June 2009 - FFmpeg-soc - lists.ffmpeg.org

[soc]: r4394 - seek_api/mpeg.c
by spyfeng 08 Jun '09

08 Jun '09

Author: spyfeng Date: Mon Jun 8 16:18:20 2009 New Revision: 4394 Log: use pts instead of e->timestamp for code more readable Modified: seek_api/mpeg.c Modified: seek_api/mpeg.c ============================================================================== --- seek_api/mpeg.c Mon Jun 8 16:16:45 2009 (r4393) +++ seek_api/mpeg.c Mon Jun 8 16:18:20 2009 (r4394) @@ -671,7 +671,7 @@ static int mpegps_read_seek1(struct AVFo url_fseek(s->pb, pos, SEEK_SET); av_log(s, AV_LOG_DEBUG, "the seek pos = %"PRId64", pts = %"PRId64", targe timestamp = %"PRId64"\n", pos, pts, ts); - if (e->timestamp == ts) { // find the target timestamp + if (pts == ts) { // find the target timestamp goto sucess; } else { // seek around to get the keyframe, then seek there if (find_keyframe(s, &pos,&pts,ts, flags) == 0) {

1 0

[soc]: r4393 - seek_api/mpeg.c
by spyfeng 08 Jun '09

08 Jun '09

Author: spyfeng Date: Mon Jun 8 16:16:45 2009 New Revision: 4393 Log: because we don't need to seek to the keyframe, in order to get the pts around the target timestamp, introduce AVSEEK_FLAG_ANY flag Modified: seek_api/mpeg.c Modified: seek_api/mpeg.c ============================================================================== --- seek_api/mpeg.c Mon Jun 8 16:13:46 2009 (r4392) +++ seek_api/mpeg.c Mon Jun 8 16:16:45 2009 (r4393) @@ -662,7 +662,7 @@ static int mpegps_read_seek1(struct AVFo if (st->index_entries) { AVIndexEntry *e; - index = av_index_search_timestamp(st, ts, flags | AVSEEK_FLAG_BACKWARD); + index = av_index_search_timestamp(st, ts, flags | AVSEEK_FLAG_ANY); if (index >= 0) { e = &st->index_entries[index];

1 0

[soc]: r4392 - seek_api/mpeg.c
by spyfeng 08 Jun '09

08 Jun '09

Author: spyfeng Date: Mon Jun 8 16:13:46 2009 New Revision: 4392 Log: return the pre-keyframe when it is more closer to target timestamp Modified: seek_api/mpeg.c Modified: seek_api/mpeg.c ============================================================================== --- seek_api/mpeg.c Mon Jun 8 15:23:40 2009 (r4391) +++ seek_api/mpeg.c Mon Jun 8 16:13:46 2009 (r4392) @@ -622,7 +622,7 @@ static int find_keyframe(AVFormatContext break; } - if (flags & AVSEEK_FLAG_BACKWARD) { + if (flags & AVSEEK_FLAG_BACKWARD && ((pre_pts + *pts)) >> 2 > target_ts) { *ret_pos = pre_pts; *pts = pre_pos; }

1 0

[soc]: r4389 - seek_api/mpeg.c
by spyfeng 07 Jun '09

07 Jun '09

Author: spyfeng Date: Sun Jun 7 20:19:38 2009 New Revision: 4389 Log: add new functions for seeking, still undertest. Modified: seek_api/mpeg.c Modified: seek_api/mpeg.c ============================================================================== --- seek_api/mpeg.c Sat Jun 6 11:30:26 2009 (r4388) +++ seek_api/mpeg.c Sun Jun 7 20:19:38 2009 (r4389) @@ -391,7 +391,7 @@ static int mpegps_read_pes_header(AVForm if(startcode == s->streams[i]->id && !url_is_streamed(s->pb) /* index useless on streams anyway */) { ff_reduce_index(s, i); - av_add_index_entry(s->streams[i], *ppos, dts, 0, 0, AVINDEX_KEYFRAME /* FIXME keyframe? */); + av_add_index_entry(s->streams[i], *ppos, dts, 0, 0, AVFMT_GENERIC_INDEX /* FIXME keyframe? */); } } } @@ -597,6 +597,116 @@ static int64_t mpegps_read_dts(AVFormatC return dts; } +/* 0 for sucess and -1 for error */ +static int find_keyframe(AVFormatContext *s, int64_t *ret_pos, int64_t *pts, int64_t target_ts, int flags) +{ + AVPacket pkt1, *pkt = &pkt1; + int64_t pre_pts, pre_pos, size; + + for (;;) { + if (av_read_frame(s, pkt) < 0) + return -1; + + size = pkt->size; + av_free_packet(pkt); + if (!(pkt->flags & PKT_FLAG_KEY)) + continue; + + *ret_pos += size; + *pts = pkt->pts; + + if (*pts < target_ts) { + pre_pts = *pts; + pre_pos = *ret_pos; + } else + break; + } + + if (flags & AVSEEK_FLAG_BACKWARD) { + *ret_pos = pre_pts; + *pts = pre_pos; + } + return 0; +} + +static int mpegps_read_seek1(struct AVFormatContext *s, int stream_index, + int64_t min_ts, int64_t ts, int64_t max_ts, int flags) +{ + AVStream* st; + int index; + int64_t pos, av_uninit(pos_min), av_uninit(pos_max), pos_limit; + int64_t ts_min, ts_max, ret_ts, pts; + + if (min_ts > ts || max_ts < ts) + return -1; + + if (stream_index < 0) { + stream_index= av_find_default_stream_index(s); + if(stream_index < 0) + return -1; + st = s->streams[stream_index]; + ts = av_rescale(ts, st->time_base.den, AV_TIME_BASE * (int64_t)st->time_base.num); + } else + st = s->streams[stream_index]; + + if (ts < 0) { + pts = ts = 0; + url_fseek(s->pb, 0, SEEK_SET); + goto sucess; + } + + if (st->discard >= AVDISCARD_ALL) { + av_log(s, AV_LOG_ERROR, "Not active stream!\n"); + return -1; + } + + if (st->index_entries) { + AVIndexEntry *e; + index = av_index_search_timestamp(st, ts, flags | AVSEEK_FLAG_BACKWARD); + + if (index >= 0) { + e = &st->index_entries[index]; + pos = e->pos; + pts = e->timestamp; + url_fseek(s->pb, pos, SEEK_SET); + av_log(s, AV_LOG_DEBUG, "the seek pos = %"PRId64", pts = %"PRId64", targe timestamp = %"PRId64"\n", pos, pts, ts); + + if (e->timestamp == ts) { // find the target timestamp + goto sucess; + } else { // seek around to get the keyframe, then seek there + if (find_keyframe(s, &pos,&pts,ts, flags) == 0) { + url_fseek(s->pb, pos, SEEK_SET); + goto sucess; + } else { + av_update_cur_dts(s, st, pts); + return -1; + } + } + } + } + // search the scr use binary search + ts_max= + ts_min= AV_NOPTS_VALUE; + pos_limit= -1; + pos= av_gen_search(s, stream_index, ts, pos_min, pos_max, pos_limit, ts_min, ts_max, flags | AVSEEK_FLAG_BACKWARD, &ret_ts, mpegps_read_dts); + av_log(s, AV_LOG_DEBUG, "the seek pos = %"PRId64", ret_ts = %"PRId64"\n", pos, ret_ts); + if(pos<0) + return -1; + + url_fseek(s->pb, pos, SEEK_SET); + if (find_keyframe(s, &pos, &ret_ts, ts, flags) == 0) { + url_fseek(s->pb, pos, SEEK_SET); + pts = ret_ts; + goto sucess; + } else { + av_update_cur_dts(s, st, ret_ts); + return -1; + } +sucess: + av_update_cur_dts(s, st, pts); + return 0; +} + static int mpegps_read_seek(struct AVFormatContext *s, int stream_index, int64_t min_ts, int64_t ts, int64_t max_ts, int flags) { @@ -683,5 +793,5 @@ AVInputFormat mpegps_demuxer = { NULL, //mpegps_read_seek, mpegps_read_dts, .flags = AVFMT_SHOW_IDS|AVFMT_TS_DISCONT, - .read_seek2 = mpegps_read_seek, + .read_seek2 = mpegps_read_seek1, };

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[soc]: r4387 - als/ffmpeg-patches/hack4als.patch
by thilo.borgmann 06 Jun '09

06 Jun '09

Author: thilo.borgmann Date: Fri Jun 5 13:06:54 2009 New Revision: 4387 Log: Adds a patch for the FFmpeg trunk to enable ALS demuxing. This patch has been written by Jai Menon but has not passed review yet. Refer to thread "[PATCH] als in mp4" for details. This patch is not meant to be applied as-is in the end but enables me to continue work on the ALS decoder for the time being. If necessary, I will continue to work on this patch to pass the review process if it has not been done as soon as the ALS decoder is to be applied to the codebase. Added: als/ffmpeg-patches/hack4als.patch Added: als/ffmpeg-patches/hack4als.patch ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ als/ffmpeg-patches/hack4als.patch Fri Jun 5 13:06:54 2009 (r4387) @@ -0,0 +1,65 @@ +Index: libavcodec/mpeg4audio.c +=================================================================== +--- libavcodec/mpeg4audio.c (Revision 19089) ++++ libavcodec/mpeg4audio.c (Arbeitskopie) +@@ -57,6 +57,8 @@ + c->object_type = get_object_type(&gb); + c->sample_rate = get_sample_rate(&gb, &c->sampling_index); + c->chan_config = get_bits(&gb, 4); ++ c->absolute_channels = 0; ++ c->bits_per_sample = 0; + c->sbr = -1; + if (c->object_type == 5) { + c->ext_object_type = c->object_type; +@@ -84,6 +86,12 @@ + get_bits1(&gb); // skip 1 bit + } + } ++ if (c->object_type == AOT_ALS) { ++ c->absolute_channels = 1; ++ c->sample_rate = AV_RB32(buf + 10); ++ c->chan_config = AV_RB16(buf + 18) + 1; ++ c->bits_per_sample = (((buf[20] >> 2) & 0x07) + 1) << 3; ++ } + return specific_config_bitindex; + } + +Index: libavcodec/mpeg4audio.h +=================================================================== +--- libavcodec/mpeg4audio.h (Revision 19089) ++++ libavcodec/mpeg4audio.h (Arbeitskopie) +@@ -36,6 +36,8 @@ + int ext_sampling_index; + int ext_sample_rate; + int ext_chan_config; ++ int bits_per_sample; ++ int absolute_channels; + } MPEG4AudioConfig; + + extern const int ff_mpeg4audio_sample_rates[16]; +Index: libavformat/mov.c +=================================================================== +--- libavformat/mov.c (Revision 19089) ++++ libavformat/mov.c (Arbeitskopie) +@@ -425,13 +425,18 @@ + MPEG4AudioConfig cfg; + ff_mpeg4audio_get_config(&cfg, st->codec->extradata, + st->codec->extradata_size); +- if (cfg.chan_config > 7) +- return -1; +- st->codec->channels = ff_mpeg4audio_channels[cfg.chan_config]; ++ if (!cfg.absolute_channels) { ++ if (cfg.chan_config > 7) ++ return -1; ++ st->codec->channels = ff_mpeg4audio_channels[cfg.chan_config]; ++ } else st->codec->channels = cfg.chan_config; ++ + if (cfg.object_type == 29 && cfg.sampling_index < 3) // old mp3on4 + st->codec->sample_rate = ff_mpa_freq_tab[cfg.sampling_index]; + else + st->codec->sample_rate = cfg.sample_rate; // ext sample rate ? ++ if (cfg.bits_per_sample) ++ st->codec->bits_per_coded_sample = cfg.bits_per_sample; + dprintf(c->fc, "mp4a config channels %d obj %d ext obj %d " + "sample rate %d ext sample rate %d\n", st->codec->channels, + cfg.object_type, cfg.ext_object_type,

4 5

[soc]: r4388 - rtmp/TODO
by kostya 06 Jun '09

06 Jun '09

Author: kostya Date: Sat Jun 6 11:30:26 2009 New Revision: 4388 Log: Extend TODO list for RTMP Modified: rtmp/TODO Modified: rtmp/TODO ============================================================================== --- rtmp/TODO Fri Jun 5 13:06:54 2009 (r4387) +++ rtmp/TODO Sat Jun 6 11:30:26 2009 (r4388) @@ -1,3 +1,4 @@ +Implement SHA-256 digest needed for proper handshake Proper handshake Proper "connect" call (additional parameters and filename parsing) Factorize a lot of FLV demuxer code

1 0

[soc]: r4345 - in wmapro: wma3.h wma3dec.c
by faust3 05 Jun '09

05 Jun '09

Author: faust3 Date: Sat May 30 21:22:28 2009 New Revision: 4345 Log: decode decorrelation matrix in wma_decode_decorrelation_matrix and reduced the channel group context size Modified: wmapro/wma3.h wmapro/wma3dec.c Modified: wmapro/wma3.h ============================================================================== --- wmapro/wma3.h Sat May 30 19:58:50 2009 (r4344) +++ wmapro/wma3.h Sat May 30 21:22:28 2009 (r4345) @@ -74,12 +74,11 @@ typedef struct { * @brief channel group for channel transformations */ typedef struct { - int num_channels; - int transform; ///< controls the type of the transform - char transform_band[MAX_BANDS]; ///< controls if the transform is enabled for a certain band - char rotation_offset[MAX_CHANNELS * MAX_CHANNELS]; - float decorrelation_matrix[MAX_CHANNELS*MAX_CHANNELS]; - char use_channel[MAX_CHANNELS]; + char num_channels; ///< number of channels in the group + char transform; ///< controls the type of the transform + char transform_band[MAX_BANDS]; ///< controls if the transform is enabled for a certain band + float decorrelation_matrix[MAX_CHANNELS*MAX_CHANNELS]; ///< decorrelation matrix + char use_channel[MAX_CHANNELS]; ///< channels that are part of the group } WMA3ChannelGroup; /** Modified: wmapro/wma3dec.c ============================================================================== --- wmapro/wma3dec.c Sat May 30 19:58:50 2009 (r4344) +++ wmapro/wma3dec.c Sat May 30 21:22:28 2009 (r4345) @@ -613,12 +613,24 @@ static int wma_decode_tilehdr(WMA3Decode /** *@brief Calculate a decorrelation matrix from the bitstream parameters. + *@param s codec context *@param chgroup channel group for which the matrix needs to be calculated */ -static void wma_calc_decorrelation_matrix(WMA3ChannelGroup* chgroup) +static void wma_decode_decorrelation_matrix(WMA3DecodeContext* s, WMA3ChannelGroup* chgroup) { int i; int offset = 0; + char rotation_offset[MAX_CHANNELS * MAX_CHANNELS]; + + for(i=0;i<chgroup->num_channels * (chgroup->num_channels - 1) >> 1;i++) + rotation_offset[i] = get_bits(&s->gb,6); + + for(i=0;i<chgroup->num_channels;i++){ + if(get_bits1(&s->gb)) + chgroup->decorrelation_matrix[chgroup->num_channels * i + i]= 1.0; + else + chgroup->decorrelation_matrix[chgroup->num_channels * i + i]= -1.0; + } for(i=1;i<chgroup->num_channels;i++){ int x; @@ -631,7 +643,7 @@ static void wma_calc_decorrelation_matri for(y=0;y < i + 1 ; y++){ float v1 = tmp1[y]; float v2 = tmp2[y]; - int n = chgroup->rotation_offset[offset + x]; + int n = rotation_offset[offset + x]; float cosv = sin(n*M_PI / 64.0); // FIXME: use one table for this float sinv = -cos(n*M_PI / 64.0); @@ -714,22 +726,13 @@ static int wma_decode_channel_transform( } }else if(chgroup->num_channels > 2){ if(get_bits1(&s->gb)){ - if(get_bits1(&s->gb)){ - int n_offset = chgroup->num_channels * (chgroup->num_channels - 1) >> 1; - int i; - chgroup->transform = 2; - for(i=0;i<n_offset;i++){ - chgroup->rotation_offset[i] = get_bits(&s->gb,6); - } - for(i=0;i<chgroup->num_channels;i++) - chgroup->decorrelation_matrix[chgroup->num_channels * i + i] = get_bits1(&s->gb)?1.0:-1.0; - wma_calc_decorrelation_matrix(chgroup); - }else{ - chgroup->transform = 2; + chgroup->transform = 2; + if(get_bits1(&s->gb)) + wma_decode_decorrelation_matrix(s, chgroup); + else memcpy(chgroup->decorrelation_matrix, s->def_decorrelation_mat[chgroup->num_channels], chgroup->num_channels * chgroup->num_channels * sizeof(float)); - } } }

3 2

[soc] [propchange]: r4345 - svn:log
by faust3 05 Jun '09

05 Jun '09

Author: faust3 Revision: 4345 Property Name: svn:log Action: modified Property diff: --- old property value +++ new property value @@ -1 +1,2 @@ -decode decorrelation matrix in wma_decode_decorrelation_matrix and reduced the channel group context size \ No newline at end of file +decode decorrelation matrix in wma_decode_decorrelation_matrix +reduce channel group context size

1 0

[soc] [propchange]: r4333 - svn:log
by faust3 05 Jun '09

05 Jun '09

Author: faust3 Revision: 4333 Property Name: svn:log Action: modified Property diff: --- old property value +++ new property value @@ -1 +1 @@ -added av_cold wma_dump_context() \ No newline at end of file +added av_cold to wma_dump_context()

1 0

[soc]: r4382 - in rtmp: TODO checkout.sh rtmp.h rtmp.patch rtmpdec.c rtmppkt.c rtmppkt.h rtmpproto.c
by kostya 04 Jun '09

04 Jun '09

Author: kostya Date: Thu Jun 4 08:57:37 2009 New Revision: 4382 Log: Some initial work on RTMP client. Added: rtmp/TODO rtmp/checkout.sh (contents, props changed) rtmp/rtmp.h rtmp/rtmp.patch rtmp/rtmpdec.c rtmp/rtmppkt.c rtmp/rtmppkt.h rtmp/rtmpproto.c Added: rtmp/TODO ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/TODO Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,4 @@ +Proper handshake +Proper "connect" call (additional parameters and filename parsing) +Factorize a lot of FLV demuxer code +Set correct timestamps Added: rtmp/checkout.sh ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/checkout.sh Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,14 @@ +#!/bin/sh + +echo "Checking out FFmpeg SVN trunk code..." +svn co svn://svn.ffmpeg.org/ffmpeg/trunk ffmpeg && \ +cd ffmpeg + +echo "Patching build system" +patch -p1 < ../rtmp.patch && \ +cd libavformat + +echo "Copying source code to libavformat dir" +cp ../../rtmp*.[ch] . + +echo "Done! Now enter the ffmpeg dir, configure and make FFmpeg with some RTMP support" Added: rtmp/rtmp.h ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmp.h Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,39 @@ +/* + * RTMP definitions + * Copyright (c) 2009 Kostya Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVFORMAT_RTMP_H +#define AVFORMAT_RTMP_H +#include "avformat.h" + +/** RTMP default port */ +#define RTMP_DEFAULT_PORT 1935 + +/** RTMP handshake data size */ +#define RTMP_HANDSHAKE_PACKET_SIZE 1536 + +#define RTMP_CLIENT_VER1 9 +#define RTMP_CLIENT_VER2 0 +#define RTMP_CLIENT_VER3 124 +#define RTMP_CLIENT_VER4 2 + +#define RTMP_CLIENT_VER_STR "LNX" #RTMP_CLIENT_VER1 "," #RTMP_CLIENT_VER2 + +#endif /* AVFORMAT_RTMP_H */ Added: rtmp/rtmp.patch ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmp.patch Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,40 @@ +Index: libavformat/Makefile +=================================================================== +--- libavformat/Makefile (revision 18770) ++++ libavformat/Makefile (working copy) +@@ -187,6 +187,7 @@ + OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o + OBJS-$(CONFIG_ROQ_MUXER) += raw.o + OBJS-$(CONFIG_RPL_DEMUXER) += rpl.o ++OBJS-$(CONFIG_RTMP_DEMUXER) += rtmpdec.o rtmppkt.o + OBJS-$(CONFIG_RTP_MUXER) += rtp.o \ + rtp_aac.o \ + rtp_amr.o \ +@@ -243,6 +244,7 @@ + OBJS-$(CONFIG_GOPHER_PROTOCOL) += gopher.o + OBJS-$(CONFIG_HTTP_PROTOCOL) += http.o + OBJS-$(CONFIG_PIPE_PROTOCOL) += file.o ++OBJS-$(CONFIG_RTMP_PROTOCOL) += rtmpproto.o + OBJS-$(CONFIG_RTP_PROTOCOL) += rtpproto.o + OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o + OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o +Index: libavformat/allformats.c +=================================================================== +--- libavformat/allformats.c (revision 18770) ++++ libavformat/allformats.c (working copy) +@@ -164,6 +164,7 @@ + REGISTER_MUXDEMUX (RM, rm); + REGISTER_MUXDEMUX (ROQ, roq); + REGISTER_DEMUXER (RPL, rpl); ++ REGISTER_DEMUXER (RTMP, rtmp); + REGISTER_MUXER (RTP, rtp); + REGISTER_DEMUXER (RTSP, rtsp); + REGISTER_DEMUXER (SDP, sdp); +@@ -207,6 +208,7 @@ + REGISTER_PROTOCOL (GOPHER, gopher); + REGISTER_PROTOCOL (HTTP, http); + REGISTER_PROTOCOL (PIPE, pipe); ++ REGISTER_PROTOCOL (RTMP, rtmp); + REGISTER_PROTOCOL (RTP, rtp); + REGISTER_PROTOCOL (TCP, tcp); + REGISTER_PROTOCOL (UDP, udp); Added: rtmp/rtmpdec.c ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmpdec.c Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,350 @@ +/* + * RTMP input format + * Copyright (c) 2009 Kostya Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/* needed for gethostname() */ +#define _XOPEN_SOURCE 600 + +#include "libavcodec/bytestream.h" +#include "libavutil/avstring.h" +#include "avformat.h" + +#include <unistd.h> +#include <sys/time.h> +#include "network.h" + +#include "flv.h" +#include "rtmp.h" +#include "rtmppkt.h" + +#define DEBUG 1 + +typedef struct RTMPState { + URLContext *rtmp_hd; + RTMPPacketHistory rhist, whist; +} RTMPState; + +static void gen_connect(AVFormatContext *s, RTMPState *rt, const char *name) +{ + uint8_t buf[65536], *p = buf; + double num = 1.0; + int size; + + bytestream_put_byte(&p, 0x3); + bytestream_put_be24(&p, 1); + bytestream_put_be24(&p, 0); + bytestream_put_byte(&p, RTMP_PT_INVOKE); + bytestream_put_le32(&p, 0); + + rtmp_amf_write_tag(&p, AMF_STRING, "connect"); + rtmp_amf_write_tag(&p, AMF_NUMBER, &num); + rtmp_amf_write_tag(&p, AMF_OBJECT, NULL); + rtmp_amf_write_tag(&p, AMF_STRING_IN_OBJECT, "tcUrl"); + rtmp_amf_write_tag(&p, AMF_STRING, name); + rtmp_amf_write_tag(&p, AMF_STRING_IN_OBJECT, "app"); + rtmp_amf_write_tag(&p, AMF_STRING, "app"); + rtmp_amf_write_tag(&p, AMF_STRING_IN_OBJECT, "flashVer"); + rtmp_amf_write_tag(&p, AMF_STRING, "LNX 6,0,82,0"); + rtmp_amf_write_tag(&p, AMF_STRING_IN_OBJECT, "playpath"); + rtmp_amf_write_tag(&p, AMF_STRING, "1.mp4"); + rtmp_amf_write_tag(&p, AMF_OBJECT_END, NULL); + + size = p - buf; + p = buf + 4; + bytestream_put_be24(&p, size - 12); + + url_write(rt->rtmp_hd, buf, size); +} + +static void gen_play(AVFormatContext *s, RTMPState *rt) +{ + RTMPPacket pkt; + uint8_t *p; + double num; + + rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 25); + + num = 3.0; + p = pkt.data; + rtmp_amf_write_tag(&p, AMF_STRING, "createStream"); + rtmp_amf_write_tag(&p, AMF_NUMBER, &num); + rtmp_amf_write_tag(&p, AMF_NULL, NULL); + + rtmp_packet_write(s, rt->rtmp_hd, &pkt, &rt->whist); + rtmp_packet_destroy(&pkt); + + rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 17); + + num = 0.0; + p = pkt.data; + rtmp_amf_write_tag(&p, AMF_STRING, "play"); + rtmp_amf_write_tag(&p, AMF_NUMBER, &num); + rtmp_amf_write_tag(&p, AMF_NULL, NULL); + + rtmp_packet_write(s, rt->rtmp_hd, &pkt, &rt->whist); + rtmp_packet_destroy(&pkt); +} + +static int rtmp_handshake(AVFormatContext *s, RTMPState *rt) +{ + uint8_t buf [1+RTMP_HANDSHAKE_PACKET_SIZE]; + uint8_t buf2[1+RTMP_HANDSHAKE_PACKET_SIZE*2]; + int i; + + av_log(s, AV_LOG_DEBUG, "Handshaking...\n"); + + // generate handshake packet - 1536 bytes of pseudorandom data + buf[0] = 3; //unencrypted data + memset(buf+1, 0, RTMP_HANDSHAKE_PACKET_SIZE); + //write client "version" + buf[5] = 9; + buf[5] = 0; + buf[7] = 124; + buf[8] = 2; + + url_write(rt->rtmp_hd, buf, RTMP_HANDSHAKE_PACKET_SIZE + 1); + i = url_read_complete(rt->rtmp_hd, buf2, RTMP_HANDSHAKE_PACKET_SIZE*2 + 1); + if (i != 1 + RTMP_HANDSHAKE_PACKET_SIZE*2 || memcmp(buf + 1, buf2 + 1 + RTMP_HANDSHAKE_PACKET_SIZE, RTMP_HANDSHAKE_PACKET_SIZE)) { + av_log(s, AV_LOG_ERROR, "RTMP handshake failed\n"); + return -1; + } + // write reply back to server + url_write(rt->rtmp_hd, buf2 + 1, RTMP_HANDSHAKE_PACKET_SIZE); + + return 0; +} + +static void rtmp_init_hist(RTMPPacketHistory *hist) +{ + int i; + + for (i = 0; i < RTMP_CHANNELS; i++) { + hist->chunk_size[i] = (i == RTMP_AUDIO_CHANNEL) ? 64 : 128; + } +} + +static int rtmp_probe(AVProbeData *p) +{ + if (av_strstart(p->filename, "rtmp:", NULL)) + return AVPROBE_SCORE_MAX; + return 0; +} + +static int rtmp_read_header(AVFormatContext *s, + AVFormatParameters *ap) +{ + RTMPState *rt = s->priv_data; + char hostname[256]; + int port; + uint8_t buf[65536]; + AVStream *st; + + url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &port, + NULL, 0, s->filename); + + rtmp_init_hist(&rt->rhist); + rtmp_init_hist(&rt->whist); + if(port == -1) + port = RTMP_DEFAULT_PORT; + snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port); + url_open(&rt->rtmp_hd, buf, URL_RDWR); + if (rtmp_handshake(s, rt)) + return -1; + gen_connect(s, rt, s->filename); + gen_play(s, rt); + + st = av_new_stream(s, 0); + if (!st) + return -1; + st->codec->codec_type = CODEC_TYPE_VIDEO; + av_set_pts_info(st, 32, 1, 1000); /* 32 bit pts in ms */ + + st = av_new_stream(s, 1); + if (!st) + return -1; + st->codec->codec_type = CODEC_TYPE_AUDIO; + av_set_pts_info(st, 32, 1, 1000); /* 32 bit pts in ms */ + + return 0; +} + +static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_codecid) { + AVCodecContext *acodec = astream->codec; + switch(flv_codecid) { + //no distinction between S16 and S8 PCM codec flags + case FLV_CODECID_PCM: + acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : +#ifdef WORDS_BIGENDIAN + CODEC_ID_PCM_S16BE; +#else + CODEC_ID_PCM_S16LE; +#endif + break; + case FLV_CODECID_PCM_LE: + acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break; + case FLV_CODECID_AAC : acodec->codec_id = CODEC_ID_AAC; break; + case FLV_CODECID_ADPCM: acodec->codec_id = CODEC_ID_ADPCM_SWF; break; + case FLV_CODECID_SPEEX: + acodec->codec_id = CODEC_ID_SPEEX; + acodec->sample_rate = 16000; + break; + case FLV_CODECID_MP3 : acodec->codec_id = CODEC_ID_MP3 ; astream->need_parsing = AVSTREAM_PARSE_FULL; break; + case FLV_CODECID_NELLYMOSER_8KHZ_MONO: + acodec->sample_rate = 8000; //in case metadata does not otherwise declare samplerate + case FLV_CODECID_NELLYMOSER: + acodec->codec_id = CODEC_ID_NELLYMOSER; + break; + default: + av_log(s, AV_LOG_INFO, "Unsupported audio codec (%x)\n", flv_codecid >> FLV_AUDIO_CODECID_OFFSET); + acodec->codec_tag = flv_codecid >> FLV_AUDIO_CODECID_OFFSET; + } +} + +static int flv_set_video_codec(AVFormatContext *s, AVStream *vstream, int flv_codecid) { + AVCodecContext *vcodec = vstream->codec; + switch(flv_codecid) { + case FLV_CODECID_H263 : vcodec->codec_id = CODEC_ID_FLV1 ; break; + case FLV_CODECID_SCREEN: vcodec->codec_id = CODEC_ID_FLASHSV; break; + case FLV_CODECID_VP6 : vcodec->codec_id = CODEC_ID_VP6F ; + case FLV_CODECID_VP6A : + if(flv_codecid == FLV_CODECID_VP6A) + vcodec->codec_id = CODEC_ID_VP6A; + if(vcodec->extradata_size != 1) { + vcodec->extradata_size = 1; + vcodec->extradata = av_malloc(1); + } + vcodec->extradata[0] = get_byte(s->pb); + return 1; // 1 byte body size adjustment for flv_read_packet() + case FLV_CODECID_H264: + vcodec->codec_id = CODEC_ID_H264; + return 3; // not 4, reading packet type will consume one byte + default: + av_log(s, AV_LOG_INFO, "Unsupported video codec (%x)\n", flv_codecid); + vcodec->codec_tag = flv_codecid; + } + + return 0; +} + +static int rtmp_read_packet(AVFormatContext *s, AVPacket *pkt) +{ + RTMPState *rt = s->priv_data; + AVStream *st = NULL; + struct timespec ts; + int i, ret; + + ts.tv_sec = 0; + ts.tv_nsec = 500000000; + for (;;) { + RTMPPacket rpkt; + int i; + if ((ret = rtmp_packet_read(s, rt->rtmp_hd, &rpkt, &rt->rhist)) != 0) { + if (ret > 0) { + nanosleep(&ts, NULL); + continue; + } else { + return AVERROR(EIO); + } + } + +#ifdef DEBUG +if(rpkt.type != RTMP_PT_VIDEO && rpkt.type != RTMP_PT_AUDIO) rtmp_packet_inspect(s, &rpkt); +//if(rpkt.data_size)for(i = 0; i < rpkt.data_size;i++)av_log(NULL,0," %02X",rpkt.data[i]);av_log(NULL,0,"\n"); +#endif + + if (rpkt.type == RTMP_PT_CHUNK_SIZE) + for(i=0;i<RTMP_CHANNELS;i++)rt->rhist.chunk_size[i/*rpkt.stream_id*/] = AV_RB32(rpkt.data); + if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO) { + int is_audio = rpkt.type == RTMP_PT_AUDIO; + int flags; + int off = 1; + + flags = rpkt.data[0]; + for (i = 0; i < s->nb_streams; i++) { + st = s->streams[i]; + if (st->id == is_audio) + break; + } + if (is_audio && (!st->codec->channels || !st->codec->sample_rate || !st->codec->bits_per_coded_sample)) { + st->codec->channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1; + st->codec->sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3); + st->codec->bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8; + } + if (!st->codec->codec_id) { + if (is_audio) + flv_set_audio_codec(s, st, flags & FLV_AUDIO_CODECID_MASK); + else + flv_set_video_codec(s, st, flags & FLV_VIDEO_CODECID_MASK); + } + if (st->codec->codec_id == CODEC_ID_AAC || + st->codec->codec_id == CODEC_ID_H264) { + int type = rpkt.data[off++]; + + if (st->codec->codec_id == CODEC_ID_H264) { + off += 3; + } + if (!type) { + av_free(st->codec->extradata); + st->codec->extradata = av_mallocz(rpkt.data_size - off + FF_INPUT_BUFFER_PADDING_SIZE); + if (!st->codec->extradata) { + av_log(s, AV_LOG_ERROR, "Cannot allocate extradata\n"); + return AVERROR(ENOMEM); + } + st->codec->extradata_size = rpkt.data_size - off; + memcpy(st->codec->extradata, rpkt.data + off, rpkt.data_size - off); + rtmp_packet_destroy(&rpkt); + continue; + } + } + if (off < rpkt.data_size) { + if (av_new_packet(pkt, rpkt.data_size - off) < 0) { + rtmp_packet_destroy(&rpkt); + return -1; + } + memcpy(pkt->data, rpkt.data + off, rpkt.data_size - off); + pkt->stream_index = st->index; + //pkt->pts = rpkt.timestamp; + rtmp_packet_destroy(&rpkt); + + return pkt->size; + } + } + rtmp_packet_destroy(&rpkt); + } + return AVERROR(EIO); +} + +static int rtmp_read_close(AVFormatContext *s) +{ + RTMPState *rt = s->priv_data; + + url_close(rt->rtmp_hd); + return 0; +} + +AVInputFormat rtmp_demuxer = { + "rtmp", + NULL_IF_CONFIG_SMALL("RTMP"), + sizeof(RTMPState), + rtmp_probe, + rtmp_read_header, + rtmp_read_packet, + rtmp_read_close, +}; Added: rtmp/rtmppkt.c ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmppkt.c Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,267 @@ +/* + * RTMP input format + * Copyright (c) 2009 Kostya Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/* needed for gethostname() */ +#define _XOPEN_SOURCE 600 + +#include "libavcodec/bytestream.h" +#include "libavutil/avstring.h" +#include "avformat.h" + +#include "rtmppkt.h" + +static const int8_t amf_tag_sizes[17] = { + 8, 1, 0, 0, 0, 0, 0, 0, 0, 0, 4, 0, 4, 0, 0, 0, 0 +}; + +int rtmp_amf_tag_size(int type, const void *data) +{ + if (type == AMF_STRING_IN_OBJECT || type == AMF_STRING) + return strlen(data) + 2; + if (type > AMF_TYPED_OBJECT) + return -1; + if (type == AMF_LONG_STRING) + return strlen(data) + 4; + return amf_tag_sizes[type]; +} + +void rtmp_amf_write_tag(uint8_t **dst, AMFType type, const void *data) +{ + if (type != AMF_OBJECT_END && type != AMF_STRING_IN_OBJECT) + bytestream_put_byte(dst, type); + switch(type){ + case AMF_NUMBER: + bytestream_put_be64(dst, av_dbl2int(*(const double*)data)); + break; + case AMF_BOOLEAN: + bytestream_put_byte(dst, *(const uint8_t*)data); + break; + case AMF_STRING: + case AMF_STRING_IN_OBJECT: + bytestream_put_be16(dst, strlen(data)); + bytestream_put_buffer(dst, data, strlen(data)); + break; + case AMF_OBJECT_END: + bytestream_put_be24(dst, AMF_OBJECT_END); + break; + case AMF_STRICT_ARRAY: + bytestream_put_be32(dst, *(const uint32_t*)data); + break; + } +} + +int rtmp_packet_read(AVFormatContext *ctx, URLContext *h, RTMPPacket *p, + RTMPPacketHistory *hist) +{ + uint8_t hdr, t, buf[16]; + int stream_id, timestamp, data_size, offset = 0; + uint8_t type; + + if (url_read(h, &hdr, 1) != 1) { + av_log(ctx, AV_LOG_ERROR, "Cannot read packet header\n"); + return -1; + } + stream_id = hdr & 0x3F; + + hdr >>= 6; + if (hdr == RTMP_PS_ONEBYTE) { + av_log(ctx, AV_LOG_ERROR, "Onebyte header!\n\n\n"); + //todo + return -1; + } else { + if (url_read_complete(h, buf, 3) != 3) { + av_log(ctx, AV_LOG_ERROR, "reading timestamp failed\n"); + return -1; + } + timestamp = AV_RB24(buf); + if (hdr != RTMP_PS_FOURBYTES) { + if (url_read_complete(h, buf, 3) != 3) { + av_log(ctx, AV_LOG_ERROR, "reading size failed\n"); + return -1; + } + data_size = AV_RB24(buf); + if (url_read_complete(h, &type, 1) != 1) { + av_log(ctx, AV_LOG_ERROR, "reading type failed\n"); + return -1; + } + if (hdr == RTMP_PS_TWELVEBYTES) + if (url_read(h, buf, 4) != 4) { + av_log(ctx, AV_LOG_ERROR, "reading timestamp2 failed\n"); + return -1; + } + } else { + av_log(ctx, AV_LOG_ERROR, "Fourbytes!\n\n\n"); + //todo + return -1; + } + } + rtmp_packet_create(p, stream_id, type, type, data_size); + while (data_size > 0) { + int toread = FFMIN(data_size, hist->chunk_size[stream_id]); + int r; + if ((r = url_read_complete(h, p->data + offset, toread)) != toread) { + av_log(ctx, AV_LOG_ERROR, "Need %d read %d\n", toread, r); + rtmp_packet_destroy(p); + return -1; + } + data_size -= hist->chunk_size[stream_id]; + offset += hist->chunk_size[stream_id]; + if (data_size > 0) { + url_read_complete(h, &t, 1); //marker + } + } + return 0; +} + +int rtmp_packet_write(AVFormatContext *ctx, URLContext *h, RTMPPacket *pkt, + RTMPPacketHistory *hist) +{ + uint8_t pkt_hdr[16], *p = pkt_hdr; + int mode = RTMP_PS_TWELVEBYTES; + + if (pkt->type != RTMP_PT_INVOKE) + mode = RTMP_PS_EIGHTBYTES; + bytestream_put_byte(&p, pkt->stream_id | (mode << 6)); + if (mode != RTMP_PS_ONEBYTE) { + bytestream_put_be24(&p, pkt->timestamp); + if (mode != RTMP_PS_FOURBYTES) { + bytestream_put_be24(&p, pkt->data_size); + bytestream_put_byte(&p, pkt->type); + if (mode == RTMP_PS_TWELVEBYTES) + bytestream_put_le32(&p, 0); //FIXME put real data here + } + } + url_write(h, pkt_hdr, p-pkt_hdr); + url_write(h, pkt->data, pkt->data_size); + return 0; +} + +int rtmp_packet_create(RTMPPacket *pkt, int stream_id, RTMPPacketType type, + int timestamp, int size) +{ + pkt->data = av_malloc(size); + if (!pkt->data) + return -1; + pkt->data_size = size; + pkt->stream_id = stream_id; + pkt->type = type; + pkt->timestamp = timestamp; + + return 0; +} + +void rtmp_packet_destroy(RTMPPacket *pkt) +{ + if (!pkt) + return; + av_freep(&pkt->data); + pkt->data_size = 0; +} + + +static void parse_amf(const uint8_t *data, int size) +{ + const uint8_t *ptr = data, *end = data+size; + int object = 0,array=0,i,len; + char buf[65536]; + while(ptr < end){ + if(object){ + len=bytestream_get_be16(&ptr); + if(len){ + for(i=0;i<object;i++)av_log(NULL,0," "); + bytestream_get_buffer(&ptr, buf, len); + buf[len] = 0; + av_log(NULL,0,"%s: ",buf); + } + } + switch (*ptr++) { + case AMF_NUMBER: + { + double d; + d = av_int2dbl(bytestream_get_be64(&ptr)); + av_log(NULL,0,"Number %g\n",d); + } + break; + case AMF_BOOLEAN: + av_log(NULL,0,"Boolean %d\n",*ptr++); + break; + case AMF_STRING: + len=bytestream_get_be16(&ptr); + bytestream_get_buffer(&ptr, buf, len); + buf[len] = 0; + av_log(NULL,0,"String '%s'\n",buf); + break; + case AMF_NULL: + av_log(NULL,0,"NULL\n"); + break; + case AMF_OBJECT: + for(i=0;i<object;i++)av_log(NULL,0," "); + av_log(NULL,0,"Object{\n"); + object++; + break; + case AMF_ECMA_ARRAY: + array = bytestream_get_be32(&ptr); + av_log(NULL,0,"Array of %d elements: [\n",array); + object++; + break; + case AMF_OBJECT_END: + object--; + for(i=0;i<object;i++)av_log(NULL,0," "); + if(array){ + array = 0; + av_log(NULL,0,"]\n"); + }else + av_log(NULL,0,"}\n"); + break; + default: + av_log(NULL,0,"Type %02X\n",ptr[-1]); + return; + } + } +} + +void rtmp_packet_inspect(AVFormatContext *ctx, RTMPPacket *pkt) +{ + av_log(NULL,0,"Packet on "); + switch (pkt->stream_id) { + case RTMP_NETWORK_CHANNEL: av_log(NULL,0,"network channel");break; + case RTMP_SYSTEM_CHANNEL: av_log(NULL,0,"system channel");break; + case RTMP_VIDEO_CHANNEL: av_log(NULL,0,"video channel");break; + case RTMP_AUDIO_CHANNEL: av_log(NULL,0,"audio channel");break; + default: av_log(NULL,0,"channel %d",pkt->stream_id); + } + av_log(NULL,0," type "); + switch (pkt->type) { + case RTMP_PT_CHUNK_SIZE: av_log(NULL,0,"chunk size %d",AV_RB32(pkt->data));break; + case RTMP_PT_BYTES_READ: av_log(NULL,0,"bytes read");break; + case RTMP_PT_PING: av_log(NULL,0,"ping");break; + case RTMP_PT_SERVER_BW: av_log(NULL,0,"server BW=%d",AV_RB32(pkt->data));break; + case RTMP_PT_CLIENT_BW: av_log(NULL,0,"client BW=%d",AV_RB32(pkt->data));break; + case RTMP_PT_AUDIO: av_log(NULL,0,"audio");break; + case RTMP_PT_VIDEO: av_log(NULL,0,"video");break; + case RTMP_PT_NOTIFY: av_log(NULL,0,"notify");break; + case RTMP_PT_INVOKE: av_log(NULL,0,"invoke");break; + default: av_log(NULL,0,"%X",pkt->type); + } + av_log(NULL,0," ts %d size %d\n",pkt->timestamp, pkt->data_size); + if (pkt->type == RTMP_PT_INVOKE || pkt->type == RTMP_PT_NOTIFY) + parse_amf(pkt->data, pkt->data_size); +} Added: rtmp/rtmppkt.h ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmppkt.h Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,135 @@ +/* + * RTMP packet utilities + * Copyright (c) 2009 Kostya Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVFORMAT_RTMPPKT_H +#define AVFORMAT_RTMPPKT_H + +#include "avformat.h" + +/* maximum possible number of different RTMP channels */ +#define RTMP_CHANNELS 64 + +enum RTMPChannel { + RTMP_NETWORK_CHANNEL = 2, ///< channel for network-related messages (bandwidth report, ping, etc) + RTMP_SYSTEM_CHANNEL, ///< channel for sending server control messages + RTMP_VIDEO_CHANNEL = 8, ///< channel for video data + RTMP_AUDIO_CHANNEL, ///< channel for audio data +}; + +typedef enum RTMPPacketType { + RTMP_PT_CHUNK_SIZE = 1, ///< chunk size change + RTMP_PT_BYTES_READ = 3, ///< number of bytes read + RTMP_PT_PING, ///< ping + RTMP_PT_SERVER_BW, ///< server bandwidth + RTMP_PT_CLIENT_BW, ///< client bandwidth + RTMP_PT_AUDIO = 8, ///< audio packet + RTMP_PT_VIDEO, ///< video packet + RTMP_PT_FLEX_STREAM = 15, ///< Flex shared stream + RTMP_PT_FLEX_OBJECT, ///< Flex shared object + RTMP_PT_FLEX_MESSAGE, ///< Flex shared message + RTMP_PT_NOTIFY, ///< some notification + RTMP_PT_SHARED_OBJ, ///< shared object + RTMP_PT_INVOKE, ///< invoke some stream action + RTMP_PT_METADATA = 22, ///< FLV metadata +} RTMPPacketType; + +typedef enum RTMPObjectType { + RTMP_OT_CONNECT, + RTMP_OT_DISCONNECT, + RTMP_OT_SET_ATTR, + RTMP_OT_UPDATE_DATA, + RTMP_OT_UPDATE_ATTR, + RTMP_OT_SEND_MSG, + RTMP_OT_STATUS, + RTMP_OT_CLEAR_DATA, + RTMP_OT_DELETE_DATA, + RTMP_OT_DELETE_ATTR, + RTMP_OT_INITIAL_DATA, +} RTMPObjectType; + +enum RTMPPacketSize { + RTMP_PS_TWELVEBYTES = 0, + RTMP_PS_EIGHTBYTES, + RTMP_PS_FOURBYTES, + RTMP_PS_ONEBYTE +}; + +typedef enum AMFType { + AMF_NUMBER = 0, + AMF_BOOLEAN, + AMF_STRING, + AMF_OBJECT, + AMF_MOVIE, + AMF_NULL, + AMF_UNDEFINED, + AMF_REFERENCE, + AMF_ECMA_ARRAY, + AMF_OBJECT_END, + AMF_STRICT_ARRAY, + AMF_DATE, + AMF_LONG_STRING, + AMF_UNSUPPORTED, + AMD_RECORD_SET, + AMF_XML_OBJECT, + AMF_TYPED_OBJECT, + + AMF_STRING_IN_OBJECT = 99, +} AMFType; + +/** + * structure for holding RTMP packets + */ +typedef struct RTMPPacket { + uint8_t stream_id; ///< stream ID + RTMPPacketType type; ///< packet type + int timestamp; ///< packet timestamp + uint8_t *data; ///< packet payload + int data_size; ///< packet payload size +} RTMPPacket; + +/** + * saved parameters for reading RTMP packets + * + * Since RTMP server may choose to send partial packet for some channel we need + * to set missing parameters from the previous packet on the same channel. + */ +typedef struct RTMPPacketHistory { + RTMPPacket prev_pkt[RTMP_CHANNELS]; ///< previous read packet parameters + int chunk_size[RTMP_CHANNELS]; ///< chunk size for each channel +} RTMPPacketHistory; + + +int rtmp_packet_create(RTMPPacket *pkt, int stream_id, RTMPPacketType type, + int timestamp, int size); + +void rtmp_packet_destroy(RTMPPacket *pkt); + +int rtmp_packet_read(AVFormatContext *ctx, URLContext *h, RTMPPacket *p, RTMPPacketHistory *hist); + +int rtmp_packet_write(AVFormatContext *ctx, URLContext *h, RTMPPacket *p, RTMPPacketHistory *hist); + +int rtmp_amf_tag_size(int type, const void *data); + +void rtmp_amf_write_tag(uint8_t **dst, AMFType type, const void *data); + +void rtmp_packet_inspect(AVFormatContext *ctx, RTMPPacket *pkt); + +#endif /* AVFORMAT_RTMPPKT_H */ Added: rtmp/rtmpproto.c ============================================================================== --- /dev/null 00:00:00 1970 (empty, because file is newly added) +++ rtmp/rtmpproto.c Thu Jun 4 08:57:37 2009 (r4382) @@ -0,0 +1,117 @@ +/* + * RTMP network protocol + * Copyright (c) 2009 Kostya Shishkov + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file libavformat/rtmpproto.c + * RTMP protocol + */ + +#include "libavutil/avstring.h" +#include "avformat.h" + +#include <unistd.h> +#include <stdarg.h> +#include "network.h" +#include "os_support.h" +#include <fcntl.h> +#if HAVE_SYS_SELECT_H +#include <sys/select.h> +#endif + +#define RTMP_DEFAULT_PORT 1935 +#define RTMP_HANDSHAKE_PACKET_SIZE 1536 + +typedef struct RTMPContext { + URLContext *rtmp_hd; + int wrote; +} RTMPContext; + + +/** + * url syntax: rtp://host:port[?option=val...] + * option: 'ttl=n' : set the ttl value (for multicast only) + * 'localport=n' : set the local port to n + * + */ + +static int rtmp_open(URLContext *h, const char *uri, int flags) +{ + RTMPContext *s; + int port, is_input; + char hostname[256]; + char buf[1024]; + char path[1024]; + + is_input = !(flags & URL_WRONLY); + + s = av_mallocz(sizeof(RTMPContext)); + if (!s) + return AVERROR(ENOMEM); + h->priv_data = s; + + url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &port, + path, sizeof(path), uri); + + if (port == -1) + port = RTMP_DEFAULT_PORT; + snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port); + + if (url_open(&s->rtmp_hd, buf, URL_RDWR) < 0) + goto fail; + + h->max_packet_size = url_get_max_packet_size(s->rtmp_hd); + h->is_streamed = 1; + url_close(s->rtmp_hd); + return 0; + + fail: + if (s->rtmp_hd) + url_close(s->rtmp_hd); + av_free(s); + return AVERROR(EIO); +} + +static int rtmp_read(URLContext *h, uint8_t *buf, int size) +{ + return 0; +} + +static int rtmp_write(URLContext *h, uint8_t *buf, int size) +{ + return 0; +} + +static int rtmp_close(URLContext *h) +{ + RTMPContext *s = h->priv_data; + + av_free(s); + return 0; +} + +URLProtocol rtmp_protocol = { + "rtmp", + rtmp_open, + rtmp_read, + rtmp_write, + NULL, /* seek */ + rtmp_close, +};

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