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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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mpegaudioparse: support parsing freeform bitrate stream
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parent
1fe69911a4
commit
0d5b5d839a
2 changed files with 156 additions and 14 deletions
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@ -200,6 +200,7 @@ gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
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mp3parse->sent_codec_tag = FALSE;
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mp3parse->last_posted_crc = CRC_UNKNOWN;
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mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
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mp3parse->freerate = 0;
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mp3parse->hdr_bitrate = 0;
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@ -307,14 +308,16 @@ mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
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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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if (!bitrate) {
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GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
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bitrate = mp3parse->freerate;
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}
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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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/* force 0 length if 0 bitrate */
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padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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@ -419,8 +422,7 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
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(guint) next_header & HDRMASK, bpf);
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*valid = FALSE;
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return TRUE;
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} else if ((((next_header >> 12) & 0xf) == 0) ||
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(((next_header >> 12) & 0xf) == 0xf)) {
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} else if (((next_header >> 12) & 0xf) == 0xf) {
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/* The essential parts were the same, but the bitrate held an
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invalid value - also reject */
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GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
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@ -431,6 +433,13 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
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bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
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NULL, NULL, NULL, NULL, NULL, NULL, NULL);
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/* if no bitrate, and no freeform rate known, then fail */
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if (G_UNLIKELY (!bpf)) {
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GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
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*valid = FALSE;
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return TRUE;
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}
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offset += bpf;
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frames_found++;
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}
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@ -461,11 +470,6 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
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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 (mp3parse, "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 (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
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return FALSE;
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@ -486,6 +490,115 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
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return TRUE;
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}
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/* Determines possible freeform frame rate/size by looking for next
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* header with valid bitrate (0 or otherwise valid) (and sufficiently
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* matching current header).
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*
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* Returns TRUE if we've found such one, and *rate then contains rate
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* (or *rate contains 0 if decided no freeframe size could be determined).
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* If not enough data, returns FALSE.
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*/
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static gboolean
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gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstBuffer * buf,
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guint32 header, gboolean at_eos, gint * _rate)
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{
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guint32 next_header;
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const guint8 *data;
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guint available;
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int offset = 4;
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gulong samplerate, rate, layer, padding;
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gboolean valid;
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gint lsf, mpg25;
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available = GST_BUFFER_SIZE (buf);
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data = GST_BUFFER_DATA (buf);
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*_rate = 0;
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/* pick apart header again partially */
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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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layer = 4 - ((header >> 17) & 0x3);
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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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for (; offset < available; ++offset) {
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/* Check if we have enough data for all these frames, plus the next
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frame header. */
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if (available < offset + 4) {
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if (at_eos) {
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/* Running out of data; failed to determine size */
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return TRUE;
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} else {
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return FALSE;
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}
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}
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valid = FALSE;
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next_header = GST_READ_UINT32_BE (data + offset);
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if ((next_header & 0xFFE00000) != 0xFFE00000)
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goto next;
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GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
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offset, (unsigned int) header, (unsigned int) next_header);
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if ((next_header & HDRMASK) != (header & HDRMASK)) {
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/* If any of the unmasked bits don't match, then it's not valid */
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GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
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"(header=%08X (%08X), header2=%08X (%08X))",
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(guint) header, (guint) header & HDRMASK, (guint) next_header,
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(guint) next_header & HDRMASK);
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goto next;
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} else if (((next_header >> 12) & 0xf) == 0xf) {
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/* The essential parts were the same, but the bitrate held an
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invalid value - also reject */
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GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
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goto next;
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}
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valid = TRUE;
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next:
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/* almost accept as free frame */
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if (layer == 1) {
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rate = samplerate * (offset - 4 * padding + 4) / 48000;
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} else {
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rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
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}
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if (valid) {
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GST_LOG_OBJECT (mp3parse, "calculated rate %d", rate * 1000);
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if (rate < 8 || (layer == 3 && rate > 640)) {
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GST_DEBUG_OBJECT (mp3parse, "rate invalid");
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if (rate < 8) {
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/* maybe some hope */
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continue;
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} else {
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GST_DEBUG_OBJECT (mp3parse, "aborting");
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/* give up */
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break;
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}
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}
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*_rate = rate * 1000;
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break;
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} else {
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/* avoid indefinite searching */
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if (rate > 1000) {
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GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
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break;
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}
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}
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}
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return TRUE;
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}
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static gboolean
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gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
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@ -527,9 +640,14 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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GST_LOG_OBJECT (parse, "got frame");
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lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
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draining = GST_BASE_PARSE_DRAINING (parse);
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if (G_UNLIKELY (lost_sync))
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mp3parse->freerate = 0;
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bpf = mp3_type_frame_length_from_header (mp3parse, header,
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&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
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g_assert (bpf != 0);
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if (channels != mp3parse->channels || rate != mp3parse->rate ||
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layer != mp3parse->layer || version != mp3parse->version)
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@ -537,8 +655,30 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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else
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caps_change = FALSE;
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lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
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draining = GST_BASE_PARSE_DRAINING (parse);
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/* maybe free format */
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if (bpf == 0) {
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GST_LOG_OBJECT (mp3parse, "possibly free format");
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if (lost_sync || mp3parse->freerate == 0) {
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GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
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if (!gst_mp3parse_find_freerate (mp3parse, buf, header, draining, &valid)) {
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/* not enough data */
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*framesize = G_MAXUINT;
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*skipsize = 0;
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return FALSE;
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} else {
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GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
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mp3parse->freerate = valid;
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}
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}
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/* try again */
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bpf = mp3_type_frame_length_from_header (mp3parse, header,
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&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
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if (!bpf) {
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/* did not come up with valid freeform length, reject after all */
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*skipsize = 1;
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return FALSE;
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}
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}
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if (!draining && (lost_sync || caps_change)) {
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if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
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@ -60,6 +60,8 @@ struct _GstMpegAudioParse {
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/* samples per frame */
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gint spf;
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gint freerate;
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gboolean sent_codec_tag;
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guint last_posted_bitrate;
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gint last_posted_crc, last_crc;
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