lamemp3enc: ensure parsed output

... by doing some basic parsing of encoded lame data.

Fixes #652150.
This commit is contained in:
Mark Nauwelaerts 2011-12-26 18:23:52 +01:00
parent ee31252201
commit e21ba604a4
3 changed files with 237 additions and 6 deletions

View file

@ -2,8 +2,8 @@ plugin_LTLIBRARIES = libgstlame.la
libgstlame_la_SOURCES = gstlame.c gstlamemp3enc.c plugin.c
libgstlame_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
libgstlame_la_LIBADD = $(LAME_LIBS) $(GST_PLUGINS_BASE_LIBS) \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
libgstlame_la_LIBADD = $(LAME_LIBS) $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
-lgstaudio-$(GST_MAJORMINOR) $(GST_LIBS)
libgstlame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstlame_la_LIBTOOLFLAGS = --tag=disable-static

View file

@ -289,6 +289,11 @@ gst_lamemp3enc_start (GstAudioEncoder * enc)
GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
GST_DEBUG_OBJECT (lame, "start");
if (!lame->adapter)
lame->adapter = gst_adapter_new ();
gst_adapter_clear (lame->adapter);
return TRUE;
}
@ -299,6 +304,11 @@ gst_lamemp3enc_stop (GstAudioEncoder * enc)
GST_DEBUG_OBJECT (lame, "stop");
if (lame->adapter) {
g_object_unref (lame->adapter);
lame->adapter = NULL;
}
gst_lamemp3enc_release_memory (lame);
return TRUE;
}
@ -334,6 +344,7 @@ gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
"output samplerate %d is different from incoming samplerate %d",
out_samplerate, lame->samplerate);
}
lame->out_samplerate = out_samplerate;
version = lame_get_version (lame->lgf);
if (version == 0)
@ -488,12 +499,209 @@ gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
}
}
/* **** credits go to mpegaudioparse **** */
static const guint mp3types_bitrates[2][3][16] = {
{
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
},
{
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
},
};
static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}
};
static inline guint
mp3_type_frame_length_from_header (GstLameMP3Enc * lame, guint32 header,
guint * put_version, guint * put_layer, guint * put_channels,
guint * put_bitrate, guint * put_samplerate, guint * put_mode,
guint * put_crc)
{
guint length;
gulong mode, samplerate, bitrate, layer, channels, padding, crc;
gulong version;
gint lsf, mpg25;
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
version = 1 + lsf + mpg25;
layer = 4 - ((header >> 17) & 0x3);
crc = (header >> 16) & 0x1;
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
/* The caller has ensured we have a valid header, so bitrate can't be
zero here. */
g_assert (bitrate != 0);
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
switch (layer) {
case 1:
length = 4 * ((bitrate * 12) / samplerate + padding);
break;
case 2:
length = (bitrate * 144) / samplerate + padding;
break;
default:
case 3:
length = (bitrate * 144) / (samplerate << lsf) + padding;
break;
}
GST_DEBUG_OBJECT (lame, "Calculated mp3 frame length of %u bytes", length);
GST_DEBUG_OBJECT (lame, "samplerate = %lu, bitrate = %lu, version = %lu, "
"layer = %lu, channels = %lu", samplerate, bitrate, version,
layer, channels);
if (put_version)
*put_version = version;
if (put_layer)
*put_layer = layer;
if (put_channels)
*put_channels = channels;
if (put_bitrate)
*put_bitrate = bitrate;
if (put_samplerate)
*put_samplerate = samplerate;
if (put_mode)
*put_mode = mode;
if (put_crc)
*put_crc = crc;
return length;
}
static gboolean
mp3_sync_check (GstLameMP3Enc * lame, unsigned long head)
{
GST_DEBUG_OBJECT (lame, "checking mp3 header 0x%08lx", head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
GST_WARNING_OBJECT (lame, "invalid sync");
return FALSE;
}
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
GST_WARNING_OBJECT (lame, "invalid MPEG version: 0x%lx", (head >> 19) & 3);
return FALSE;
}
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
GST_WARNING_OBJECT (lame, "invalid layer: 0x%lx", (head >> 17) & 3);
return FALSE;
}
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx."
"Free format files are not supported yet", (head >> 12) & 0xf);
return FALSE;
}
if (((head >> 12) & 0xf) == 0xf) {
GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
return FALSE;
}
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
GST_WARNING_OBJECT (lame, "invalid samplerate: 0x%lx", (head >> 10) & 0x3);
return FALSE;
}
if ((head & 0x3) == 0x2) {
/* Ignore this as there are some files with emphasis 0x2 that can
* be played fine. See BGO #537235 */
GST_WARNING_OBJECT (lame, "invalid emphasis: 0x%lx", head & 0x3);
}
return TRUE;
}
/* **** end mpegaudioparse **** */
static GstFlowReturn
gst_lamemp3enc_finish_frames (GstLameMP3Enc * lame)
{
gint av;
guint header;
GstFlowReturn result = GST_FLOW_OK;
/* limited parsing, we don't expect to lose sync here */
while ((result == GST_FLOW_OK) &&
((av = gst_adapter_available (lame->adapter)) > 4)) {
guint rate, version, layer, size;
GstBuffer *mp3_buf;
const guint8 *data;
data = gst_adapter_peek (lame->adapter, 4);
header = GST_READ_UINT32_BE (data);
if (!mp3_sync_check (lame, header))
goto invalid_header;
size = mp3_type_frame_length_from_header (lame, header, &version, &layer,
NULL, NULL, &rate, NULL, NULL);
if (G_UNLIKELY (layer != 3 || rate != lame->out_samplerate)) {
GST_DEBUG_OBJECT (lame,
"unexpected mp3 header with (rate, layer): (%u, %u)",
rate, version, layer);
goto invalid_header;
}
if (size > av) {
/* pretty likely to occur when lame is holding back on us */
GST_LOG_OBJECT (lame, "frame size %u (> %d)", size, av);
break;
}
/* should be ok now */
mp3_buf = gst_adapter_take_buffer (lame->adapter, size);
/* number of samples for MPEG-1, layer 3 */
result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame),
mp3_buf, version == 1 ? 1152 : 576);
}
exit:
return result;
/* ERRORS */
invalid_header:
{
GST_ELEMENT_ERROR (lame, STREAM, ENCODE,
("invalid lame mp3 sync header %08X", header), (NULL));
result = GST_FLOW_ERROR;
goto exit;
}
}
static GstFlowReturn
gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
{
GstBuffer *buf;
gint size;
GstFlowReturn result = GST_FLOW_OK;
gint av;
if (!lame->lgf)
return GST_FLOW_OK;
@ -501,15 +709,31 @@ gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
buf = gst_buffer_new_and_alloc (7200);
size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
if (size > 0 && push) {
if (size > 0) {
GST_BUFFER_SIZE (buf) = size;
GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
GST_DEBUG_OBJECT (lame, "collecting final %d bytes", size);
gst_adapter_push (lame->adapter, buf);
} else {
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
gst_buffer_unref (buf);
result = GST_FLOW_OK;
}
if (push) {
result = gst_lamemp3enc_finish_frames (lame);
} else {
/* never mind */
gst_adapter_clear (lame->adapter);
}
/* either way, we expect nothing left */
if ((av = gst_adapter_available (lame->adapter))) {
/* should this be more fatal ?? */
GST_WARNING_OBJECT (lame, "unparsed %d bytes left after flushing", av);
/* clean up anyway */
gst_adapter_clear (lame->adapter);
}
return result;
}
@ -563,7 +787,10 @@ gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
if (G_LIKELY (mp3_size > 0)) {
GST_BUFFER_SIZE (mp3_buf) = mp3_size;
result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
/* unfortunately lame does not provide frame delineated output,
* so collect output and parse into frames ... */
gst_adapter_push (lame->adapter, mp3_buf);
result = gst_lamemp3enc_finish_frames (lame);
} else {
if (mp3_size < 0) {
/* eat error ? */

View file

@ -25,6 +25,7 @@
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
@ -54,6 +55,7 @@ struct _GstLameMP3Enc {
/*< private >*/
gint samplerate;
gint out_samplerate;
gint num_channels;
/* properties */
@ -65,6 +67,8 @@ struct _GstLameMP3Enc {
gboolean mono;
lame_global_flags *lgf;
GstAdapter *adapter;
};
struct _GstLameMP3EncClass {