From c145813ad3b3aa4f207cee98fe3f7041f420a7d2 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Fri, 18 May 2007 08:42:25 +0000 Subject: [PATCH] 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. --- ChangeLog | 11 ++ gst/mpegaudioparse/gstmpegaudioparse.c | 226 +++++++++++++++++++++---- gst/mpegaudioparse/gstmpegaudioparse.h | 12 +- 3 files changed, 213 insertions(+), 36 deletions(-) diff --git a/ChangeLog b/ChangeLog index a721b4c821..7af2be61f7 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,14 @@ +2007-05-18 Jan Schmidt + + * 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. + 2007-05-16 Tim-Philipp Müller * gst/realmedia/rmdemux.c: (gst_rmdemux_descramble_cook_audio): diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c index b5046ec04e..a36dd291fc 100644 --- a/gst/mpegaudioparse/gstmpegaudioparse.c +++ b/gst/mpegaudioparse/gstmpegaudioparse.c @@ -21,6 +21,9 @@ #ifdef HAVE_CONFIG_H #include "config.h" #endif + +#include + #include "gstmpegaudioparse.h" GST_DEBUG_CATEGORY_STATIC (mp3parse_debug); @@ -138,8 +141,8 @@ static guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, static inline guint mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header, - guint * put_layer, guint * put_channels, guint * put_bitrate, - guint * put_samplerate) + guint * put_version, guint * put_layer, guint * put_channels, + guint * put_bitrate, guint * put_samplerate) { guint length; gulong mode, samplerate, bitrate, layer, channels, padding; @@ -186,6 +189,8 @@ mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header, GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, layer = %lu, " "channels = %lu", samplerate, bitrate, layer, channels); + if (put_version) + *put_version = lsf ? 2 : 1; if (put_layer) *put_layer = layer; if (put_channels) @@ -268,11 +273,13 @@ gst_mp3parse_reset (GstMPEGAudioParse * mp3parse) gst_adapter_clear (mp3parse->adapter); mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; + mp3parse->version = 1; mp3parse->avg_bitrate = 0; mp3parse->bitrate_sum = 0; mp3parse->last_posted_bitrate = 0; mp3parse->frame_count = 0; + mp3parse->sent_codec_tag = FALSE; } static void @@ -390,26 +397,13 @@ static GstFlowReturn gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size) { GstBuffer *outbuf; - gint spf; /* samples per frame */ GST_DEBUG_OBJECT (mp3parse, "pushing buffer of %d bytes", size); outbuf = gst_adapter_take_buffer (mp3parse->adapter, size); - /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */ - if (mp3parse->layer == 1) - spf = 384; - else if (mp3parse->layer == 2) - spf = 1152; - else { - /* Any sample_rate < 32000 indicates MPEG-2 or MPEG-2.5 */ - if (mp3parse->rate < 32000) - spf = 576; - else - spf = 1152; - } GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale (GST_SECOND, spf, mp3parse->rate); + gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate); GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset; @@ -441,12 +435,6 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size) } } - /* Update our byte offset tracking */ - if (mp3parse->cur_offset != -1) { - mp3parse->cur_offset += size; - } - mp3parse->tracked_offset += size; - /* Decide what timestamp we're going to apply */ if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) { GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts; @@ -462,6 +450,12 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size) } } + /* Update our byte offset tracking */ + if (mp3parse->cur_offset != -1) { + mp3parse->cur_offset += size; + } + mp3parse->tracked_offset += size; + mp3parse->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); @@ -482,6 +476,142 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size) return gst_pad_push (mp3parse->srcpad, outbuf); } +#define XING_FRAMES_FLAG 0x0001 +#define XING_BYTES_FLAG 0x0002 +#define XING_TOC_FLAG 0x0004 +#define XING_VBR_SCALE_FLAG 0x0008 + +static void +gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse) +{ + GstTagList *taglist; + gchar *codec; + const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */ + const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */ + const guint XING_HDR_MIN = 8; + gint xing_offset; + + guint64 avail; + guint32 read_id; + const guint8 *data; + + /* Output codec tag */ + if (!mp3parse->sent_codec_tag) { + if (mp3parse->layer == 3) { + codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)", + mp3parse->version, mp3parse->layer); + } else { + codec = g_strdup_printf ("MPEG %d Audio, Layer %d", + mp3parse->version, mp3parse->layer); + } + + taglist = gst_tag_list_new (); + gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, + GST_TAG_AUDIO_CODEC, codec, NULL); + gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse), + mp3parse->srcpad, taglist); + g_free (codec); + + mp3parse->sent_codec_tag = TRUE; + } + /* end setting the tag */ + + /* Check first frame for Xing info */ + if (mp3parse->version == 1) { /* MPEG-1 file */ + if (mp3parse->channels == 1) + xing_offset = 0x11; + else + xing_offset = 0x20; + } else { /* MPEG-2 header */ + if (mp3parse->channels == 1) + xing_offset = 0x09; + else + xing_offset = 0x11; + } + /* Skip the 4 bytes of the MP3 header too */ + xing_offset += 4; + + /* Check if we have enough data to read the Xing header */ + avail = gst_adapter_available (mp3parse->adapter); + + if (avail < xing_offset + XING_HDR_MIN) + return; + + data = gst_adapter_peek (mp3parse->adapter, xing_offset + XING_HDR_MIN); + if (data == NULL) + return; + /* The header starts at the provided offset */ + data += xing_offset; + + read_id = GST_READ_UINT32_BE (data); + if (read_id == xing_id || read_id == info_id) { + guint32 xing_flags; + guint bytes_needed = xing_offset + XING_HDR_MIN; + + GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id); + + /* Read 4 base bytes of flags, big-endian */ + xing_flags = GST_READ_UINT32_BE (data + 4); + if (xing_flags & XING_FRAMES_FLAG) + bytes_needed += 4; + if (xing_flags & XING_BYTES_FLAG) + bytes_needed += 4; + if (xing_flags & XING_TOC_FLAG) + bytes_needed += 100; + if (xing_flags & XING_VBR_SCALE_FLAG) + bytes_needed += 4; + if (avail < bytes_needed) { + GST_DEBUG_OBJECT (mp3parse, + "Not enough data to read Xing header (need %d)", bytes_needed); + return; + } + + GST_DEBUG_OBJECT (mp3parse, "Reading Xing header"); + mp3parse->xing_flags = xing_flags; + data = gst_adapter_peek (mp3parse->adapter, bytes_needed); + data += xing_offset + XING_HDR_MIN; + + if (xing_flags & XING_FRAMES_FLAG) { + mp3parse->xing_frames = GST_READ_UINT32_BE (data); + mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND, + (guint64) (mp3parse->xing_frames) * (mp3parse->spf), mp3parse->rate); + data += 4; + } else { + mp3parse->xing_frames = 0; + mp3parse->xing_total_time = 0; + } + + if (xing_flags & XING_BYTES_FLAG) { + mp3parse->xing_bytes = GST_READ_UINT32_BE (data); + data += 4; + } else + mp3parse->xing_bytes = 0; + + if (xing_flags & XING_TOC_FLAG) { + gint i; + + for (i = 0; i < 100; i++) { + mp3parse->xing_seek_table[i] = data[0]; + data++; + } + } else { + memset (mp3parse->xing_seek_table, 0, 100); + } + + if (xing_flags & XING_VBR_SCALE_FLAG) { + mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data); + data += 4; + } else + mp3parse->xing_vbr_scale = 0; + + GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %" + G_GUINT64_FORMAT ", vbr scale %u", mp3parse->xing_frames, + mp3parse->xing_total_time, mp3parse->xing_vbr_scale); + } else { + GST_DEBUG_OBJECT (mp3parse, "Xing header not found in first frame"); + } +} + static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) { @@ -540,10 +670,10 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) header = GST_READ_UINT32_BE (data); /* if it's a valid header, go ahead and send off the frame */ if (head_check (mp3parse, header)) { - guint bitrate = 0, layer = 0, rate = 0, channels = 0; + guint bitrate = 0, layer = 0, rate = 0, channels = 0, version; - if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header, &layer, - &channels, &bitrate, &rate))) + if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header, + &version, &layer, &channels, &bitrate, &rate))) goto header_error; /************************************************************************* @@ -612,6 +742,24 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) mp3parse->layer = layer; mp3parse->rate = rate; mp3parse->bit_rate = bitrate; + mp3parse->version = version; + + /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */ + if (mp3parse->layer == 1) + mp3parse->spf = 384; + else if (mp3parse->layer == 2) + mp3parse->spf = 1152; + else if (mp3parse->version == 2) { + mp3parse->spf = 576; + } else + mp3parse->spf = 1152; + } + + /* Check the first frame for a Xing header to get our total length */ + if (mp3parse->frame_count == 0) { + /* For the first frame in the file, look for a Xing frame after + * the header, and output a codec tag */ + gst_mp3parse_handle_first_frame (mp3parse); } /* Update VBR stats */ @@ -813,21 +961,24 @@ mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total) GstQuery *query; GstPad *peer; - if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) == NULL) - return FALSE; + if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) { + query = gst_query_new_duration (GST_FORMAT_BYTES); + gst_query_set_duration (query, GST_FORMAT_BYTES, -1); - query = gst_query_new_duration (GST_FORMAT_BYTES); - gst_query_set_duration (query, GST_FORMAT_BYTES, -1); - - if (!gst_pad_query (peer, query)) { + if (gst_pad_query (peer, query)) { + gst_object_unref (peer); + gst_query_parse_duration (query, NULL, total); + return TRUE; + } gst_object_unref (peer); - return FALSE; } - gst_object_unref (peer); - gst_query_parse_duration (query, NULL, total); + if (mp3parse->xing_flags & XING_BYTES_FLAG) { + *total = mp3parse->xing_bytes; + return TRUE; + } - return TRUE; + return FALSE; } static gboolean @@ -837,6 +988,11 @@ mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total) *total = GST_CLOCK_TIME_NONE; + if (mp3parse->xing_flags & XING_FRAMES_FLAG) { + *total = mp3parse->xing_total_time; + return TRUE; + } + /* Calculate time from the measured bitrate */ if (!mp3parse_total_bytes (mp3parse, &total_bytes)) return FALSE; diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h index 101db8ee67..a405fcf0f7 100644 --- a/gst/mpegaudioparse/gstmpegaudioparse.h +++ b/gst/mpegaudioparse/gstmpegaudioparse.h @@ -61,16 +61,26 @@ struct _GstMPEGAudioParse { guint skip; /* number of frames to skip */ guint bit_rate; /* in kbps */ - gint channels, rate, layer; + gint channels, rate, layer, version; + gint spf; /* Samples per frame */ gboolean resyncing; /* True when attempting to resync (stricter checks are performed) */ + gboolean sent_codec_tag; /* VBR tracking */ guint avg_bitrate; guint64 bitrate_sum; guint frame_count; guint last_posted_bitrate; + + /* Xing info */ + guint32 xing_flags; + guint32 xing_frames; + GstClockTime xing_total_time; + guint32 xing_bytes; + guchar xing_seek_table[100]; + guint32 xing_vbr_scale; }; struct _GstMPEGAudioParseClass {