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gst/mpegaudioparse/gstmpegaudioparse.*: Implement parsing of Xing headers from the first frame of the stream, and use...
Original commit message from CVS: * gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_type_frame_length_from_header), (gst_mp3parse_reset), (gst_mp3parse_emit_frame), (gst_mp3parse_handle_first_frame), (gst_mp3parse_chain), (mp3parse_total_bytes), (mp3parse_total_time): * gst/mpegaudioparse/gstmpegaudioparse.h: Implement parsing of Xing headers from the first frame of the stream, and use it to report duration correctly where possible.
This commit is contained in:
Jan Schmidt
parent
897f7039f8
commit
c145813ad3
3 changed files with 213 additions and 36 deletions
11
ChangeLog
11
ChangeLog
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@ -1,3 +1,14 @@
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2007-05-18 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/mpegaudioparse/gstmpegaudioparse.c:
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(mp3_type_frame_length_from_header), (gst_mp3parse_reset),
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(gst_mp3parse_emit_frame), (gst_mp3parse_handle_first_frame),
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(gst_mp3parse_chain), (mp3parse_total_bytes),
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(mp3parse_total_time):
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* gst/mpegaudioparse/gstmpegaudioparse.h:
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Implement parsing of Xing headers from the first frame of the stream,
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and use it to report duration correctly where possible.
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2007-05-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/realmedia/rmdemux.c: (gst_rmdemux_descramble_cook_audio):
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@ -21,6 +21,9 @@
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstmpegaudioparse.h"
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GST_DEBUG_CATEGORY_STATIC (mp3parse_debug);
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@ -138,8 +141,8 @@ static guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
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static inline guint
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mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
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guint * put_layer, guint * put_channels, guint * put_bitrate,
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guint * put_samplerate)
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guint * put_version, guint * put_layer, guint * put_channels,
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guint * put_bitrate, guint * put_samplerate)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding;
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@ -186,6 +189,8 @@ mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
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GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, layer = %lu, "
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"channels = %lu", samplerate, bitrate, layer, channels);
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if (put_version)
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*put_version = lsf ? 2 : 1;
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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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@ -268,11 +273,13 @@ gst_mp3parse_reset (GstMPEGAudioParse * mp3parse)
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gst_adapter_clear (mp3parse->adapter);
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mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
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mp3parse->version = 1;
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mp3parse->avg_bitrate = 0;
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mp3parse->bitrate_sum = 0;
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mp3parse->last_posted_bitrate = 0;
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mp3parse->frame_count = 0;
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mp3parse->sent_codec_tag = FALSE;
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}
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static void
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@ -390,26 +397,13 @@ static GstFlowReturn
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gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size)
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{
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GstBuffer *outbuf;
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gint spf; /* samples per frame */
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GST_DEBUG_OBJECT (mp3parse, "pushing buffer of %d bytes", size);
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outbuf = gst_adapter_take_buffer (mp3parse->adapter, size);
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/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
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if (mp3parse->layer == 1)
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spf = 384;
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else if (mp3parse->layer == 2)
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spf = 1152;
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else {
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/* Any sample_rate < 32000 indicates MPEG-2 or MPEG-2.5 */
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if (mp3parse->rate < 32000)
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spf = 576;
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else
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spf = 1152;
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}
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale (GST_SECOND, spf, mp3parse->rate);
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gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate);
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GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset;
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@ -441,12 +435,6 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size)
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}
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}
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/* Update our byte offset tracking */
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if (mp3parse->cur_offset != -1) {
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mp3parse->cur_offset += size;
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}
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mp3parse->tracked_offset += size;
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/* Decide what timestamp we're going to apply */
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if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) {
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GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts;
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@ -462,6 +450,12 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size)
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}
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}
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/* Update our byte offset tracking */
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if (mp3parse->cur_offset != -1) {
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mp3parse->cur_offset += size;
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}
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mp3parse->tracked_offset += size;
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mp3parse->next_ts =
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GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
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@ -482,6 +476,142 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size)
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return gst_pad_push (mp3parse->srcpad, outbuf);
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}
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#define XING_FRAMES_FLAG 0x0001
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#define XING_BYTES_FLAG 0x0002
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#define XING_TOC_FLAG 0x0004
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#define XING_VBR_SCALE_FLAG 0x0008
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static void
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gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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{
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GstTagList *taglist;
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gchar *codec;
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const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
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const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
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const guint XING_HDR_MIN = 8;
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gint xing_offset;
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guint64 avail;
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guint32 read_id;
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const guint8 *data;
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/* Output codec tag */
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if (!mp3parse->sent_codec_tag) {
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if (mp3parse->layer == 3) {
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codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)",
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mp3parse->version, mp3parse->layer);
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} else {
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codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
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mp3parse->version, mp3parse->layer);
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}
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taglist = gst_tag_list_new ();
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gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
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GST_TAG_AUDIO_CODEC, codec, NULL);
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gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
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mp3parse->srcpad, taglist);
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g_free (codec);
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mp3parse->sent_codec_tag = TRUE;
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}
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/* end setting the tag */
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/* Check first frame for Xing info */
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if (mp3parse->version == 1) { /* MPEG-1 file */
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if (mp3parse->channels == 1)
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xing_offset = 0x11;
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else
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xing_offset = 0x20;
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} else { /* MPEG-2 header */
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if (mp3parse->channels == 1)
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xing_offset = 0x09;
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else
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xing_offset = 0x11;
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}
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/* Skip the 4 bytes of the MP3 header too */
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xing_offset += 4;
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/* Check if we have enough data to read the Xing header */
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avail = gst_adapter_available (mp3parse->adapter);
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if (avail < xing_offset + XING_HDR_MIN)
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return;
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data = gst_adapter_peek (mp3parse->adapter, xing_offset + XING_HDR_MIN);
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if (data == NULL)
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return;
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/* The header starts at the provided offset */
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data += xing_offset;
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read_id = GST_READ_UINT32_BE (data);
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if (read_id == xing_id || read_id == info_id) {
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guint32 xing_flags;
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guint bytes_needed = xing_offset + XING_HDR_MIN;
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GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
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/* Read 4 base bytes of flags, big-endian */
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xing_flags = GST_READ_UINT32_BE (data + 4);
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if (xing_flags & XING_FRAMES_FLAG)
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bytes_needed += 4;
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if (xing_flags & XING_BYTES_FLAG)
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bytes_needed += 4;
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if (xing_flags & XING_TOC_FLAG)
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bytes_needed += 100;
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if (xing_flags & XING_VBR_SCALE_FLAG)
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bytes_needed += 4;
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if (avail < bytes_needed) {
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GST_DEBUG_OBJECT (mp3parse,
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"Not enough data to read Xing header (need %d)", bytes_needed);
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return;
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}
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GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
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mp3parse->xing_flags = xing_flags;
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data = gst_adapter_peek (mp3parse->adapter, bytes_needed);
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data += xing_offset + XING_HDR_MIN;
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if (xing_flags & XING_FRAMES_FLAG) {
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mp3parse->xing_frames = GST_READ_UINT32_BE (data);
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mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
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(guint64) (mp3parse->xing_frames) * (mp3parse->spf), mp3parse->rate);
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data += 4;
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} else {
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mp3parse->xing_frames = 0;
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mp3parse->xing_total_time = 0;
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}
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if (xing_flags & XING_BYTES_FLAG) {
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mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
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data += 4;
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} else
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mp3parse->xing_bytes = 0;
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if (xing_flags & XING_TOC_FLAG) {
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gint i;
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for (i = 0; i < 100; i++) {
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mp3parse->xing_seek_table[i] = data[0];
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data++;
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}
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} else {
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memset (mp3parse->xing_seek_table, 0, 100);
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}
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if (xing_flags & XING_VBR_SCALE_FLAG) {
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mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
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data += 4;
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} else
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mp3parse->xing_vbr_scale = 0;
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GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
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G_GUINT64_FORMAT ", vbr scale %u", mp3parse->xing_frames,
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mp3parse->xing_total_time, mp3parse->xing_vbr_scale);
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} else {
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GST_DEBUG_OBJECT (mp3parse, "Xing header not found in first frame");
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}
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}
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static GstFlowReturn
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gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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{
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@ -540,10 +670,10 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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header = GST_READ_UINT32_BE (data);
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/* if it's a valid header, go ahead and send off the frame */
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if (head_check (mp3parse, header)) {
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guint bitrate = 0, layer = 0, rate = 0, channels = 0;
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guint bitrate = 0, layer = 0, rate = 0, channels = 0, version;
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if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header, &layer,
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&channels, &bitrate, &rate)))
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if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header,
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&version, &layer, &channels, &bitrate, &rate)))
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goto header_error;
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/*************************************************************************
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@ -612,6 +742,24 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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mp3parse->layer = layer;
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mp3parse->rate = rate;
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mp3parse->bit_rate = bitrate;
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mp3parse->version = version;
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/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
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if (mp3parse->layer == 1)
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mp3parse->spf = 384;
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else if (mp3parse->layer == 2)
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mp3parse->spf = 1152;
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else if (mp3parse->version == 2) {
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mp3parse->spf = 576;
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} else
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mp3parse->spf = 1152;
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}
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/* Check the first frame for a Xing header to get our total length */
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if (mp3parse->frame_count == 0) {
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/* For the first frame in the file, look for a Xing frame after
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* the header, and output a codec tag */
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gst_mp3parse_handle_first_frame (mp3parse);
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}
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/* Update VBR stats */
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@ -813,21 +961,24 @@ mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total)
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GstQuery *query;
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GstPad *peer;
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if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) == NULL)
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return FALSE;
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if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
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query = gst_query_new_duration (GST_FORMAT_BYTES);
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gst_query_set_duration (query, GST_FORMAT_BYTES, -1);
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query = gst_query_new_duration (GST_FORMAT_BYTES);
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gst_query_set_duration (query, GST_FORMAT_BYTES, -1);
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if (!gst_pad_query (peer, query)) {
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if (gst_pad_query (peer, query)) {
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gst_object_unref (peer);
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gst_query_parse_duration (query, NULL, total);
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return TRUE;
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}
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gst_object_unref (peer);
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return FALSE;
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}
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gst_object_unref (peer);
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gst_query_parse_duration (query, NULL, total);
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if (mp3parse->xing_flags & XING_BYTES_FLAG) {
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*total = mp3parse->xing_bytes;
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return TRUE;
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}
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return TRUE;
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return FALSE;
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}
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static gboolean
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@ -837,6 +988,11 @@ mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total)
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*total = GST_CLOCK_TIME_NONE;
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if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
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*total = mp3parse->xing_total_time;
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return TRUE;
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}
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/* Calculate time from the measured bitrate */
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if (!mp3parse_total_bytes (mp3parse, &total_bytes))
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return FALSE;
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@ -61,16 +61,26 @@ struct _GstMPEGAudioParse {
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guint skip; /* number of frames to skip */
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guint bit_rate; /* in kbps */
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gint channels, rate, layer;
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gint channels, rate, layer, version;
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gint spf; /* Samples per frame */
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gboolean resyncing; /* True when attempting to resync (stricter checks are
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performed) */
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gboolean sent_codec_tag;
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/* VBR tracking */
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guint avg_bitrate;
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guint64 bitrate_sum;
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guint frame_count;
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guint last_posted_bitrate;
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/* Xing info */
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guint32 xing_flags;
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guint32 xing_frames;
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GstClockTime xing_total_time;
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guint32 xing_bytes;
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guchar xing_seek_table[100];
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guint32 xing_vbr_scale;
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};
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struct _GstMPEGAudioParseClass {
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