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flvdemux: use aac codec-data to adjust samplerate if needed

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Based on patch by Fabien Lebaillif-Delamare <fabien@arq-media.com>

Fixes #636621.
This commit is contained in:
Mark Nauwelaerts 2010-12-07 13:11:48 +01:00
parent 8ca094795a
commit 6f8ce30c20
2 changed files with 29 additions and 7 deletions
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6
gst/flv/Makefile.am
View file

@ -1,9 +1,9 @@
plugin_LTLIBRARIES = libgstflv.la
libgstflv_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstflv_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) \
-lgstpbutils-@GST_MAJORMINOR@
libgstflv_la_LDFLAGS = ${GST_PLUGIN_LDFLAGS}
libgstflv_la_LIBADD = -lgstpbutils-@GST_MAJORMINOR@ \
$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS)
libgstflv_la_LDFLAGS = ${GST_PLUGIN_LDFLAGS}
libgstflv_la_SOURCES = gstflvdemux.c gstflvmux.c
libgstflv_la_LIBTOOLFLAGS = --tag=disable-static

30
gst/flv/gstflvdemux.c
View file

@ -40,6 +40,7 @@
#include <string.h>
#include <gst/base/gstbytereader.h>
#include <gst/pbutils/descriptions.h>
#include <gst/pbutils/pbutils.h>
static GstStaticPadTemplate flv_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
@ -575,6 +576,7 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
GstCaps *caps = NULL;
gchar *codec_name = NULL;
gboolean ret = FALSE;
guint adjusted_rate = rate;
switch (codec_tag) {
case 1:
@ -603,9 +605,29 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
caps = gst_caps_new_simple ("audio/x-nellymoser", NULL);
break;
case 10:
{
/* use codec-data to extract and verify samplerate */
if (demux->audio_codec_data &&
GST_BUFFER_SIZE (demux->audio_codec_data) >= 2) {
gint freq_index;
freq_index =
((GST_READ_UINT16_BE (GST_BUFFER_DATA (demux->audio_codec_data))));
freq_index = (freq_index & 0x0780) >> 7;
adjusted_rate =
gst_codec_utils_aac_get_sample_rate_from_index (freq_index);
if (adjusted_rate && (rate != adjusted_rate)) {
GST_LOG_OBJECT (demux, "Ajusting AAC sample rate %d -> %d", rate,
adjusted_rate);
} else {
adjusted_rate = rate;
}
}
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
break;
}
case 7:
caps = gst_caps_new_simple ("audio/x-alaw", NULL);
break;
@ -624,8 +646,8 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
goto beach;
}
gst_caps_set_simple (caps,
"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, channels, NULL);
gst_caps_set_simple (caps, "rate", G_TYPE_INT, adjusted_rate,
"channels", G_TYPE_INT, channels, NULL);
if (demux->audio_codec_data) {
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER,
@ -635,7 +657,7 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag,
ret = gst_pad_set_caps (demux->audio_pad, caps);
if (G_LIKELY (ret)) {
/* Store the caps we have set */
/* Store the caps we got from tags */
demux->audio_codec_tag = codec_tag;
demux->rate = rate;
demux->channels = channels;
@ -851,7 +873,7 @@ gst_flv_demux_parse_tag_audio (GstFlvDemux * demux, GstBuffer * buffer)
switch (aac_packet_type) {
case 0:
{
/* AudioSpecificConfic data */
/* AudioSpecificConfig data */
GST_LOG_OBJECT (demux, "got an AAC codec data packet");
if (demux->audio_codec_data) {
gst_buffer_unref (demux->audio_codec_data);