Mercurial > libavformat.hg
annotate rtp_asf.c @ 5349:d2cdd13aacff libavformat
Initialize block_parsed before reading first block header in WavPack demuxer,
it will be useful later.
author | kostya |
---|---|
date | Thu, 05 Nov 2009 08:10:50 +0000 |
parents | 51b1a9987537 |
children | 8fddfdd5e9e7 |
rev | line source |
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1 /* |
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2 * Microsoft RTP/ASF support. |
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3 * Copyright (c) 2008 Ronald S. Bultje |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
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21 |
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22 /** |
4746 | 23 * @file libavformat/rtp_asf.c |
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24 * @brief Microsoft RTP/ASF support |
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25 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net> |
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26 */ |
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27 |
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28 #include <libavutil/base64.h> |
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29 #include <libavutil/avstring.h> |
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30 #include <libavutil/intreadwrite.h> |
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31 #include "rtp.h" |
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32 #include "rtp_asf.h" |
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33 #include "rtsp.h" |
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34 #include "asf.h" |
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35 |
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36 /** |
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37 * From MSDN 2.2.1.4, we learn that ASF data packets over RTP should not |
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38 * contain any padding. Unfortunately, the header min/max_pktsize are not |
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39 * updated (thus making min_pktsize invalid). Here, we "fix" these faulty |
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40 * min_pktsize values in the ASF file header. |
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41 * @return 0 on success, <0 on failure (currently -1). |
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42 */ |
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43 static int rtp_asf_fix_header(uint8_t *buf, int len) |
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44 { |
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45 uint8_t *p = buf, *end = buf + len; |
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46 |
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47 if (len < sizeof(ff_asf_guid) * 2 + 22 || |
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48 memcmp(p, ff_asf_header, sizeof(ff_asf_guid))) { |
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49 return -1; |
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50 } |
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51 p += sizeof(ff_asf_guid) + 14; |
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52 do { |
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53 uint64_t chunksize = AV_RL64(p + sizeof(ff_asf_guid)); |
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54 if (memcmp(p, ff_asf_file_header, sizeof(ff_asf_guid))) { |
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55 if (chunksize > end - p) |
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56 return -1; |
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57 p += chunksize; |
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58 continue; |
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59 } |
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60 |
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61 /* skip most of the file header, to min_pktsize */ |
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62 p += 6 * 8 + 3 * 4 + sizeof(ff_asf_guid) * 2; |
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63 if (p + 8 <= end && AV_RL32(p) == AV_RL32(p + 4)) { |
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64 /* and set that to zero */ |
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65 AV_WL32(p, 0); |
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66 return 0; |
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67 } |
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68 break; |
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69 } while (end - p >= sizeof(ff_asf_guid) + 8); |
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70 |
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71 return -1; |
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72 } |
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73 |
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74 /** |
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75 * The following code is basically a buffered ByteIOContext, |
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76 * with the added benefit of returning -EAGAIN (instead of 0) |
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77 * on packet boundaries, such that the ASF demuxer can return |
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78 * safely and resume business at the next packet. |
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79 */ |
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80 static int packetizer_read(void *opaque, uint8_t *buf, int buf_size) |
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81 { |
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82 return AVERROR(EAGAIN); |
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83 } |
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84 |
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85 static void init_packetizer(ByteIOContext *pb, uint8_t *buf, int len) |
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86 { |
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87 init_put_byte(pb, buf, len, 0, NULL, packetizer_read, NULL, NULL); |
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88 |
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89 /* this "fills" the buffer with its current content */ |
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90 pb->pos = len; |
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91 pb->buf_end = buf + len; |
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92 } |
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93 |
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94 void ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p) |
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95 { |
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96 if (av_strstart(p, "pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,", &p)) { |
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97 ByteIOContext pb; |
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98 RTSPState *rt = s->priv_data; |
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99 int len = strlen(p) * 6 / 8; |
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100 char *buf = av_mallocz(len); |
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101 av_base64_decode(buf, p, len); |
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102 |
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103 if (rtp_asf_fix_header(buf, len) < 0) |
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104 av_log(s, AV_LOG_ERROR, |
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105 "Failed to fix invalid RTSP-MS/ASF min_pktsize\n"); |
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106 init_packetizer(&pb, buf, len); |
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107 if (rt->asf_ctx) { |
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108 av_close_input_stream(rt->asf_ctx); |
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109 rt->asf_ctx = NULL; |
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110 } |
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111 av_open_input_stream(&rt->asf_ctx, &pb, "", &asf_demuxer, NULL); |
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112 rt->asf_pb_pos = url_ftell(&pb); |
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113 av_free(buf); |
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114 rt->asf_ctx->pb = NULL; |
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115 } |
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116 } |
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117 |
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118 static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index, |
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119 PayloadContext *asf, const char *line) |
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120 { |
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121 if (av_strstart(line, "stream:", &line)) { |
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122 RTSPState *rt = s->priv_data; |
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123 |
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124 s->streams[stream_index]->id = strtol(line, NULL, 10); |
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125 |
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126 if (rt->asf_ctx) { |
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127 int i; |
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128 |
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129 for (i = 0; i < rt->asf_ctx->nb_streams; i++) { |
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130 if (s->streams[stream_index]->id == rt->asf_ctx->streams[i]->id) { |
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131 *s->streams[stream_index]->codec = |
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132 *rt->asf_ctx->streams[i]->codec; |
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133 rt->asf_ctx->streams[i]->codec->extradata_size = 0; |
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134 rt->asf_ctx->streams[i]->codec->extradata = NULL; |
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135 av_set_pts_info(s->streams[stream_index], 32, 1, 1000); |
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136 } |
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137 } |
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138 } |
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139 } |
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140 |
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141 return 0; |
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142 } |
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143 |
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144 struct PayloadContext { |
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145 ByteIOContext *pktbuf, pb; |
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146 char *buf; |
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147 }; |
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148 |
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149 /** |
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150 * @return 0 when a packet was written into /p pkt, and no more data is left; |
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151 * 1 when a packet was written into /p pkt, and more packets might be left; |
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152 * <0 when not enough data was provided to return a full packet, or on error. |
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153 */ |
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154 static int asfrtp_parse_packet(AVFormatContext *s, PayloadContext *asf, |
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155 AVStream *st, AVPacket *pkt, |
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156 uint32_t *timestamp, |
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157 const uint8_t *buf, int len, int flags) |
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158 { |
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159 ByteIOContext *pb = &asf->pb; |
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160 int res, mflags, len_off; |
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161 RTSPState *rt = s->priv_data; |
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162 |
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163 if (!rt->asf_ctx) |
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164 return -1; |
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165 |
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166 if (len > 0) { |
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167 int off, out_len; |
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168 |
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169 if (len < 4) |
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170 return -1; |
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171 |
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172 init_put_byte(pb, buf, len, 0, NULL, NULL, NULL, NULL); |
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173 mflags = get_byte(pb); |
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174 if (mflags & 0x80) |
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175 flags |= RTP_FLAG_KEY; |
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176 len_off = get_be24(pb); |
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177 if (mflags & 0x20) /**< relative timestamp */ |
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178 url_fskip(pb, 4); |
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179 if (mflags & 0x10) /**< has duration */ |
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180 url_fskip(pb, 4); |
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181 if (mflags & 0x8) /**< has location ID */ |
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182 url_fskip(pb, 4); |
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183 off = url_ftell(pb); |
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184 |
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185 av_freep(&asf->buf); |
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186 if (!(mflags & 0x40)) { |
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187 /** |
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188 * If 0x40 is not set, the len_off field specifies an offset of this |
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189 * packet's payload data in the complete (reassembled) ASF packet. |
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190 * This is used to spread one ASF packet over multiple RTP packets. |
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191 */ |
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192 if (asf->pktbuf && len_off != url_ftell(asf->pktbuf)) { |
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193 uint8_t *p; |
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194 url_close_dyn_buf(asf->pktbuf, &p); |
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195 asf->pktbuf = NULL; |
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196 av_free(p); |
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197 } |
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198 if (!len_off && !asf->pktbuf && |
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199 !(res = url_open_dyn_packet_buf(&asf->pktbuf, rt->asf_ctx->packet_size))) |
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200 return res; |
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201 if (!asf->pktbuf) |
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202 return AVERROR(EIO); |
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203 |
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204 put_buffer(asf->pktbuf, buf + off, len - off); |
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205 if (!(flags & RTP_FLAG_MARKER)) |
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206 return -1; |
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207 out_len = url_close_dyn_buf(asf->pktbuf, &asf->buf); |
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208 asf->pktbuf = NULL; |
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209 } else { |
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210 /** |
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211 * If 0x40 is set, the len_off field specifies the length of the |
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212 * next ASF packet that can be read from this payload data alone. |
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213 * This is commonly the same as the payload size, but could be |
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214 * less in case of packet splitting (i.e. multiple ASF packets in |
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215 * one RTP packet). |
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216 */ |
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217 if (len_off != len) { |
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218 av_log_missing_feature(s, |
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219 "RTSP-MS packet splitting", 1); |
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220 return -1; |
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221 } |
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222 asf->buf = av_malloc(len - off); |
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223 out_len = len - off; |
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224 memcpy(asf->buf, buf + off, len - off); |
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225 } |
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226 |
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227 init_packetizer(pb, asf->buf, out_len); |
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228 pb->pos += rt->asf_pb_pos; |
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229 pb->eof_reached = 0; |
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230 rt->asf_ctx->pb = pb; |
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231 } |
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232 |
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233 for (;;) { |
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234 int i; |
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235 |
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236 res = av_read_packet(rt->asf_ctx, pkt); |
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237 rt->asf_pb_pos = url_ftell(pb); |
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238 if (res != 0) |
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239 break; |
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240 for (i = 0; i < s->nb_streams; i++) { |
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241 if (s->streams[i]->id == rt->asf_ctx->streams[pkt->stream_index]->id) { |
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242 pkt->stream_index = i; |
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243 return 1; // FIXME: return 0 if last packet |
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244 } |
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245 } |
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246 av_free_packet(pkt); |
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247 } |
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248 |
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249 return res == 1 ? -1 : res; |
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250 } |
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251 |
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252 static PayloadContext *asfrtp_new_context(void) |
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253 { |
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254 return av_mallocz(sizeof(PayloadContext)); |
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255 } |
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256 |
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257 static void asfrtp_free_context(PayloadContext *asf) |
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258 { |
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259 if (asf->pktbuf) { |
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260 uint8_t *p = NULL; |
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261 url_close_dyn_buf(asf->pktbuf, &p); |
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262 asf->pktbuf = NULL; |
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263 av_free(p); |
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264 } |
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265 av_freep(&asf->buf); |
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266 av_free(asf); |
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267 } |
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268 |
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269 #define RTP_ASF_HANDLER(n, s, t) \ |
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270 RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \ |
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271 .enc_name = s, \ |
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272 .codec_type = t, \ |
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273 .codec_id = CODEC_ID_NONE, \ |
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274 .parse_sdp_a_line = asfrtp_parse_sdp_line, \ |
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275 .open = asfrtp_new_context, \ |
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276 .close = asfrtp_free_context, \ |
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277 .parse_packet = asfrtp_parse_packet, \ |
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278 }; |
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279 |
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280 RTP_ASF_HANDLER(asf_pfv, "x-asf-pf", CODEC_TYPE_VIDEO); |
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281 RTP_ASF_HANDLER(asf_pfa, "x-asf-pf", CODEC_TYPE_AUDIO); |