X-Git-Url: https://git.pterodactylus.net/?a=blobdiff_plain;f=synfig-core%2Ftags%2Fsynfig_0_61_03%2Fsynfig-core%2Fsrc%2Fmodules%2Fmod_libavcodec%2Flibavformat%2Frtp.c;fp=synfig-core%2Ftags%2Fsynfig_0_61_03%2Fsynfig-core%2Fsrc%2Fmodules%2Fmod_libavcodec%2Flibavformat%2Frtp.c;h=113828475eef5145d1f284a85ef33407088c7957;hb=2362dce2bb8c6a5fd5f3bd049975c3d89411f5d3;hp=0000000000000000000000000000000000000000;hpb=6f41e88a004056081fbf3e45c4eb9780cc08cd91;p=synfig.git diff --git a/synfig-core/tags/synfig_0_61_03/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c b/synfig-core/tags/synfig_0_61_03/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c new file mode 100644 index 0000000..1138284 --- /dev/null +++ b/synfig-core/tags/synfig_0_61_03/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c @@ -0,0 +1,702 @@ +/* + * RTP input/output format + * Copyright (c) 2002 Fabrice Bellard. + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include "avformat.h" + +#include +#include +#include +#include +#ifndef __BEOS__ +# include +#else +# include "barpainet.h" +#endif +#include + +//#define DEBUG + + +/* TODO: - add RTCP statistics reporting (should be optional). + + - add support for h263/mpeg4 packetized output : IDEA: send a + buffer to 'rtp_write_packet' contains all the packets for ONE + frame. Each packet should have a four byte header containing + the length in big endian format (same trick as + 'url_open_dyn_packet_buf') +*/ + +#define RTP_VERSION 2 + +#define RTP_MAX_SDES 256 /* maximum text length for SDES */ + +/* RTCP paquets use 0.5 % of the bandwidth */ +#define RTCP_TX_RATIO_NUM 5 +#define RTCP_TX_RATIO_DEN 1000 + +typedef enum { + RTCP_SR = 200, + RTCP_RR = 201, + RTCP_SDES = 202, + RTCP_BYE = 203, + RTCP_APP = 204 +} rtcp_type_t; + +typedef enum { + RTCP_SDES_END = 0, + RTCP_SDES_CNAME = 1, + RTCP_SDES_NAME = 2, + RTCP_SDES_EMAIL = 3, + RTCP_SDES_PHONE = 4, + RTCP_SDES_LOC = 5, + RTCP_SDES_TOOL = 6, + RTCP_SDES_NOTE = 7, + RTCP_SDES_PRIV = 8, + RTCP_SDES_IMG = 9, + RTCP_SDES_DOOR = 10, + RTCP_SDES_SOURCE = 11 +} rtcp_sdes_type_t; + +enum RTPPayloadType { + RTP_PT_ULAW = 0, + RTP_PT_GSM = 3, + RTP_PT_G723 = 4, + RTP_PT_ALAW = 8, + RTP_PT_S16BE_STEREO = 10, + RTP_PT_S16BE_MONO = 11, + RTP_PT_MPEGAUDIO = 14, + RTP_PT_JPEG = 26, + RTP_PT_H261 = 31, + RTP_PT_MPEGVIDEO = 32, + RTP_PT_MPEG2TS = 33, + RTP_PT_H263 = 34, /* old H263 encapsulation */ + RTP_PT_PRIVATE = 96, +}; + +typedef struct RTPContext { + int payload_type; + uint32_t ssrc; + uint16_t seq; + uint32_t timestamp; + uint32_t base_timestamp; + uint32_t cur_timestamp; + int max_payload_size; + /* rtcp sender statistics receive */ + int64_t last_rtcp_ntp_time; + int64_t first_rtcp_ntp_time; + uint32_t last_rtcp_timestamp; + /* rtcp sender statistics */ + unsigned int packet_count; + unsigned int octet_count; + unsigned int last_octet_count; + int first_packet; + /* buffer for output */ + uint8_t buf[RTP_MAX_PACKET_LENGTH]; + uint8_t *buf_ptr; +} RTPContext; + +int rtp_get_codec_info(AVCodecContext *codec, int payload_type) +{ + switch(payload_type) { + case RTP_PT_ULAW: + codec->codec_id = CODEC_ID_PCM_MULAW; + codec->channels = 1; + codec->sample_rate = 8000; + break; + case RTP_PT_ALAW: + codec->codec_id = CODEC_ID_PCM_ALAW; + codec->channels = 1; + codec->sample_rate = 8000; + break; + case RTP_PT_S16BE_STEREO: + codec->codec_id = CODEC_ID_PCM_S16BE; + codec->channels = 2; + codec->sample_rate = 44100; + break; + case RTP_PT_S16BE_MONO: + codec->codec_id = CODEC_ID_PCM_S16BE; + codec->channels = 1; + codec->sample_rate = 44100; + break; + case RTP_PT_MPEGAUDIO: + codec->codec_id = CODEC_ID_MP2; + break; + case RTP_PT_JPEG: + codec->codec_id = CODEC_ID_MJPEG; + break; + case RTP_PT_MPEGVIDEO: + codec->codec_id = CODEC_ID_MPEG1VIDEO; + break; + default: + return -1; + } + return 0; +} + +/* return < 0 if unknown payload type */ +int rtp_get_payload_type(AVCodecContext *codec) +{ + int payload_type; + + /* compute the payload type */ + payload_type = -1; + switch(codec->codec_id) { + case CODEC_ID_PCM_MULAW: + payload_type = RTP_PT_ULAW; + break; + case CODEC_ID_PCM_ALAW: + payload_type = RTP_PT_ALAW; + break; + case CODEC_ID_PCM_S16BE: + if (codec->channels == 1) { + payload_type = RTP_PT_S16BE_MONO; + } else if (codec->channels == 2) { + payload_type = RTP_PT_S16BE_STEREO; + } + break; + case CODEC_ID_MP2: + case CODEC_ID_MP3: + payload_type = RTP_PT_MPEGAUDIO; + break; + case CODEC_ID_MJPEG: + payload_type = RTP_PT_JPEG; + break; + case CODEC_ID_MPEG1VIDEO: + payload_type = RTP_PT_MPEGVIDEO; + break; + default: + break; + } + return payload_type; +} + +static inline uint32_t decode_be32(const uint8_t *p) +{ + return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; +} + +static inline uint64_t decode_be64(const uint8_t *p) +{ + return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); +} + +static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len) +{ + RTPContext *s = s1->priv_data; + + if (buf[1] != 200) + return -1; + s->last_rtcp_ntp_time = decode_be64(buf + 8); + if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) + s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; + s->last_rtcp_timestamp = decode_be32(buf + 16); + return 0; +} + +/** + * Parse an RTP packet directly sent as raw data. Can only be used if + * 'raw' is given as input file + * @param s1 media file context + * @param pkt returned packet + * @param buf input buffer + * @param len buffer len + * @return zero if no error. + */ +int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, + const unsigned char *buf, int len) +{ + RTPContext *s = s1->priv_data; + unsigned int ssrc, h; + int payload_type, seq, delta_timestamp; + AVStream *st; + uint32_t timestamp; + + if (len < 12) + return -1; + + if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) + return -1; + if (buf[1] >= 200 && buf[1] <= 204) { + rtcp_parse_packet(s1, buf, len); + return -1; + } + payload_type = buf[1] & 0x7f; + seq = (buf[2] << 8) | buf[3]; + timestamp = decode_be32(buf + 4); + ssrc = decode_be32(buf + 8); + + if (s->payload_type < 0) { + s->payload_type = payload_type; + + if (payload_type == RTP_PT_MPEG2TS) { + /* XXX: special case : not a single codec but a whole stream */ + return -1; + } else { + st = av_new_stream(s1, 0); + if (!st) + return -1; + rtp_get_codec_info(&st->codec, payload_type); + } + } + + /* NOTE: we can handle only one payload type */ + if (s->payload_type != payload_type) + return -1; +#if defined(DEBUG) || 1 + if (seq != ((s->seq + 1) & 0xffff)) { + printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", + payload_type, seq, ((s->seq + 1) & 0xffff)); + } + s->seq = seq; +#endif + len -= 12; + buf += 12; + st = s1->streams[0]; + switch(st->codec.codec_id) { + case CODEC_ID_MP2: + /* better than nothing: skip mpeg audio RTP header */ + if (len <= 4) + return -1; + h = decode_be32(buf); + len -= 4; + buf += 4; + av_new_packet(pkt, len); + memcpy(pkt->data, buf, len); + break; + case CODEC_ID_MPEG1VIDEO: + /* better than nothing: skip mpeg audio RTP header */ + if (len <= 4) + return -1; + h = decode_be32(buf); + buf += 4; + len -= 4; + if (h & (1 << 26)) { + /* mpeg2 */ + if (len <= 4) + return -1; + buf += 4; + len -= 4; + } + av_new_packet(pkt, len); + memcpy(pkt->data, buf, len); + break; + default: + av_new_packet(pkt, len); + memcpy(pkt->data, buf, len); + break; + } + + switch(st->codec.codec_id) { + case CODEC_ID_MP2: + case CODEC_ID_MPEG1VIDEO: + if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { + int64_t addend; + /* XXX: is it really necessary to unify the timestamp base ? */ + /* compute pts from timestamp with received ntp_time */ + delta_timestamp = timestamp - s->last_rtcp_timestamp; + /* convert to 90 kHz without overflow */ + addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; + addend = (addend * 5625) >> 14; + pkt->pts = addend + delta_timestamp; + } + break; + default: + /* no timestamp info yet */ + break; + } + return 0; +} + +static int rtp_read_header(AVFormatContext *s1, + AVFormatParameters *ap) +{ + RTPContext *s = s1->priv_data; + s->payload_type = -1; + s->last_rtcp_ntp_time = AV_NOPTS_VALUE; + s->first_rtcp_ntp_time = AV_NOPTS_VALUE; + return 0; +} + +static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt) +{ + char buf[RTP_MAX_PACKET_LENGTH]; + int ret; + + /* XXX: needs a better API for packet handling ? */ + for(;;) { + ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf)); + if (ret < 0) + return AVERROR_IO; + if (rtp_parse_packet(s1, pkt, buf, ret) == 0) + break; + } + return 0; +} + +static int rtp_read_close(AVFormatContext *s1) +{ + // RTPContext *s = s1->priv_data; + return 0; +} + +static int rtp_probe(AVProbeData *p) +{ + if (strstart(p->filename, "rtp://", NULL)) + return AVPROBE_SCORE_MAX; + return 0; +} + +/* rtp output */ + +static int rtp_write_header(AVFormatContext *s1) +{ + RTPContext *s = s1->priv_data; + int payload_type, max_packet_size; + AVStream *st; + + if (s1->nb_streams != 1) + return -1; + st = s1->streams[0]; + + payload_type = rtp_get_payload_type(&st->codec); + if (payload_type < 0) + payload_type = RTP_PT_PRIVATE; /* private payload type */ + s->payload_type = payload_type; + + s->base_timestamp = random(); + s->timestamp = s->base_timestamp; + s->ssrc = random(); + s->first_packet = 1; + + max_packet_size = url_fget_max_packet_size(&s1->pb); + if (max_packet_size <= 12) + return AVERROR_IO; + s->max_payload_size = max_packet_size - 12; + + switch(st->codec.codec_id) { + case CODEC_ID_MP2: + case CODEC_ID_MP3: + s->buf_ptr = s->buf + 4; + s->cur_timestamp = 0; + break; + case CODEC_ID_MPEG1VIDEO: + s->cur_timestamp = 0; + break; + default: + s->buf_ptr = s->buf; + break; + } + + return 0; +} + +/* send an rtcp sender report packet */ +static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) +{ + RTPContext *s = s1->priv_data; +#if defined(DEBUG) + printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp); +#endif + put_byte(&s1->pb, (RTP_VERSION << 6)); + put_byte(&s1->pb, 200); + put_be16(&s1->pb, 6); /* length in words - 1 */ + put_be32(&s1->pb, s->ssrc); + put_be64(&s1->pb, ntp_time); + put_be32(&s1->pb, s->timestamp); + put_be32(&s1->pb, s->packet_count); + put_be32(&s1->pb, s->octet_count); + put_flush_packet(&s1->pb); +} + +/* send an rtp packet. sequence number is incremented, but the caller + must update the timestamp itself */ +static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len) +{ + RTPContext *s = s1->priv_data; + +#ifdef DEBUG + printf("rtp_send_data size=%d\n", len); +#endif + + /* build the RTP header */ + put_byte(&s1->pb, (RTP_VERSION << 6)); + put_byte(&s1->pb, s->payload_type & 0x7f); + put_be16(&s1->pb, s->seq); + put_be32(&s1->pb, s->timestamp); + put_be32(&s1->pb, s->ssrc); + + put_buffer(&s1->pb, buf1, len); + put_flush_packet(&s1->pb); + + s->seq++; + s->octet_count += len; + s->packet_count++; +} + +/* send an integer number of samples and compute time stamp and fill + the rtp send buffer before sending. */ +static void rtp_send_samples(AVFormatContext *s1, + const uint8_t *buf1, int size, int sample_size) +{ + RTPContext *s = s1->priv_data; + int len, max_packet_size, n; + + max_packet_size = (s->max_payload_size / sample_size) * sample_size; + /* not needed, but who nows */ + if ((size % sample_size) != 0) + av_abort(); + while (size > 0) { + len = (max_packet_size - (s->buf_ptr - s->buf)); + if (len > size) + len = size; + + /* copy data */ + memcpy(s->buf_ptr, buf1, len); + s->buf_ptr += len; + buf1 += len; + size -= len; + n = (s->buf_ptr - s->buf); + /* if buffer full, then send it */ + if (n >= max_packet_size) { + rtp_send_data(s1, s->buf, n); + s->buf_ptr = s->buf; + /* update timestamp */ + s->timestamp += n / sample_size; + } + } +} + +/* NOTE: we suppose that exactly one frame is given as argument here */ +/* XXX: test it */ +static void rtp_send_mpegaudio(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, count, max_packet_size; + + max_packet_size = s->max_payload_size; + + /* test if we must flush because not enough space */ + len = (s->buf_ptr - s->buf); + if ((len + size) > max_packet_size) { + if (len > 4) { + rtp_send_data(s1, s->buf, s->buf_ptr - s->buf); + s->buf_ptr = s->buf + 4; + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + (s->cur_timestamp * 90000LL) / st->codec.sample_rate; + } + } + + /* add the packet */ + if (size > max_packet_size) { + /* big packet: fragment */ + count = 0; + while (size > 0) { + len = max_packet_size - 4; + if (len > size) + len = size; + /* build fragmented packet */ + s->buf[0] = 0; + s->buf[1] = 0; + s->buf[2] = count >> 8; + s->buf[3] = count; + memcpy(s->buf + 4, buf1, len); + rtp_send_data(s1, s->buf, len + 4); + size -= len; + buf1 += len; + count += len; + } + } else { + if (s->buf_ptr == s->buf + 4) { + /* no fragmentation possible */ + s->buf[0] = 0; + s->buf[1] = 0; + s->buf[2] = 0; + s->buf[3] = 0; + } + memcpy(s->buf_ptr, buf1, size); + s->buf_ptr += size; + } + s->cur_timestamp += st->codec.frame_size; +} + +/* NOTE: a single frame must be passed with sequence header if + needed. XXX: use slices. */ +static void rtp_send_mpegvideo(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, h, max_packet_size; + uint8_t *q; + + max_packet_size = s->max_payload_size; + + while (size > 0) { + /* XXX: more correct headers */ + h = 0; + if (st->codec.sub_id == 2) + h |= 1 << 26; /* mpeg 2 indicator */ + q = s->buf; + *q++ = h >> 24; + *q++ = h >> 16; + *q++ = h >> 8; + *q++ = h; + + if (st->codec.sub_id == 2) { + h = 0; + *q++ = h >> 24; + *q++ = h >> 16; + *q++ = h >> 8; + *q++ = h; + } + + len = max_packet_size - (q - s->buf); + if (len > size) + len = size; + + memcpy(q, buf1, len); + q += len; + + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate); + rtp_send_data(s1, s->buf, q - s->buf); + + buf1 += len; + size -= len; + } + s->cur_timestamp++; +} + +static void rtp_send_raw(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, max_packet_size; + + max_packet_size = s->max_payload_size; + + while (size > 0) { + len = max_packet_size; + if (len > size) + len = size; + + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate); + rtp_send_data(s1, buf1, len); + + buf1 += len; + size -= len; + } + s->cur_timestamp++; +} + +/* write an RTP packet. 'buf1' must contain a single specific frame. */ +static int rtp_write_packet(AVFormatContext *s1, int stream_index, + const uint8_t *buf1, int size, int64_t pts) +{ + RTPContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int rtcp_bytes; + int64_t ntp_time; + +#ifdef DEBUG + printf("%d: write len=%d\n", stream_index, size); +#endif + + /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ + rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / + RTCP_TX_RATIO_DEN; + if (s->first_packet || rtcp_bytes >= 28) { + /* compute NTP time */ + /* XXX: 90 kHz timestamp hardcoded */ + ntp_time = (pts << 28) / 5625; + rtcp_send_sr(s1, ntp_time); + s->last_octet_count = s->octet_count; + s->first_packet = 0; + } + + switch(st->codec.codec_id) { + case CODEC_ID_PCM_MULAW: + case CODEC_ID_PCM_ALAW: + case CODEC_ID_PCM_U8: + case CODEC_ID_PCM_S8: + rtp_send_samples(s1, buf1, size, 1 * st->codec.channels); + break; + case CODEC_ID_PCM_U16BE: + case CODEC_ID_PCM_U16LE: + case CODEC_ID_PCM_S16BE: + case CODEC_ID_PCM_S16LE: + rtp_send_samples(s1, buf1, size, 2 * st->codec.channels); + break; + case CODEC_ID_MP2: + case CODEC_ID_MP3: + rtp_send_mpegaudio(s1, buf1, size); + break; + case CODEC_ID_MPEG1VIDEO: + rtp_send_mpegvideo(s1, buf1, size); + break; + default: + /* better than nothing : send the codec raw data */ + rtp_send_raw(s1, buf1, size); + break; + } + return 0; +} + +static int rtp_write_trailer(AVFormatContext *s1) +{ + // RTPContext *s = s1->priv_data; + return 0; +} + +AVInputFormat rtp_demux = { + "rtp", + "RTP input format", + sizeof(RTPContext), + rtp_probe, + rtp_read_header, + rtp_read_packet, + rtp_read_close, + .flags = AVFMT_NOHEADER, +}; + +AVOutputFormat rtp_mux = { + "rtp", + "RTP output format", + NULL, + NULL, + sizeof(RTPContext), + CODEC_ID_PCM_MULAW, + CODEC_ID_NONE, + rtp_write_header, + rtp_write_packet, + rtp_write_trailer, +}; + +int rtp_init(void) +{ + av_register_output_format(&rtp_mux); + av_register_input_format(&rtp_demux); + return 0; +}