE (outdata + outpos,
blocksize + sizeof (Wavpack4Header) - 8);
GST_WRITE_UINT16_LE (outdata + outpos + 4, wvh.version);
GST_WRITE_UINT8 (outdata + outpos + 6, wvh.track_no);
GST_WRITE_UINT8 (outdata + outpos + 7, wvh.index_no);
GST_WRITE_UINT32_LE (outdata + outpos + 8, wvh.total_samples);
GST_WRITE_UINT32_LE (outdata + outpos + 12, wvh.block_index);
GST_WRITE_UINT32_LE (outdata + outpos + 16, block_samples);
GST_WRITE_UINT32_LE (outdata + outpos + 20, flags);
GST_WRITE_UINT32_LE (outdata + outpos + 24, crc);
outpos += 28;
g_memmove (outdata + outpos, data, blocksize);
outpos += blocksize;
data += blocksize;
size -= blocksize;
}
gst_buffer_unref (*buf);
*buf = newbuf;
audiocontext->wvpk_block_index += block_samples;
}
return cret;
}
......@@ -5066,6 +5153,7 @@ gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext *
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
*codec_name = g_strdup ("Wavpack audio");
context->postprocess_frame = gst_matroska_demux_add_wvpk_header;
audiocontext->wvpk_block_index = 0;
} else if ((!strcmp (codec_id, GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4)) ||
(!strcmp (codec_id, GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4)) ||
(!strcmp (codec_id, GST_MATROSKA_CODEC_ID_AUDIO_REAL_COOK))) {
......
......@@ -523,6 +523,8 @@ typedef struct _GstMatroskaTrackAudioContext {
GstMatroskaTrackContext parent;
guint samplerate, channels, bitdepth;
guint32 wvpk_block_index;
} GstMatroskaTrackAudioContext;
typedef struct _GstMatroskaTrackSubtitleContext {
......
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