virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
Code Issues Releases Activity
59cbc4eee2
FFmpeg/libavformat/rtmpproto.c
Samuel Pitoiset b2e495afa8 rtmp: Support 'rtmp_live', an option which specifies if the media is a live stream. 
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-05-08 13:21:35 +03:00

1080 lines
37 KiB
C
Raw Blame History

/*
* RTMP network protocol
* Copyright (c) 2009 Kostya Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* RTMP protocol
*/
#include "libavcodec/bytestream.h"
#include "libavutil/avstring.h"
#include "libavutil/intfloat.h"
#include "libavutil/lfg.h"
#include "libavutil/opt.h"
#include "libavutil/sha.h"
#include "avformat.h"
#include "internal.h"
#include "network.h"
#include "flv.h"
#include "rtmp.h"
#include "rtmppkt.h"
#include "url.h"
//#define DEBUG
#define APP_MAX_LENGTH 128
#define PLAYPATH_MAX_LENGTH 256
/** RTMP protocol handler state */
typedef enum {
STATE_START, ///< client has not done anything yet
STATE_HANDSHAKED, ///< client has performed handshake
STATE_RELEASING, ///< client releasing stream before publish it (for output)
STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
STATE_CONNECTING, ///< client connected to server successfully
STATE_READY, ///< client has sent all needed commands and waits for server reply
STATE_PLAYING, ///< client has started receiving multimedia data from server
STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
STATE_STOPPED, ///< the broadcast has been stopped
} ClientState;
/** protocol handler context */
typedef struct RTMPContext {
const AVClass *class;
URLContext* stream; ///< TCP stream used in interactions with RTMP server
RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
int chunk_size; ///< size of the chunks RTMP packets are divided into
int is_input; ///< input/output flag
char *playpath; ///< stream identifier to play (with possible "mp4:" prefix)
int live; ///< 0: recorded, -1: live, -2: both
char *app; ///< name of application
ClientState state; ///< current state
int main_channel_id; ///< an additional channel ID which is used for some invocations
uint8_t* flv_data; ///< buffer with data for demuxer
int flv_size; ///< current buffer size
int flv_off; ///< number of bytes read from current buffer
RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
uint32_t client_report_size; ///< number of bytes after which client should report to server
uint32_t bytes_read; ///< number of bytes read from server
uint32_t last_bytes_read; ///< number of bytes read last reported to server
int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
uint8_t flv_header[11]; ///< partial incoming flv packet header
int flv_header_bytes; ///< number of initialized bytes in flv_header
int nb_invokes; ///< keeps track of invoke messages
int create_stream_invoke; ///< invoke id for the create stream command
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
/** Client key used for digest signing */
static const uint8_t rtmp_player_key[] = {
'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
};
#define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
/** Key used for RTMP server digest signing */
static const uint8_t rtmp_server_key[] = {
'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
};
/**
* Generate 'connect' call and send it to the server.
*/
static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
const char *host, int port)
{
RTMPPacket pkt;
uint8_t ver[64], *p;
char tcurl[512];
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
p = pkt.data;
ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
ff_amf_write_string(&p, "connect");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_object_start(&p);
ff_amf_write_field_name(&p, "app");
ff_amf_write_string(&p, rt->app);
if (rt->is_input) {
snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
} else {
snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
ff_amf_write_field_name(&p, "type");
ff_amf_write_string(&p, "nonprivate");
}
ff_amf_write_field_name(&p, "flashVer");
ff_amf_write_string(&p, ver);
ff_amf_write_field_name(&p, "tcUrl");
ff_amf_write_string(&p, tcurl);
if (rt->is_input) {
ff_amf_write_field_name(&p, "fpad");
ff_amf_write_bool(&p, 0);
ff_amf_write_field_name(&p, "capabilities");
ff_amf_write_number(&p, 15.0);
/* Tell the server we support all the audio codecs except
* SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
* which are unused in the RTMP protocol implementation. */
ff_amf_write_field_name(&p, "audioCodecs");
ff_amf_write_number(&p, 4071.0);
ff_amf_write_field_name(&p, "videoCodecs");
ff_amf_write_number(&p, 252.0);
ff_amf_write_field_name(&p, "videoFunction");
ff_amf_write_number(&p, 1.0);
}
ff_amf_write_object_end(&p);
pkt.data_size = p - pkt.data;
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'releaseStream' call and send it to the server. It should make
* the server release some channel for media streams.
*/
static void gen_release_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
29 + strlen(rt->playpath));
av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "releaseStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'FCPublish' call and send it to the server. It should make
* the server preapare for receiving media streams.
*/
static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
25 + strlen(rt->playpath));
av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "FCPublish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'FCUnpublish' call and send it to the server. It should make
* the server destroy stream.
*/
static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
27 + strlen(rt->playpath));
av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "FCUnpublish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'createStream' call and send it to the server. It should make
* the server allocate some channel for media streams.
*/
static void gen_create_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
p = pkt.data;
ff_amf_write_string(&p, "createStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
rt->create_stream_invoke = rt->nb_invokes;
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'deleteStream' call and send it to the server. It should make
* the server remove some channel for media streams.
*/
static void gen_delete_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
p = pkt.data;
ff_amf_write_string(&p, "deleteStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_number(&p, rt->main_channel_id);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'play' call and send it to the server, then ping the server
* to start actual playing.
*/
static void gen_play(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
29 + strlen(rt->playpath));
pkt.extra = rt->main_channel_id;
p = pkt.data;
ff_amf_write_string(&p, "play");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_amf_write_number(&p, rt->live);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
// set client buffer time disguised in ping packet
ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
p = pkt.data;
bytestream_put_be16(&p, 3);
bytestream_put_be32(&p, 1);
bytestream_put_be32(&p, 256); //TODO: what is a good value here?
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate 'publish' call and send it to the server.
*/
static void gen_publish(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
30 + strlen(rt->playpath));
pkt.extra = rt->main_channel_id;
p = pkt.data;
ff_amf_write_string(&p, "publish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_amf_write_string(&p, "live");
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate ping reply and send it to the server.
*/
static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
p = pkt.data;
bytestream_put_be16(&p, 7);
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate server bandwidth message and send it to the server.
*/
static void gen_server_bw(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW, 0, 4);
p = pkt.data;
bytestream_put_be32(&p, 2500000);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
/**
* Generate report on bytes read so far and send it to the server.
*/
static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
{
RTMPPacket pkt;
uint8_t *p;
ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
p = pkt.data;
bytestream_put_be32(&p, rt->bytes_read);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
}
//TODO: Move HMAC code somewhere. Eventually.
#define HMAC_IPAD_VAL 0x36
#define HMAC_OPAD_VAL 0x5C
/**
* Calculate HMAC-SHA2 digest for RTMP handshake packets.
*
* @param src input buffer
* @param len input buffer length (should be 1536)
* @param gap offset in buffer where 32 bytes should not be taken into account
* when calculating digest (since it will be used to store that digest)
* @param key digest key
* @param keylen digest key length
* @param dst buffer where calculated digest will be stored (32 bytes)
*/
static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
const uint8_t *key, int keylen, uint8_t *dst)
{
struct AVSHA *sha;
uint8_t hmac_buf[64+32] = {0};
int i;
sha = av_mallocz(av_sha_size);
if (keylen < 64) {
memcpy(hmac_buf, key, keylen);
} else {
av_sha_init(sha, 256);
av_sha_update(sha,key, keylen);
av_sha_final(sha, hmac_buf);
}
for (i = 0; i < 64; i++)
hmac_buf[i] ^= HMAC_IPAD_VAL;
av_sha_init(sha, 256);
av_sha_update(sha, hmac_buf, 64);
if (gap <= 0) {
av_sha_update(sha, src, len);
} else { //skip 32 bytes used for storing digest
av_sha_update(sha, src, gap);
av_sha_update(sha, src + gap + 32, len - gap - 32);
}
av_sha_final(sha, hmac_buf + 64);
for (i = 0; i < 64; i++)
hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
av_sha_init(sha, 256);
av_sha_update(sha, hmac_buf, 64+32);
av_sha_final(sha, dst);
av_free(sha);
}
/**
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
* will be stored) into that packet.
*
* @param buf handshake data (1536 bytes)
* @return offset to the digest inside input data
*/
static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
{
int i, digest_pos = 0;
for (i = 8; i < 12; i++)
digest_pos += buf[i];
digest_pos = (digest_pos % 728) + 12;
rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
buf + digest_pos);
return digest_pos;
}
/**
* Verify that the received server response has the expected digest value.
*
* @param buf handshake data received from the server (1536 bytes)
* @param off position to search digest offset from
* @return 0 if digest is valid, digest position otherwise
*/
static int rtmp_validate_digest(uint8_t *buf, int off)
{
int i, digest_pos = 0;
uint8_t digest[32];
for (i = 0; i < 4; i++)
digest_pos += buf[i + off];
digest_pos = (digest_pos % 728) + off + 4;
rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
digest);
if (!memcmp(digest, buf + digest_pos, 32))
return digest_pos;
return 0;
}
/**
* Perform handshake with the server by means of exchanging pseudorandom data
* signed with HMAC-SHA2 digest.
*
* @return 0 if handshake succeeds, negative value otherwise
*/
static int rtmp_handshake(URLContext *s, RTMPContext *rt)
{
AVLFG rnd;
uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
3, // unencrypted data
0, 0, 0, 0, // client uptime
RTMP_CLIENT_VER1,
RTMP_CLIENT_VER2,
RTMP_CLIENT_VER3,
RTMP_CLIENT_VER4,
};
uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
int i;
int server_pos, client_pos;
uint8_t digest[32];
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
av_lfg_init(&rnd, 0xDEADC0DE);
// generate handshake packet - 1536 bytes of pseudorandom data
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
tosend[i] = av_lfg_get(&rnd) >> 24;
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
return -1;
}
i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
return -1;
}
av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
if (rt->is_input && serverdata[5] >= 3) {
server_pos = rtmp_validate_digest(serverdata + 1, 772);
if (!server_pos) {
server_pos = rtmp_validate_digest(serverdata + 1, 8);
if (!server_pos) {
av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
return -1;
}
}
rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
rtmp_server_key, sizeof(rtmp_server_key),
digest);
rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
digest, 32,
digest);
if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
return -1;
}
for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
tosend[i] = av_lfg_get(&rnd) >> 24;
rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
rtmp_player_key, sizeof(rtmp_player_key),
digest);
rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
digest, 32,
tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
// write reply back to the server
ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
} else {
ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
}
return 0;
}
/**
* Parse received packet and possibly perform some action depending on
* the packet contents.
* @return 0 for no errors, negative values for serious errors which prevent
* further communications, positive values for uncritical errors
*/
static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
{
int i, t;
const uint8_t *data_end = pkt->data + pkt->data_size;
#ifdef DEBUG
ff_rtmp_packet_dump(s, pkt);
#endif
switch (pkt->type) {
case RTMP_PT_CHUNK_SIZE:
if (pkt->data_size != 4) {
av_log(s, AV_LOG_ERROR,
"Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
return -1;
}
if (!rt->is_input)
ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
rt->chunk_size = AV_RB32(pkt->data);
if (rt->chunk_size <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
return -1;
}
av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
break;
case RTMP_PT_PING:
t = AV_RB16(pkt->data);
if (t == 6)
gen_pong(s, rt, pkt);
break;
case RTMP_PT_CLIENT_BW:
if (pkt->data_size < 4) {
av_log(s, AV_LOG_ERROR,
"Client bandwidth report packet is less than 4 bytes long (%d)\n",
pkt->data_size);
return -1;
}
av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
rt->client_report_size = AV_RB32(pkt->data) >> 1;
break;
case RTMP_PT_INVOKE:
//TODO: check for the messages sent for wrong state?
if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
uint8_t tmpstr[256];
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
"description", tmpstr, sizeof(tmpstr)))
av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
return -1;
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
switch (rt->state) {
case STATE_HANDSHAKED:
if (!rt->is_input) {
gen_release_stream(s, rt);
gen_fcpublish_stream(s, rt);
rt->state = STATE_RELEASING;
} else {
gen_server_bw(s, rt);
rt->state = STATE_CONNECTING;
}
gen_create_stream(s, rt);
break;
case STATE_FCPUBLISH:
rt->state = STATE_CONNECTING;
break;
case STATE_RELEASING:
rt->state = STATE_FCPUBLISH;
/* hack for Wowza Media Server, it does not send result for
* releaseStream and FCPublish calls */
if (!pkt->data[10]) {
int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
if (pkt_id == rt->create_stream_invoke)
rt->state = STATE_CONNECTING;
}
if (rt->state != STATE_CONNECTING)
break;
case STATE_CONNECTING:
//extract a number from the result
if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
} else {
rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
}
if (rt->is_input) {
gen_play(s, rt);
} else {
gen_publish(s, rt);
}
rt->state = STATE_READY;
break;
}
} else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
const uint8_t* ptr = pkt->data + 11;
uint8_t tmpstr[256];
for (i = 0; i < 2; i++) {
t = ff_amf_tag_size(ptr, data_end);
if (t < 0)
return 1;
ptr += t;
}
t = ff_amf_get_field_value(ptr, data_end,
"level", tmpstr, sizeof(tmpstr));
if (!t && !strcmp(tmpstr, "error")) {
if (!ff_amf_get_field_value(ptr, data_end,
"description", tmpstr, sizeof(tmpstr)))
av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
return -1;
}
t = ff_amf_get_field_value(ptr, data_end,
"code", tmpstr, sizeof(tmpstr));
if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
}
break;
}
return 0;
}
/**
* Interact with the server by receiving and sending RTMP packets until
* there is some significant data (media data or expected status notification).
*
* @param s reading context
* @param for_header non-zero value tells function to work until it
* gets notification from the server that playing has been started,
* otherwise function will work until some media data is received (or
* an error happens)
* @return 0 for successful operation, negative value in case of error
*/
static int get_packet(URLContext *s, int for_header)
{
RTMPContext *rt = s->priv_data;
int ret;
uint8_t *p;
const uint8_t *next;
uint32_t data_size;
uint32_t ts, cts, pts=0;
if (rt->state == STATE_STOPPED)
return AVERROR_EOF;
for (;;) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
rt->chunk_size, rt->prev_pkt[0])) <= 0) {
if (ret == 0) {
return AVERROR(EAGAIN);
} else {
return AVERROR(EIO);
}
}
rt->bytes_read += ret;
if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
gen_bytes_read(s, rt, rpkt.timestamp + 1);
rt->last_bytes_read = rt->bytes_read;
}
ret = rtmp_parse_result(s, rt, &rpkt);
if (ret < 0) {//serious error in current packet
ff_rtmp_packet_destroy(&rpkt);
return -1;
}
if (rt->state == STATE_STOPPED) {
ff_rtmp_packet_destroy(&rpkt);
return AVERROR_EOF;
}
if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
ff_rtmp_packet_destroy(&rpkt);
return 0;
}
if (!rpkt.data_size || !rt->is_input) {
ff_rtmp_packet_destroy(&rpkt);
continue;
}
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
(rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
ts = rpkt.timestamp;
// generate packet header and put data into buffer for FLV demuxer
rt->flv_off = 0;
rt->flv_size = rpkt.data_size + 15;
rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
bytestream_put_byte(&p, rpkt.type);
bytestream_put_be24(&p, rpkt.data_size);
bytestream_put_be24(&p, ts);
bytestream_put_byte(&p, ts >> 24);
bytestream_put_be24(&p, 0);
bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
bytestream_put_be32(&p, 0);
ff_rtmp_packet_destroy(&rpkt);
return 0;
} else if (rpkt.type == RTMP_PT_METADATA) {
// we got raw FLV data, make it available for FLV demuxer
rt->flv_off = 0;
rt->flv_size = rpkt.data_size;
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
/* rewrite timestamps */
next = rpkt.data;
ts = rpkt.timestamp;
while (next - rpkt.data < rpkt.data_size - 11) {
next++;
data_size = bytestream_get_be24(&next);
p=next;
cts = bytestream_get_be24(&next);
cts |= bytestream_get_byte(&next) << 24;
if (pts==0)
pts=cts;
ts += cts - pts;
pts = cts;
bytestream_put_be24(&p, ts);
bytestream_put_byte(&p, ts >> 24);
next += data_size + 3 + 4;
}
memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
ff_rtmp_packet_destroy(&rpkt);
return 0;
}
ff_rtmp_packet_destroy(&rpkt);
}
}
static int rtmp_close(URLContext *h)
{
RTMPContext *rt = h->priv_data;
if (!rt->is_input) {
rt->flv_data = NULL;
if (rt->out_pkt.data_size)
ff_rtmp_packet_destroy(&rt->out_pkt);
if (rt->state > STATE_FCPUBLISH)
gen_fcunpublish_stream(h, rt);
}
if (rt->state > STATE_HANDSHAKED)
gen_delete_stream(h, rt);
av_freep(&rt->flv_data);
ffurl_close(rt->stream);
return 0;
}
/**
* Open RTMP connection and verify that the stream can be played.
*
* URL syntax: rtmp://server[:port][/app][/playpath]
* where 'app' is first one or two directories in the path
* (e.g. /ondemand/, /flash/live/, etc.)
* and 'playpath' is a file name (the rest of the path,
* may be prefixed with "mp4:")
*/
static int rtmp_open(URLContext *s, const char *uri, int flags)
{
RTMPContext *rt = s->priv_data;
char proto[8], hostname[256], path[1024], *fname;
char *old_app;
uint8_t buf[2048];
int port;
int ret;
rt->is_input = !(flags & AVIO_FLAG_WRITE);
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
path, sizeof(path), s->filename);
if (port < 0)
port = RTMP_DEFAULT_PORT;
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL) < 0) {
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
goto fail;
}
rt->state = STATE_START;
if (rtmp_handshake(s, rt))
goto fail;
rt->chunk_size = 128;
rt->state = STATE_HANDSHAKED;
// Keep the application name when it has been defined by the user.
old_app = rt->app;
rt->app = av_malloc(APP_MAX_LENGTH);
if (!rt->app) {
rtmp_close(s);
return AVERROR(ENOMEM);
}
//extract "app" part from path
if (!strncmp(path, "/ondemand/", 10)) {
fname = path + 10;
memcpy(rt->app, "ondemand", 9);
} else {
char *p = strchr(path + 1, '/');
if (!p) {
fname = path + 1;
rt->app[0] = '\0';
} else {
char *c = strchr(p + 1, ':');
fname = strchr(p + 1, '/');
if (!fname || c < fname) {
fname = p + 1;
av_strlcpy(rt->app, path + 1, p - path);
} else {
fname++;
av_strlcpy(rt->app, path + 1, fname - path - 1);
}
}
}
if (old_app) {
// The name of application has been defined by the user, override it.
av_free(rt->app);
rt->app = old_app;
}
if (!rt->playpath) {
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
if (!rt->playpath) {
rtmp_close(s);
return AVERROR(ENOMEM);
}
if (!strchr(fname, ':') &&
(!strcmp(fname + strlen(fname) - 4, ".f4v") ||
!strcmp(fname + strlen(fname) - 4, ".mp4"))) {
memcpy(rt->playpath, "mp4:", 5);
} else {
rt->playpath[0] = 0;
}
strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
}
rt->client_report_size = 1048576;
rt->bytes_read = 0;
rt->last_bytes_read = 0;
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
proto, path, rt->app, rt->playpath);
gen_connect(s, rt, proto, hostname, port);
do {
ret = get_packet(s, 1);
} while (ret == EAGAIN);
if (ret < 0)
goto fail;
if (rt->is_input) {
// generate FLV header for demuxer
rt->flv_size = 13;
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
} else {
rt->flv_size = 0;
rt->flv_data = NULL;
rt->flv_off = 0;
rt->skip_bytes = 13;
}
s->max_packet_size = rt->stream->max_packet_size;
s->is_streamed = 1;
return 0;
fail:
rtmp_close(s);
return AVERROR(EIO);
}
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int orig_size = size;
int ret;
while (size > 0) {
int data_left = rt->flv_size - rt->flv_off;
if (data_left >= size) {
memcpy(buf, rt->flv_data + rt->flv_off, size);
rt->flv_off += size;
return orig_size;
}
if (data_left > 0) {
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
buf += data_left;
size -= data_left;
rt->flv_off = rt->flv_size;
return data_left;
}
if ((ret = get_packet(s, 0)) < 0)
return ret;
}
return orig_size;
}
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int size_temp = size;
int pktsize, pkttype;
uint32_t ts;
const uint8_t *buf_temp = buf;
do {
if (rt->skip_bytes) {
int skip = FFMIN(rt->skip_bytes, size_temp);
buf_temp += skip;
size_temp -= skip;
rt->skip_bytes -= skip;
continue;
}
if (rt->flv_header_bytes < 11) {
const uint8_t *header = rt->flv_header;
int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
rt->flv_header_bytes += copy;
size_temp -= copy;
if (rt->flv_header_bytes < 11)
break;
pkttype = bytestream_get_byte(&header);
pktsize = bytestream_get_be24(&header);
ts = bytestream_get_be24(&header);
ts |= bytestream_get_byte(&header) << 24;
bytestream_get_be24(&header);
rt->flv_size = pktsize;
//force 12bytes header
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
pkttype == RTMP_PT_NOTIFY) {
if (pkttype == RTMP_PT_NOTIFY)
pktsize += 16;
rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
}
//this can be a big packet, it's better to send it right here
ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
rt->out_pkt.extra = rt->main_channel_id;
rt->flv_data = rt->out_pkt.data;
if (pkttype == RTMP_PT_NOTIFY)
ff_amf_write_string(&rt->flv_data, "@setDataFrame");
}
if (rt->flv_size - rt->flv_off > size_temp) {
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
rt->flv_off += size_temp;
size_temp = 0;
} else {
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
size_temp -= rt->flv_size - rt->flv_off;
rt->flv_off += rt->flv_size - rt->flv_off;
}
if (rt->flv_off == rt->flv_size) {
rt->skip_bytes = 4;
ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&rt->out_pkt);
rt->flv_size = 0;
rt->flv_off = 0;
rt->flv_header_bytes = 0;
}
} while (buf_temp - buf < size);
return size;
}
#define OFFSET(x) offsetof(RTMPContext, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
#define ENC AV_OPT_FLAG_ENCODING_PARAM
static const AVOption rtmp_options[] = {
{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
{"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{ NULL },
};
static const AVClass rtmp_class = {
.class_name = "rtmp",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmp_protocol = {
.name = "rtmp",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class= &rtmp_class,
};
Reference in New Issue View Git Blame Copy Permalink