Mercurial > libavformat.hg
annotate rtpenc.c @ 6349:93c7a56fa912 libavformat
Add RTP packetization of Theora and Vorbis
Patch by Josh Allmann, joshua dot allmann at gmail
author | mstorsjo |
---|---|
date | Sat, 07 Aug 2010 11:16:07 +0000 |
parents | 956f2e2cd288 |
children | 0bcd6a8bc5d5 |
rev | line source |
---|---|
0 | 1 /* |
2892
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Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies
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2 * RTP output format |
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cosmetics: Remove pointless period after copyright statement non-sentences.
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3 * Copyright (c) 2002 Fabrice Bellard |
0 | 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 |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
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. |
0 | 11 * |
1358
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
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 |
896
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Update licensing information: The FSF changed postal address.
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
3286 | 21 |
0 | 22 #include "avformat.h" |
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23 #include "mpegts.h" |
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24 #include "internal.h" |
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Initialize ssrc and base_timestamp using ff_random_get_seed()
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25 #include "libavutil/random_seed.h" |
0 | 26 |
4388 | 27 #include "rtpenc.h" |
1419 | 28 |
0 | 29 //#define DEBUG |
30 | |
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Use a symbolic name for the payload size of an RTCP Sender Report packet
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31 #define RTCP_SR_SIZE 28 |
0 | 32 |
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33 static int is_supported(enum CodecID id) |
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34 { |
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35 switch(id) { |
4814 | 36 case CODEC_ID_H263: |
37 case CODEC_ID_H263P: | |
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38 case CODEC_ID_H264: |
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39 case CODEC_ID_MPEG1VIDEO: |
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40 case CODEC_ID_MPEG2VIDEO: |
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41 case CODEC_ID_MPEG4: |
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42 case CODEC_ID_AAC: |
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43 case CODEC_ID_MP2: |
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44 case CODEC_ID_MP3: |
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45 case CODEC_ID_PCM_ALAW: |
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46 case CODEC_ID_PCM_MULAW: |
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47 case CODEC_ID_PCM_S8: |
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48 case CODEC_ID_PCM_S16BE: |
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49 case CODEC_ID_PCM_S16LE: |
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50 case CODEC_ID_PCM_U16BE: |
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51 case CODEC_ID_PCM_U16LE: |
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52 case CODEC_ID_PCM_U8: |
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53 case CODEC_ID_MPEG2TS: |
4836 | 54 case CODEC_ID_AMR_NB: |
55 case CODEC_ID_AMR_WB: | |
6349 | 56 case CODEC_ID_VORBIS: |
57 case CODEC_ID_THEORA: | |
4796
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58 return 1; |
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59 default: |
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60 return 0; |
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61 } |
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62 } |
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63 |
0 | 64 static int rtp_write_header(AVFormatContext *s1) |
65 { | |
4388 | 66 RTPMuxContext *s = s1->priv_data; |
5477 | 67 int max_packet_size, n; |
0 | 68 AVStream *st; |
69 | |
70 if (s1->nb_streams != 1) | |
71 return -1; | |
72 st = s1->streams[0]; | |
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73 if (!is_supported(st->codec->codec_id)) { |
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74 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); |
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75 |
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76 return -1; |
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77 } |
0 | 78 |
5477 | 79 s->payload_type = ff_rtp_get_payload_type(st->codec); |
80 if (s->payload_type < 0) | |
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81 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); |
0 | 82 |
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83 s->base_timestamp = av_get_random_seed(); |
0 | 84 s->timestamp = s->base_timestamp; |
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85 s->cur_timestamp = 0; |
6036
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86 s->ssrc = av_get_random_seed(); |
0 | 87 s->first_packet = 1; |
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88 s->first_rtcp_ntp_time = ff_ntp_time(); |
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89 if (s1->start_time_realtime) |
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90 /* Round the NTP time to whole milliseconds. */ |
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91 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + |
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92 NTP_OFFSET_US; |
0 | 93 |
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94 max_packet_size = url_fget_max_packet_size(s1->pb); |
0 | 95 if (max_packet_size <= 12) |
2274
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96 return AVERROR(EIO); |
4391 | 97 s->buf = av_malloc(max_packet_size); |
98 if (s->buf == NULL) { | |
99 return AVERROR(ENOMEM); | |
100 } | |
0 | 101 s->max_payload_size = max_packet_size - 12; |
102 | |
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103 s->max_frames_per_packet = 0; |
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104 if (s1->max_delay) { |
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105 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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106 if (st->codec->frame_size == 0) { |
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107 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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108 } else { |
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109 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN); |
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110 } |
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111 } |
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112 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { |
2587
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113 /* FIXME: We should round down here... */ |
3500 | 114 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
2587
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115 } |
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116 } |
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117 |
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118 av_set_pts_info(st, 32, 1, 90000); |
820
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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119 switch(st->codec->codec_id) { |
0 | 120 case CODEC_ID_MP2: |
232 | 121 case CODEC_ID_MP3: |
0 | 122 s->buf_ptr = s->buf + 4; |
123 break; | |
124 case CODEC_ID_MPEG1VIDEO: | |
2760 | 125 case CODEC_ID_MPEG2VIDEO: |
0 | 126 break; |
294
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127 case CODEC_ID_MPEG2TS: |
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128 n = s->max_payload_size / TS_PACKET_SIZE; |
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129 if (n < 1) |
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130 n = 1; |
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131 s->max_payload_size = n * TS_PACKET_SIZE; |
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132 s->buf_ptr = s->buf; |
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133 break; |
6136
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134 case CODEC_ID_H264: |
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135 /* check for H.264 MP4 syntax */ |
6139
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136 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { |
6136
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137 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; |
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138 } |
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139 break; |
6349 | 140 case CODEC_ID_VORBIS: |
141 case CODEC_ID_THEORA: | |
142 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15; | |
143 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); | |
144 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length | |
145 s->num_frames = 0; | |
146 goto defaultcase; | |
4836 | 147 case CODEC_ID_AMR_NB: |
148 case CODEC_ID_AMR_WB: | |
149 if (!s->max_frames_per_packet) | |
150 s->max_frames_per_packet = 12; | |
151 if (st->codec->codec_id == CODEC_ID_AMR_NB) | |
152 n = 31; | |
153 else | |
154 n = 61; | |
155 /* max_header_toc_size + the largest AMR payload must fit */ | |
156 if (1 + s->max_frames_per_packet + n > s->max_payload_size) { | |
157 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); | |
158 return -1; | |
159 } | |
160 if (st->codec->channels != 1) { | |
161 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); | |
162 return -1; | |
163 } | |
2550
e9c34ec665c6
Support for AAC streaming over RTP. Fragmentation is not implemented yet
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164 case CODEC_ID_AAC: |
4380
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Introduce a new num_frames field in RTPDemuxContext so that rtp_aac.c
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4291
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165 s->num_frames = 0; |
0 | 166 default: |
6349 | 167 defaultcase: |
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168 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
2539
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169 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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170 } |
0 | 171 s->buf_ptr = s->buf; |
172 break; | |
173 } | |
174 | |
175 return 0; | |
176 } | |
177 | |
178 /* send an rtcp sender report packet */ | |
65 | 179 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
0 | 180 { |
4388 | 181 RTPMuxContext *s = s1->priv_data; |
2539
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182 uint32_t rtp_ts; |
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183 |
3579 | 184 dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
2539
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Properly set RTP and NTP timestamps in RTCP SR packets
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diff
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185 |
2706
b1723b8da595
Do not send too many RTCP packets (according to RFC 3550, the minimum
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2705
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186 s->last_rtcp_ntp_time = ntp_time; |
3500 | 187 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
2539
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188 s1->streams[0]->time_base) + s->base_timestamp; |
2771
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189 put_byte(s1->pb, (RTP_VERSION << 6)); |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
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190 put_byte(s1->pb, 200); |
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Use dynamically allocated ByteIOContext in AVFormatContext
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191 put_be16(s1->pb, 6); /* length in words - 1 */ |
d52c718e83f9
Use dynamically allocated ByteIOContext in AVFormatContext
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192 put_be32(s1->pb, s->ssrc); |
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193 put_be32(s1->pb, ntp_time / 1000000); |
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194 put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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195 put_be32(s1->pb, rtp_ts); |
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196 put_be32(s1->pb, s->packet_count); |
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197 put_be32(s1->pb, s->octet_count); |
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198 put_flush_packet(s1->pb); |
0 | 199 } |
200 | |
201 /* send an rtp packet. sequence number is incremented, but the caller | |
202 must update the timestamp itself */ | |
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203 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
0 | 204 { |
4388 | 205 RTPMuxContext *s = s1->priv_data; |
0 | 206 |
3579 | 207 dprintf(s1, "rtp_send_data size=%d\n", len); |
0 | 208 |
209 /* build the RTP header */ | |
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210 put_byte(s1->pb, (RTP_VERSION << 6)); |
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211 put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
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212 put_be16(s1->pb, s->seq); |
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213 put_be32(s1->pb, s->timestamp); |
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214 put_be32(s1->pb, s->ssrc); |
885 | 215 |
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216 put_buffer(s1->pb, buf1, len); |
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217 put_flush_packet(s1->pb); |
885 | 218 |
0 | 219 s->seq++; |
220 s->octet_count += len; | |
221 s->packet_count++; | |
222 } | |
223 | |
224 /* send an integer number of samples and compute time stamp and fill | |
225 the rtp send buffer before sending. */ | |
226 static void rtp_send_samples(AVFormatContext *s1, | |
241 | 227 const uint8_t *buf1, int size, int sample_size) |
0 | 228 { |
4388 | 229 RTPMuxContext *s = s1->priv_data; |
0 | 230 int len, max_packet_size, n; |
231 | |
232 max_packet_size = (s->max_payload_size / sample_size) * sample_size; | |
233 /* not needed, but who nows */ | |
234 if ((size % sample_size) != 0) | |
235 av_abort(); | |
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236 n = 0; |
0 | 237 while (size > 0) { |
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238 s->buf_ptr = s->buf; |
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239 len = FFMIN(max_packet_size, size); |
0 | 240 |
241 /* copy data */ | |
242 memcpy(s->buf_ptr, buf1, len); | |
243 s->buf_ptr += len; | |
244 buf1 += len; | |
245 size -= len; | |
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246 s->timestamp = s->cur_timestamp + n / sample_size; |
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247 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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248 n += (s->buf_ptr - s->buf); |
0 | 249 } |
885 | 250 } |
0 | 251 |
252 static void rtp_send_mpegaudio(AVFormatContext *s1, | |
241 | 253 const uint8_t *buf1, int size) |
0 | 254 { |
4388 | 255 RTPMuxContext *s = s1->priv_data; |
0 | 256 int len, count, max_packet_size; |
257 | |
258 max_packet_size = s->max_payload_size; | |
259 | |
260 /* test if we must flush because not enough space */ | |
261 len = (s->buf_ptr - s->buf); | |
262 if ((len + size) > max_packet_size) { | |
263 if (len > 4) { | |
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264 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
0 | 265 s->buf_ptr = s->buf + 4; |
266 } | |
267 } | |
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268 if (s->buf_ptr == s->buf + 4) { |
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269 s->timestamp = s->cur_timestamp; |
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270 } |
0 | 271 |
272 /* add the packet */ | |
273 if (size > max_packet_size) { | |
274 /* big packet: fragment */ | |
275 count = 0; | |
276 while (size > 0) { | |
277 len = max_packet_size - 4; | |
278 if (len > size) | |
279 len = size; | |
280 /* build fragmented packet */ | |
281 s->buf[0] = 0; | |
282 s->buf[1] = 0; | |
283 s->buf[2] = count >> 8; | |
284 s->buf[3] = count; | |
285 memcpy(s->buf + 4, buf1, len); | |
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286 ff_rtp_send_data(s1, s->buf, len + 4, 0); |
0 | 287 size -= len; |
288 buf1 += len; | |
289 count += len; | |
290 } | |
291 } else { | |
292 if (s->buf_ptr == s->buf + 4) { | |
293 /* no fragmentation possible */ | |
294 s->buf[0] = 0; | |
295 s->buf[1] = 0; | |
296 s->buf[2] = 0; | |
297 s->buf[3] = 0; | |
298 } | |
299 memcpy(s->buf_ptr, buf1, size); | |
300 s->buf_ptr += size; | |
301 } | |
302 } | |
303 | |
304 static void rtp_send_raw(AVFormatContext *s1, | |
241 | 305 const uint8_t *buf1, int size) |
0 | 306 { |
4388 | 307 RTPMuxContext *s = s1->priv_data; |
0 | 308 int len, max_packet_size; |
309 | |
310 max_packet_size = s->max_payload_size; | |
311 | |
312 while (size > 0) { | |
313 len = max_packet_size; | |
314 if (len > size) | |
315 len = size; | |
316 | |
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317 s->timestamp = s->cur_timestamp; |
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318 ff_rtp_send_data(s1, buf1, len, (len == size)); |
0 | 319 |
320 buf1 += len; | |
321 size -= len; | |
322 } | |
323 } | |
324 | |
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325 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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326 static void rtp_send_mpegts_raw(AVFormatContext *s1, |
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327 const uint8_t *buf1, int size) |
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328 { |
4388 | 329 RTPMuxContext *s = s1->priv_data; |
294
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330 int len, out_len; |
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331 |
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332 while (size >= TS_PACKET_SIZE) { |
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333 len = s->max_payload_size - (s->buf_ptr - s->buf); |
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334 if (len > size) |
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335 len = size; |
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336 memcpy(s->buf_ptr, buf1, len); |
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337 buf1 += len; |
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338 size -= len; |
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339 s->buf_ptr += len; |
885 | 340 |
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341 out_len = s->buf_ptr - s->buf; |
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342 if (out_len >= s->max_payload_size) { |
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343 ff_rtp_send_data(s1, s->buf, out_len, 0); |
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344 s->buf_ptr = s->buf; |
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345 } |
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346 } |
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347 } |
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348 |
468 | 349 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 350 { |
4388 | 351 RTPMuxContext *s = s1->priv_data; |
0 | 352 AVStream *st = s1->streams[0]; |
353 int rtcp_bytes; | |
468 | 354 int size= pkt->size; |
885 | 355 |
3579 | 356 dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
0 | 357 |
885 | 358 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
0 | 359 RTCP_TX_RATIO_DEN; |
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360 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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361 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
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362 rtcp_send_sr(s1, ff_ntp_time()); |
0 | 363 s->last_octet_count = s->octet_count; |
364 s->first_packet = 0; | |
365 } | |
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366 s->cur_timestamp = s->base_timestamp + pkt->pts; |
0 | 367 |
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368 switch(st->codec->codec_id) { |
0 | 369 case CODEC_ID_PCM_MULAW: |
370 case CODEC_ID_PCM_ALAW: | |
371 case CODEC_ID_PCM_U8: | |
372 case CODEC_ID_PCM_S8: | |
5479 | 373 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels); |
0 | 374 break; |
375 case CODEC_ID_PCM_U16BE: | |
376 case CODEC_ID_PCM_U16LE: | |
377 case CODEC_ID_PCM_S16BE: | |
378 case CODEC_ID_PCM_S16LE: | |
5479 | 379 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels); |
0 | 380 break; |
381 case CODEC_ID_MP2: | |
232 | 382 case CODEC_ID_MP3: |
5479 | 383 rtp_send_mpegaudio(s1, pkt->data, size); |
0 | 384 break; |
385 case CODEC_ID_MPEG1VIDEO: | |
2760 | 386 case CODEC_ID_MPEG2VIDEO: |
5479 | 387 ff_rtp_send_mpegvideo(s1, pkt->data, size); |
0 | 388 break; |
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389 case CODEC_ID_AAC: |
5479 | 390 ff_rtp_send_aac(s1, pkt->data, size); |
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391 break; |
4836 | 392 case CODEC_ID_AMR_NB: |
393 case CODEC_ID_AMR_WB: | |
5479 | 394 ff_rtp_send_amr(s1, pkt->data, size); |
4836 | 395 break; |
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396 case CODEC_ID_MPEG2TS: |
5479 | 397 rtp_send_mpegts_raw(s1, pkt->data, size); |
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398 break; |
2960 | 399 case CODEC_ID_H264: |
5479 | 400 ff_rtp_send_h264(s1, pkt->data, size); |
2960 | 401 break; |
4814 | 402 case CODEC_ID_H263: |
403 case CODEC_ID_H263P: | |
5479 | 404 ff_rtp_send_h263(s1, pkt->data, size); |
4814 | 405 break; |
6349 | 406 case CODEC_ID_VORBIS: |
407 case CODEC_ID_THEORA: | |
408 ff_rtp_send_xiph(s1, pkt->data, size); | |
409 break; | |
0 | 410 default: |
411 /* better than nothing : send the codec raw data */ | |
5479 | 412 rtp_send_raw(s1, pkt->data, size); |
0 | 413 break; |
414 } | |
415 return 0; | |
416 } | |
417 | |
4391 | 418 static int rtp_write_trailer(AVFormatContext *s1) |
419 { | |
420 RTPMuxContext *s = s1->priv_data; | |
421 | |
422 av_freep(&s->buf); | |
423 | |
424 return 0; | |
425 } | |
426 | |
1167 | 427 AVOutputFormat rtp_muxer = { |
0 | 428 "rtp", |
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429 NULL_IF_CONFIG_SMALL("RTP output format"), |
0 | 430 NULL, |
431 NULL, | |
4388 | 432 sizeof(RTPMuxContext), |
0 | 433 CODEC_ID_PCM_MULAW, |
434 CODEC_ID_NONE, | |
435 rtp_write_header, | |
436 rtp_write_packet, | |
4391 | 437 rtp_write_trailer, |
0 | 438 }; |