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lame: ensure parsed output
... by doing some basic parsing of encoded lame data.
This commit is contained in:
Mark Nauwelaerts
parent
e8cb13fea1
commit
d08945bccf
2 changed files with 234 additions and 4 deletions
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@ -573,6 +573,7 @@ gst_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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"output samplerate %d is different from incoming samplerate %d",
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out_samplerate, lame->samplerate);
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}
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lame->out_samplerate = out_samplerate;
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version = lame_get_version (lame->lgf);
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if (version == 0)
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@ -663,6 +664,10 @@ gst_lame_start (GstAudioEncoder * enc)
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{
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GstLame *lame = GST_LAME (enc);
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if (!lame->adapter)
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lame->adapter = gst_adapter_new ();
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gst_adapter_clear (lame->adapter);
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GST_DEBUG_OBJECT (lame, "start");
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return TRUE;
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}
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@ -674,6 +679,11 @@ gst_lame_stop (GstAudioEncoder * enc)
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GST_DEBUG_OBJECT (lame, "stop");
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if (lame->adapter) {
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g_object_unref (lame->adapter);
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lame->adapter = NULL;
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}
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gst_lame_release_memory (lame);
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return TRUE;
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}
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@ -944,12 +954,209 @@ gst_lame_get_property (GObject * object, guint prop_id, GValue * value,
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}
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}
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/* **** credits go to mpegaudioparse **** */
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static const guint mp3types_bitrates[2][3][16] = {
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{
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{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
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{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
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},
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{
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
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},
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};
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static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
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{22050, 24000, 16000},
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{11025, 12000, 8000}
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};
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static inline guint
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mp3_type_frame_length_from_header (GstLame * lame, guint32 header,
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guint * put_version, guint * put_layer, guint * put_channels,
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guint * put_bitrate, guint * put_samplerate, guint * put_mode,
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guint * put_crc)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding, crc;
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gulong version;
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gint lsf, mpg25;
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if (header & (1 << 20)) {
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lsf = (header & (1 << 19)) ? 0 : 1;
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mpg25 = 0;
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} else {
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lsf = 1;
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mpg25 = 1;
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}
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version = 1 + lsf + mpg25;
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layer = 4 - ((header >> 17) & 0x3);
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crc = (header >> 16) & 0x1;
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bitrate = (header >> 12) & 0xF;
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bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
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/* The caller has ensured we have a valid header, so bitrate can't be
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zero here. */
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g_assert (bitrate != 0);
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samplerate = (header >> 10) & 0x3;
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samplerate = mp3types_freqs[lsf + mpg25][samplerate];
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padding = (header >> 9) & 0x1;
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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switch (layer) {
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case 1:
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length = 4 * ((bitrate * 12) / samplerate + padding);
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break;
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case 2:
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length = (bitrate * 144) / samplerate + padding;
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break;
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default:
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case 3:
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length = (bitrate * 144) / (samplerate << lsf) + padding;
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break;
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}
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GST_DEBUG_OBJECT (lame, "Calculated mp3 frame length of %u bytes", length);
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GST_DEBUG_OBJECT (lame, "samplerate = %lu, bitrate = %lu, version = %lu, "
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"layer = %lu, channels = %lu", samplerate, bitrate, version,
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layer, channels);
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if (put_version)
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*put_version = version;
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if (put_layer)
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*put_layer = layer;
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if (put_channels)
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*put_channels = channels;
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if (put_bitrate)
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*put_bitrate = bitrate;
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if (put_samplerate)
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*put_samplerate = samplerate;
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if (put_mode)
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*put_mode = mode;
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if (put_crc)
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*put_crc = crc;
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return length;
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}
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static gboolean
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mp3_sync_check (GstLame * lame, unsigned long head)
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{
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GST_DEBUG_OBJECT (lame, "checking mp3 header 0x%08lx", head);
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/* if it's not a valid sync */
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if ((head & 0xffe00000) != 0xffe00000) {
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GST_WARNING_OBJECT (lame, "invalid sync");
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return FALSE;
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}
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/* if it's an invalid MPEG version */
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if (((head >> 19) & 3) == 0x1) {
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GST_WARNING_OBJECT (lame, "invalid MPEG version: 0x%lx", (head >> 19) & 3);
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return FALSE;
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}
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/* if it's an invalid layer */
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if (!((head >> 17) & 3)) {
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GST_WARNING_OBJECT (lame, "invalid layer: 0x%lx", (head >> 17) & 3);
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return FALSE;
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}
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/* if it's an invalid bitrate */
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if (((head >> 12) & 0xf) == 0x0) {
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GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx."
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"Free format files are not supported yet", (head >> 12) & 0xf);
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return FALSE;
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}
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if (((head >> 12) & 0xf) == 0xf) {
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GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
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return FALSE;
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}
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/* if it's an invalid samplerate */
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if (((head >> 10) & 0x3) == 0x3) {
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GST_WARNING_OBJECT (lame, "invalid samplerate: 0x%lx", (head >> 10) & 0x3);
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return FALSE;
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}
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if ((head & 0x3) == 0x2) {
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/* Ignore this as there are some files with emphasis 0x2 that can
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* be played fine. See BGO #537235 */
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GST_WARNING_OBJECT (lame, "invalid emphasis: 0x%lx", head & 0x3);
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}
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return TRUE;
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}
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/* **** end mpegaudioparse **** */
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static GstFlowReturn
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gst_lame_finish_frames (GstLame * lame)
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{
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gint av;
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guint header;
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GstFlowReturn result = GST_FLOW_OK;
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/* limited parsing, we don't expect to lose sync here */
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while ((result == GST_FLOW_OK) &&
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((av = gst_adapter_available (lame->adapter)) > 4)) {
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guint rate, version, layer, size;
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GstBuffer *mp3_buf;
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const guint8 *data;
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data = gst_adapter_peek (lame->adapter, 4);
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header = GST_READ_UINT32_BE (data);
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if (!mp3_sync_check (lame, header))
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goto invalid_header;
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size = mp3_type_frame_length_from_header (lame, header, &version, &layer,
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NULL, NULL, &rate, NULL, NULL);
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if (G_UNLIKELY (layer != 3 || rate != lame->out_samplerate)) {
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GST_DEBUG_OBJECT (lame,
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"unexpected mp3 header with (rate, layer): (%u, %u)",
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rate, version, layer);
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goto invalid_header;
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}
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if (size > av) {
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/* pretty likely to occur when lame is holding back on us */
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GST_LOG_OBJECT (lame, "frame size %u (> %d)", size, av);
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break;
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}
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/* should be ok now */
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mp3_buf = gst_adapter_take_buffer (lame->adapter, size);
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/* number of samples for MPEG-1, layer 3 */
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result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame),
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mp3_buf, version == 1 ? 1152 : 576);
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}
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exit:
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return result;
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/* ERRORS */
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invalid_header:
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{
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GST_ELEMENT_ERROR (lame, STREAM, ENCODE,
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("invalid lame mp3 sync header %08X", header), (NULL));
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result = GST_FLOW_ERROR;
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goto exit;
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}
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}
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static GstFlowReturn
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gst_lame_flush_full (GstLame * lame, gboolean push)
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{
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GstBuffer *buf;
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gint size;
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GstFlowReturn result = GST_FLOW_OK;
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gint av;
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if (!lame->lgf)
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return GST_FLOW_OK;
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@ -957,15 +1164,31 @@ gst_lame_flush_full (GstLame * lame, gboolean push)
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buf = gst_buffer_new_and_alloc (7200);
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size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
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if (size > 0 && push) {
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if (size > 0) {
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GST_BUFFER_SIZE (buf) = size;
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GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
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result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
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GST_DEBUG_OBJECT (lame, "collecting final %d bytes", size);
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gst_adapter_push (lame->adapter, buf);
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} else {
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GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
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gst_buffer_unref (buf);
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result = GST_FLOW_OK;
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}
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if (push) {
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result = gst_lame_finish_frames (lame);
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} else {
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/* never mind */
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gst_adapter_clear (lame->adapter);
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}
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/* either way, we expect nothing left */
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if ((av = gst_adapter_available (lame->adapter))) {
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/* should this be more fatal ?? */
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GST_WARNING_OBJECT (lame, "unparsed %d bytes left after flushing", av);
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/* clean up anyway */
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gst_adapter_clear (lame->adapter);
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}
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return result;
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}
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@ -1023,7 +1246,10 @@ gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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if (G_LIKELY (mp3_size > 0)) {
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GST_BUFFER_SIZE (mp3_buf) = mp3_size;
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result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
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/* unfortunately lame does not provide frame delineated output,
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* so collect output and parse into frames ... */
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gst_adapter_push (lame->adapter, mp3_buf);
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result = gst_lame_finish_frames (lame);
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} else {
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if (mp3_size < 0) {
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/* eat error ? */
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@ -28,6 +28,7 @@ G_BEGIN_DECLS
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#include <lame/lame.h>
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#include <gst/audio/gstaudioencoder.h>
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#include <gst/base/gstadapter.h>
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#define GST_TYPE_LAME \
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(gst_lame_get_type())
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@ -54,6 +55,7 @@ struct _GstLame {
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/*< private >*/
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gint samplerate;
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gint out_samplerate;
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gint num_channels;
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gboolean setup;
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@ -91,6 +93,8 @@ struct _GstLame {
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gint preset;
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lame_global_flags *lgf;
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GstAdapter *adapter;
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};
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struct _GstLameClass {
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