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gst/wavparse/gstwavparse.*: Use information from 'fact' chunk for length calculation of compressed samples. Calculate...
Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_other), (gst_wavparse_perform_seek), (gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_pad_query): * gst/wavparse/gstwavparse.h: Use information from 'fact' chunk for length calculation of compressed samples. Calculate bps if bogus value is found in wav header (embeded mp2/mp3).
This commit is contained in:
Stefan Kost
parent
162b374ae2
commit
26e4a48271
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2006-07-24 Stefan Kost,,, <set EMAIL_ADDRESS environment variable>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
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(gst_wavparse_other), (gst_wavparse_perform_seek),
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(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
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(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
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(gst_wavparse_pad_query):
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* gst/wavparse/gstwavparse.h:
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Use information from 'fact' chunk for length calculation of compressed
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samples. Calculate bps if bogus value is found in wav header (embeded
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mp2/mp3).
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2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
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Based on patch by: Joni Valtanen <joni dot valtanen at movial fi>
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2
common
2
common
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@ -1 +1 @@
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Subproject commit 743c74bf92546638d3f4272fd5525bf6ef71f794
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Subproject commit ef97fb3278d98a1fdb32e5c6b2a7467116ffc160
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@ -45,6 +45,11 @@
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* Last reviewed on 2006-03-03 (0.10.3)
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*/
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/*
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* TODO:
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* http://replaygain.hydrogenaudio.org/file_format_wav.html
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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@ -228,6 +233,7 @@ gst_wavparse_reset (GstWavParse * wavparse)
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wavparse->channels = 0;
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wavparse->blockalign = 0;
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wavparse->bps = 0;
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wavparse->fact = 0;
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wavparse->offset = 0;
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wavparse->end_offset = 0;
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wavparse->dataleft = 0;
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@ -693,7 +699,8 @@ gst_wavparse_other (GstWavParse * wav)
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}
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}
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wav->datasize = (guint64) length;
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break;
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GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
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break;
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case GST_RIFF_TAG_cue:
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if (!gst_riff_read_skip (wav)) {
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@ -775,7 +782,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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gint64 cur, stop, upstream_size;
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gboolean flush;
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gboolean update;
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GstSegment seeksegment;
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GstSegment seeksegment = { 0, };
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if (event) {
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GST_DEBUG_OBJECT (wav, "doing seek with event");
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@ -825,23 +832,36 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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cur_type, cur, stop_type, stop, &update);
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}
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if ((stop = seeksegment.stop) == -1)
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if ((stop = seeksegment.stop) == GST_CLOCK_TIME_NONE)
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stop = seeksegment.duration;
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if (cur_type != GST_SEEK_TYPE_NONE) {
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GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
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if ((cur_type != GST_SEEK_TYPE_NONE) &&
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(seeksegment.last_stop != GST_CLOCK_TIME_NONE)) {
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wav->offset =
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gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);
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wav->offset -= wav->offset % wav->bytes_per_sample;
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GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
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wav->offset -= (wav->offset % wav->bytes_per_sample);
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GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
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wav->offset += wav->datastart;
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GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
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} else {
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GST_DEBUG_OBJECT (wav, "last_stop == -1");
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wav->offset = wav->datastart;
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GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
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}
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if (stop != -1) {
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if (stop != GST_CLOCK_TIME_NONE) {
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wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
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wav->end_offset +=
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wav->bytes_per_sample - (wav->end_offset % wav->bytes_per_sample);
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GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
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wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
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GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
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wav->end_offset += wav->datastart;
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GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
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} else {
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GST_DEBUG_OBJECT (wav, "stop == -1");
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wav->end_offset = wav->datasize + wav->datastart;
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GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
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}
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/* make sure filesize is not exceeded due to rounding errors or so,
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@ -979,13 +999,20 @@ gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
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}
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}
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/* FIXME: remove once -base 0.10.9 is out */
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#ifndef GST_RIFF_TAG_bext
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#define GST_RIFF_TAG_bext GST_MAKE_FOURCC ('b', 'e', 'x', 't')
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#endif
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#ifndef GST_RIFF_TAG_BEXT
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#define GST_RIFF_TAG_BEXT GST_MAKE_FOURCC ('B', 'E', 'X', 'T')
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#endif
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static gboolean
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gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)
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{
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gboolean res = FALSE;
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GstFormat fmt = GST_FORMAT_BYTES;
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GstPad *peer;
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if ((peer = gst_pad_get_peer (wav->sinkpad))) {
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res = gst_pad_query_duration (peer, &fmt, len);
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gst_object_unref (peer);
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}
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return res;
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}
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static GstFlowReturn
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gst_wavparse_stream_headers (GstWavParse * wav)
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@ -1048,8 +1075,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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/* Note: gst_riff_create_audio_caps might need to fix values in
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* the header header depending on the format, so call it first */
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caps =
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gst_riff_create_audio_caps (header->format, NULL, header, extra,
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caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
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NULL, &codec_name);
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if (extra)
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@ -1058,24 +1084,29 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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wav->format = header->format;
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wav->rate = header->rate;
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wav->channels = header->channels;
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wav->blockalign = header->blockalign;
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wav->depth = header->size;
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wav->bps = header->av_bps;
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g_free (header);
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if (wav->channels == 0)
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goto no_channels;
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wav->blockalign = header->blockalign;
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wav->width = (header->blockalign * 8) / header->channels;
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wav->depth = header->size;
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wav->bps = header->av_bps;
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if (wav->bps <= 0)
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goto no_bitrate;
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if (wav->bps == 0 && (wav->format == GST_RIFF_WAVE_FORMAT_MPEGL12 ||
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wav->format == GST_RIFF_WAVE_FORMAT_MPEGL3)) {
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/* Note: ugly workaround for mp2/mp3 embedded in wav, that relies on the
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* bitrate inside the mpeg stream */
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/* wav->bps = 1; */
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GST_INFO ("WAV file with bps==0 and format=mp2/3");
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}
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wav->width = (wav->blockalign * 8) / wav->channels;
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wav->bytes_per_sample = wav->channels * wav->width / 8;
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if (wav->bytes_per_sample <= 0)
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goto no_bytes_per_sample;
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g_free (header);
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if (!caps)
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goto unknown_format;
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@ -1083,6 +1114,11 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
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GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
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GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
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GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate);
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GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
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GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
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GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
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/* create pad later so we can sniff the first few bytes
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* of the real data and correct our caps if necessary */
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@ -1101,12 +1137,12 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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codec_name = NULL;
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}
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GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate,
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wav->channels);
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}
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/* loop headers until we get data */
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while (!gotdata) {
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gint64 upstream_size = 0;
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if (wav->streaming) {
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if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
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return GST_FLOW_OK;
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@ -1119,21 +1155,21 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
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}
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gst_wavparse_get_upstream_size (wav, &upstream_size);
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/*
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wav is a st00pid format, we don't know for sure where data starts.
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So we have to go bit by bit until we find the 'data' header
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*/
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switch (tag) {
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/* TODO : Implement the various cases */
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case GST_RIFF_TAG_data:{
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GstFormat fmt;
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gint64 upstream_size;
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GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
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gotdata = TRUE;
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if (wav->streaming) {
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gst_adapter_flush (wav->adapter, 8);
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gotdata = TRUE;
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} else {
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gst_buffer_unref (buf);
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}
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@ -1141,12 +1177,41 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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wav->datastart = wav->offset;
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/* file might be truncated */
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fmt = GST_FORMAT_BYTES;
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if (gst_pad_query_peer_duration (wav->sinkpad, &fmt, &upstream_size)) {
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if (upstream_size) {
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size = MIN (size, (upstream_size - wav->datastart));
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}
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wav->datasize = size;
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wav->dataleft = size;
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wav->datasize = (guint64) size;
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wav->dataleft = (guint64) size;
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wav->end_offset = size + wav->datastart;
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if (!wav->streaming) {
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/* We will continue parsing tags 'till end */
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wav->offset += size;
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}
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GST_DEBUG_OBJECT (wav, "datasize = %ld", size);
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break;
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}
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case GST_RIFF_TAG_fact:{
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/* number of samples (for compressed formats) */
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if (wav->streaming) {
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const guint8 *data = NULL;
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if (gst_adapter_available (wav->adapter) < 8 + 4) {
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return GST_FLOW_OK;
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}
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gst_adapter_flush (wav->adapter, 8);
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data = gst_adapter_peek (wav->adapter, 4);
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wav->fact = GST_READ_UINT32_LE (data);
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gst_adapter_flush (wav->adapter, 4);
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} else {
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gst_buffer_unref (buf);
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if ((res =
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gst_pad_pull_range (wav->sinkpad, wav->offset + 8, 4,
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&buf)) != GST_FLOW_OK)
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goto header_read_error;
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wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
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gst_buffer_unref (buf);
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}
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wav->offset += 8 + 4;
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break;
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}
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default:
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@ -1163,10 +1228,24 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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gst_buffer_unref (buf);
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}
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}
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if (upstream_size && (wav->offset >= upstream_size)) {
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/* Now we are gone through the whole file */
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gotdata = TRUE;
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}
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}
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GST_DEBUG_OBJECT (wav, "Finished parsing headers");
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if (wav->bps <= 0 && wav->fact) {
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wav->bps =
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(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
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(guint64) wav->fact);
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GST_DEBUG_OBJECT (wav, "calculated bps : %ld", wav->bps);
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}
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if (wav->bps <= 0)
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goto no_bitrate;
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duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
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GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
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GST_TIME_ARGS (duration));
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@ -1317,7 +1396,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
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const guint8 dts_marker[] = { 0xFF, 0x1F, 0x00, 0xE8, 0xF1, 0x07 };
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s = gst_caps_get_structure (wav->caps, 0);
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if (gst_structure_has_name (s, "audio/x-raw-int") &&
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if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf &&
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GST_BUFFER_SIZE (buf) > 6 &&
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memcmp (GST_BUFFER_DATA (buf), dts_marker, 6) == 0) {
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@ -1465,6 +1544,10 @@ found_eos:
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gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
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stop));
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} else {
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if (G_UNLIKELY (wav->first)) {
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wav->first = FALSE;
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gst_wavparse_add_src_pad (wav, NULL);
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}
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gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
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}
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return GST_FLOW_WRONG_STATE;
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@ -1475,13 +1558,15 @@ pull_error:
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if (res == GST_FLOW_UNEXPECTED)
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goto found_eos;
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GST_DEBUG_OBJECT (wav, "Error getting %" G_GINT64_FORMAT " bytes from the "
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GST_WARNING_OBJECT (wav,
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"Error getting %" G_GINT64_FORMAT " bytes from the "
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"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
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return res;
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}
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push_error:
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{
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GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
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GST_WARNING_OBJECT (wav, "Error pushing on srcpad %p, is linked? = %d",
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wav->srcpad, gst_pad_is_linked (wav->srcpad));
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return res;
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}
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}
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@ -1761,10 +1846,15 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
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gst_query_parse_duration (query, &format, NULL);
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switch (format) {
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case GST_FORMAT_TIME:
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res &=
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gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
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&format, &end);
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case GST_FORMAT_TIME:{
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if (wav->fact) {
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end = GST_SECOND * wav->fact / wav->rate;
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} else {
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res &=
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gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
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&format, &end);
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}
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}
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break;
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default:
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format = GST_FORMAT_BYTES;
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@ -81,6 +81,7 @@ struct _GstWavParse {
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guint16 blockalign;
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guint16 width;
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guint32 bps;
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guint32 fact;
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guint bytes_per_sample;
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