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-rw-r--r--ext/lame/Makefile.am4
-rw-r--r--ext/lame/gstlamemp3enc.c235
-rw-r--r--ext/lame/gstlamemp3enc.h4
3 files changed, 237 insertions, 6 deletions
diff --git a/ext/lame/Makefile.am b/ext/lame/Makefile.am
index 439a621d..79143234 100644
--- a/ext/lame/Makefile.am
+++ b/ext/lame/Makefile.am
@@ -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
diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
index b0d351bc..5ea9f17f 100644
--- a/ext/lame/gstlamemp3enc.c
+++ b/ext/lame/gstlamemp3enc.c
@@ -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 ? */
diff --git a/ext/lame/gstlamemp3enc.h b/ext/lame/gstlamemp3enc.h
index 9bf12254..d51263e1 100644
--- a/ext/lame/gstlamemp3enc.h
+++ b/ext/lame/gstlamemp3enc.h
@@ -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 {
generated by cgit v1.2.3 (git 2.39.1) at 2024-04-25 01:10:52 +0000