Commit 26e4a482 authored by Stefan Kost's avatar Stefan Kost

gst/wavparse/gstwavparse.*: Use information from 'fact' chunk for length...

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).
parent 162b374a
2006-07-24 Stefan Kost,,, <set EMAIL_ADDRESS environment variable>
* 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).
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
Based on patch by: Joni Valtanen <joni dot valtanen at movial fi>
......
common @ ef97fb32
Subproject commit 743c74bf92546638d3f4272fd5525bf6ef71f794
Subproject commit ef97fb3278d98a1fdb32e5c6b2a7467116ffc160
......@@ -45,6 +45,11 @@
* Last reviewed on 2006-03-03 (0.10.3)
*/
/*
* TODO:
* http://replaygain.hydrogenaudio.org/file_format_wav.html
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
......@@ -228,6 +233,7 @@ gst_wavparse_reset (GstWavParse * wavparse)
wavparse->channels = 0;
wavparse->blockalign = 0;
wavparse->bps = 0;
wavparse->fact = 0;
wavparse->offset = 0;
wavparse->end_offset = 0;
wavparse->dataleft = 0;
......@@ -693,7 +699,8 @@ gst_wavparse_other (GstWavParse * wav)
}
}
wav->datasize = (guint64) length;
break;
GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
break;
case GST_RIFF_TAG_cue:
if (!gst_riff_read_skip (wav)) {
......@@ -775,7 +782,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
gint64 cur, stop, upstream_size;
gboolean flush;
gboolean update;
GstSegment seeksegment;
GstSegment seeksegment = { 0, };
if (event) {
GST_DEBUG_OBJECT (wav, "doing seek with event");
......@@ -825,23 +832,36 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
cur_type, cur, stop_type, stop, &update);
}
if ((stop = seeksegment.stop) == -1)
if ((stop = seeksegment.stop) == GST_CLOCK_TIME_NONE)
stop = seeksegment.duration;
if (cur_type != GST_SEEK_TYPE_NONE) {
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE) &&
(seeksegment.last_stop != GST_CLOCK_TIME_NONE)) {
wav->offset =
gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);
wav->offset -= wav->offset % wav->bytes_per_sample;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset -= (wav->offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset += wav->datastart;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
} else {
GST_DEBUG_OBJECT (wav, "last_stop == -1");
wav->offset = wav->datastart;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
}
if (stop != -1) {
if (stop != GST_CLOCK_TIME_NONE) {
wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
wav->end_offset +=
wav->bytes_per_sample - (wav->end_offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset += wav->datastart;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
} else {
GST_DEBUG_OBJECT (wav, "stop == -1");
wav->end_offset = wav->datasize + wav->datastart;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
}
/* make sure filesize is not exceeded due to rounding errors or so,
......@@ -979,13 +999,20 @@ gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
}
}
/* FIXME: remove once -base 0.10.9 is out */
#ifndef GST_RIFF_TAG_bext
#define GST_RIFF_TAG_bext GST_MAKE_FOURCC ('b', 'e', 'x', 't')
#endif
#ifndef GST_RIFF_TAG_BEXT
#define GST_RIFF_TAG_BEXT GST_MAKE_FOURCC ('B', 'E', 'X', 'T')
#endif
static gboolean
gst_wavparse_get_upstream_size (GstWavParse * wav, gint64 * len)
{
gboolean res = FALSE;
GstFormat fmt = GST_FORMAT_BYTES;
GstPad *peer;
if ((peer = gst_pad_get_peer (wav->sinkpad))) {
res = gst_pad_query_duration (peer, &fmt, len);
gst_object_unref (peer);
}
return res;
}
static GstFlowReturn
gst_wavparse_stream_headers (GstWavParse * wav)
......@@ -1048,8 +1075,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
/* Note: gst_riff_create_audio_caps might need to fix values in
* the header header depending on the format, so call it first */
caps =
gst_riff_create_audio_caps (header->format, NULL, header, extra,
caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
NULL, &codec_name);
if (extra)
......@@ -1058,24 +1084,29 @@ gst_wavparse_stream_headers (GstWavParse * wav)
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
if (wav->channels == 0)
goto no_channels;
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
if (wav->bps <= 0)
goto no_bitrate;
g_free (header);
if (wav->channels == 0)
goto no_channels;
if (wav->bps == 0 && (wav->format == GST_RIFF_WAVE_FORMAT_MPEGL12 ||
wav->format == GST_RIFF_WAVE_FORMAT_MPEGL3)) {
/* Note: ugly workaround for mp2/mp3 embedded in wav, that relies on the
* bitrate inside the mpeg stream */
/* wav->bps = 1; */
GST_INFO ("WAV file with bps==0 and format=mp2/3");
}
wav->width = (wav->blockalign * 8) / wav->channels;
wav->bytes_per_sample = wav->channels * wav->width / 8;
if (wav->bytes_per_sample <= 0)
goto no_bytes_per_sample;
g_free (header);
if (!caps)
goto unknown_format;
......@@ -1083,6 +1114,11 @@ gst_wavparse_stream_headers (GstWavParse * wav)
GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate);
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
/* create pad later so we can sniff the first few bytes
* of the real data and correct our caps if necessary */
......@@ -1101,12 +1137,12 @@ gst_wavparse_stream_headers (GstWavParse * wav)
codec_name = NULL;
}
GST_DEBUG_OBJECT (wav, "frequency %d, channels %d", wav->rate,
wav->channels);
}
/* loop headers until we get data */
while (!gotdata) {
gint64 upstream_size = 0;
if (wav->streaming) {
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
return GST_FLOW_OK;
......@@ -1119,21 +1155,21 @@ gst_wavparse_stream_headers (GstWavParse * wav)
size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
}
gst_wavparse_get_upstream_size (wav, &upstream_size);
/*
wav is a st00pid format, we don't know for sure where data starts.
So we have to go bit by bit until we find the 'data' header
*/
switch (tag) {
/* TODO : Implement the various cases */
case GST_RIFF_TAG_data:{
GstFormat fmt;
gint64 upstream_size;
GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
gotdata = TRUE;
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8);
gotdata = TRUE;
} else {
gst_buffer_unref (buf);
}
......@@ -1141,12 +1177,41 @@ gst_wavparse_stream_headers (GstWavParse * wav)
wav->datastart = wav->offset;
/* file might be truncated */
fmt = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (wav->sinkpad, &fmt, &upstream_size)) {
if (upstream_size) {
size = MIN (size, (upstream_size - wav->datastart));
}
wav->datasize = size;
wav->dataleft = size;
wav->datasize = (guint64) size;
wav->dataleft = (guint64) size;
wav->end_offset = size + wav->datastart;
if (!wav->streaming) {
/* We will continue parsing tags 'till end */
wav->offset += size;
}
GST_DEBUG_OBJECT (wav, "datasize = %ld", size);
break;
}
case GST_RIFF_TAG_fact:{
/* number of samples (for compressed formats) */
if (wav->streaming) {
const guint8 *data = NULL;
if (gst_adapter_available (wav->adapter) < 8 + 4) {
return GST_FLOW_OK;
}
gst_adapter_flush (wav->adapter, 8);
data = gst_adapter_peek (wav->adapter, 4);
wav->fact = GST_READ_UINT32_LE (data);
gst_adapter_flush (wav->adapter, 4);
} else {
gst_buffer_unref (buf);
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8, 4,
&buf)) != GST_FLOW_OK)
goto header_read_error;
wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
gst_buffer_unref (buf);
}
wav->offset += 8 + 4;
break;
}
default:
......@@ -1163,10 +1228,24 @@ gst_wavparse_stream_headers (GstWavParse * wav)
gst_buffer_unref (buf);
}
}
if (upstream_size && (wav->offset >= upstream_size)) {
/* Now we are gone through the whole file */
gotdata = TRUE;
}
}
GST_DEBUG_OBJECT (wav, "Finished parsing headers");
if (wav->bps <= 0 && wav->fact) {
wav->bps =
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
(guint64) wav->fact);
GST_DEBUG_OBJECT (wav, "calculated bps : %ld", wav->bps);
}
if (wav->bps <= 0)
goto no_bitrate;
duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
......@@ -1317,7 +1396,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
const guint8 dts_marker[] = { 0xFF, 0x1F, 0x00, 0xE8, 0xF1, 0x07 };
s = gst_caps_get_structure (wav->caps, 0);
if (gst_structure_has_name (s, "audio/x-raw-int") &&
if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf &&
GST_BUFFER_SIZE (buf) > 6 &&
memcmp (GST_BUFFER_DATA (buf), dts_marker, 6) == 0) {
......@@ -1465,6 +1544,10 @@ found_eos:
gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
stop));
} else {
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
gst_wavparse_add_src_pad (wav, NULL);
}
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
return GST_FLOW_WRONG_STATE;
......@@ -1475,13 +1558,15 @@ pull_error:
if (res == GST_FLOW_UNEXPECTED)
goto found_eos;
GST_DEBUG_OBJECT (wav, "Error getting %" G_GINT64_FORMAT " bytes from the "
GST_WARNING_OBJECT (wav,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
return res;
}
push_error:
{
GST_DEBUG_OBJECT (wav, "Error pushing on srcpad");
GST_WARNING_OBJECT (wav, "Error pushing on srcpad %p, is linked? = %d",
wav->srcpad, gst_pad_is_linked (wav->srcpad));
return res;
}
}
......@@ -1761,10 +1846,15 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
res &=
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
&format, &end);
case GST_FORMAT_TIME:{
if (wav->fact) {
end = GST_SECOND * wav->fact / wav->rate;
} else {
res &=
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
&format, &end);
}
}
break;
default:
format = GST_FORMAT_BYTES;
......
......@@ -81,6 +81,7 @@ struct _GstWavParse {
guint16 blockalign;
guint16 width;
guint32 bps;
guint32 fact;
guint bytes_per_sample;
......
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