}
}
- data = gst_adapter_peek (lame->adapter, 4);
+ /* **** 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_map (lame->adapter, 4);
+ header = GST_READ_UINT32_BE (data);
++ gst_adapter_unmap (lame->adapter);
++
+ 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 %u, version %u, layer %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;
+ guint8 *data;
GstFlowReturn result = GST_FLOW_OK;
+ gint av;
if (!lame->lgf)
return GST_FLOW_OK;
buf = gst_buffer_new_and_alloc (7200);
- size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
+ data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
+ size = lame_encode_flush (lame->lgf, data, 7200);
if (size > 0) {
- GST_BUFFER_SIZE (buf) = size;
+ gst_buffer_unmap (buf, data, size);
- if (push) {
- 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_buffer_unmap (buf, data, 0);
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
gst_buffer_unref (buf);
result = GST_FLOW_OK;
(short int *) data,
num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
}
+ gst_buffer_unmap (in_buf, data, size);
- GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
- size, mp3_size);
+ GST_LOG_OBJECT (lame, "encoded %" G_GSIZE_FORMAT " bytes of audio "
+ "to %d bytes of mp3", size, mp3_size);
if (G_LIKELY (mp3_size > 0)) {
- GST_BUFFER_SIZE (mp3_buf) = mp3_size;
+ /* unfortunately lame does not provide frame delineated output,
+ * so collect output and parse into frames ... */
+ gst_buffer_unmap (mp3_buf, mp3_data, mp3_size);
- result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
+ gst_adapter_push (lame->adapter, mp3_buf);
+ result = gst_lamemp3enc_finish_frames (lame);
} else {
+ gst_buffer_unmap (mp3_buf, mp3_data, 0);
if (mp3_size < 0) {
/* eat error ? */
g_warning ("error %d", mp3_size);
projects / platform / upstream / gst-plugins-good.git / commitdiff