X-Git-Url: https://git.pterodactylus.net/?a=blobdiff_plain;f=synfig-core%2Ftags%2Fsynfig_0_61_05%2Fsynfig-core%2Fsrc%2Fmodules%2Fmod_libavcodec%2Flibavformat%2Frtp.c;fp=synfig-core%2Ftags%2Fsynfig_0_61_05%2Fsynfig-core%2Fsrc%2Fmodules%2Fmod_libavcodec%2Flibavformat%2Frtp.c;h=0000000000000000000000000000000000000000;hb=6fa8f2f38d4b0b35f8539bf94e27ae27015c7689;hp=113828475eef5145d1f284a85ef33407088c7957;hpb=47fce282611fbba1044921d22ca887f9b53ad91a;p=synfig.git diff --git a/synfig-core/tags/synfig_0_61_05/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c b/synfig-core/tags/synfig_0_61_05/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c deleted file mode 100644 index 1138284..0000000 --- a/synfig-core/tags/synfig_0_61_05/synfig-core/src/modules/mod_libavcodec/libavformat/rtp.c +++ /dev/null @@ -1,702 +0,0 @@ -/* - * 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; -}