[FFmpeg-soc] [soc]: r4382 - in rtmp: TODO checkout.sh rtmp.h rtmp.patch rtmpdec.c rtmppkt.c rtmppkt.h rtmpproto.c
kostya
subversion at mplayerhq.hu
Thu Jun 4 08:57:37 CEST 2009
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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