/*
* RTP output format
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"
+#include "internal.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/random_seed.h"
+#include "libavutil/opt.h"
-#include <unistd.h>
-#include "network.h"
-
-#include "rtp_internal.h"
-#include "rtp_mpv.h"
-#include "rtp_aac.h"
-#include "rtp_h264.h"
+#include "rtpenc.h"
//#define DEBUG
+static const AVOption options[] = {
+ FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
+ { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
+ { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { NULL },
+};
+
+static const AVClass rtp_muxer_class = {
+ .class_name = "RTP muxer",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
#define RTCP_SR_SIZE 28
-#define NTP_OFFSET 2208988800ULL
-#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
-static uint64_t ntp_time(void)
+static int is_supported(enum CodecID id)
{
- return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
+ switch(id) {
+ case CODEC_ID_H263:
+ case CODEC_ID_H263P:
+ case CODEC_ID_H264:
+ case CODEC_ID_MPEG1VIDEO:
+ case CODEC_ID_MPEG2VIDEO:
+ case CODEC_ID_MPEG4:
+ case CODEC_ID_AAC:
+ case CODEC_ID_MP2:
+ case CODEC_ID_MP3:
+ case CODEC_ID_PCM_ALAW:
+ case CODEC_ID_PCM_MULAW:
+ case CODEC_ID_PCM_S8:
+ case CODEC_ID_PCM_S16BE:
+ case CODEC_ID_PCM_S16LE:
+ case CODEC_ID_PCM_U16BE:
+ case CODEC_ID_PCM_U16LE:
+ case CODEC_ID_PCM_U8:
+ case CODEC_ID_MPEG2TS:
+ case CODEC_ID_AMR_NB:
+ case CODEC_ID_AMR_WB:
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ case CODEC_ID_VP8:
+ case CODEC_ID_ADPCM_G722:
+ case CODEC_ID_ADPCM_G726:
+ case CODEC_ID_ILBC:
+ return 1;
+ default:
+ return 0;
+ }
}
static int rtp_write_header(AVFormatContext *s1)
{
- RTPDemuxContext *s = s1->priv_data;
- int payload_type, max_packet_size, n;
+ RTPMuxContext *s = s1->priv_data;
+ int n;
AVStream *st;
- if (s1->nb_streams != 1)
- return -1;
+ if (s1->nb_streams != 1) {
+ av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
+ return AVERROR(EINVAL);
+ }
st = s1->streams[0];
+ if (!is_supported(st->codec->codec_id)) {
+ av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
- 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;
+ return -1;
+ }
-// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
- s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
+ if (s->payload_type < 0)
+ s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
+ s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
- s->ssrc = 0; /* FIXME: was random(), what should this be? */
+ if (!s->ssrc)
+ s->ssrc = av_get_random_seed();
s->first_packet = 1;
- s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
-
- max_packet_size = url_fget_max_packet_size(s1->pb);
- if (max_packet_size <= 12)
+ s->first_rtcp_ntp_time = ff_ntp_time();
+ if (s1->start_time_realtime)
+ /* Round the NTP time to whole milliseconds. */
+ s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
+ NTP_OFFSET_US;
+
+ if (s1->packet_size) {
+ if (s1->pb->max_packet_size)
+ s1->packet_size = FFMIN(s1->packet_size,
+ s1->pb->max_packet_size);
+ } else
+ s1->packet_size = s1->pb->max_packet_size;
+ if (s1->packet_size <= 12) {
+ av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
- s->max_payload_size = max_packet_size - 12;
+ }
+ s->buf = av_malloc(s1->packet_size);
+ if (s->buf == NULL) {
+ return AVERROR(ENOMEM);
+ }
+ s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
- if (s1->max_delay) {
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
- if (st->codec->frame_size == 0) {
+ if (s1->max_delay > 0) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+ int frame_size = av_get_audio_frame_duration(st->codec, 0);
+ if (!frame_size)
+ frame_size = st->codec->frame_size;
+ if (frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
- s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
+ s->max_frames_per_packet =
+ av_rescale_q_rnd(s1->max_delay,
+ AV_TIME_BASE_Q,
+ (AVRational){ frame_size, st->codec->sample_rate },
+ AV_ROUND_DOWN);
}
}
- if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
- s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
+ s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
}
}
- av_set_pts_info(st, 32, 1, 90000);
+ avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
+ case CODEC_ID_H264:
+ /* check for H.264 MP4 syntax */
+ if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
+ s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
+ }
+ break;
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
+ s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
+ s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
+ s->num_frames = 0;
+ goto defaultcase;
+ case CODEC_ID_VP8:
+ av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
+ "incompatible with the latest spec drafts.\n");
+ break;
+ case CODEC_ID_ADPCM_G722:
+ /* Due to a historical error, the clock rate for G722 in RTP is
+ * 8000, even if the sample rate is 16000. See RFC 3551. */
+ avpriv_set_pts_info(st, 32, 1, 8000);
+ break;
+ case CODEC_ID_ILBC:
+ if (st->codec->block_align != 38 && st->codec->block_align != 50) {
+ av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
+ goto fail;
+ }
+ if (!s->max_frames_per_packet)
+ s->max_frames_per_packet = 1;
+ s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
+ s->max_payload_size / st->codec->block_align);
+ goto defaultcase;
+ case CODEC_ID_AMR_NB:
+ case CODEC_ID_AMR_WB:
+ if (!s->max_frames_per_packet)
+ s->max_frames_per_packet = 12;
+ if (st->codec->codec_id == CODEC_ID_AMR_NB)
+ n = 31;
+ else
+ n = 61;
+ /* max_header_toc_size + the largest AMR payload must fit */
+ if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
+ av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
+ goto fail;
+ }
+ if (st->codec->channels != 1) {
+ av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
+ goto fail;
+ }
case CODEC_ID_AAC:
- s->read_buf_index = 0;
+ s->num_frames = 0;
default:
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
- av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+defaultcase:
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+ avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
+
+fail:
+ av_freep(&s->buf);
+ return AVERROR(EINVAL);
}
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
-#if defined(DEBUG)
- printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
-#endif
+ av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
- if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
s->last_rtcp_ntp_time = ntp_time;
- rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
+ rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
s1->streams[0]->time_base) + s->base_timestamp;
- 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_be32(s1->pb, ntp_time / 1000000);
- put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
- put_be32(s1->pb, rtp_ts);
- put_be32(s1->pb, s->packet_count);
- put_be32(s1->pb, s->octet_count);
- put_flush_packet(s1->pb);
+ avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, RTCP_SR);
+ avio_wb16(s1->pb, 6); /* length in words - 1 */
+ avio_wb32(s1->pb, s->ssrc);
+ avio_wb32(s1->pb, ntp_time / 1000000);
+ avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
+ avio_wb32(s1->pb, rtp_ts);
+ avio_wb32(s1->pb, s->packet_count);
+ avio_wb32(s1->pb, s->octet_count);
+ avio_flush(s1->pb);
}
/* send an rtp packet. sequence number is incremented, but the caller
must update the timestamp itself */
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
-#ifdef DEBUG
- printf("rtp_send_data size=%d\n", len);
-#endif
+ av_dlog(s1, "rtp_send_data size=%d\n", len);
/* build the RTP header */
- put_byte(s1->pb, (RTP_VERSION << 6));
- put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
- put_be16(s1->pb, s->seq);
- put_be32(s1->pb, s->timestamp);
- put_be32(s1->pb, s->ssrc);
+ avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
+ avio_wb16(s1->pb, s->seq);
+ avio_wb32(s1->pb, s->timestamp);
+ avio_wb32(s1->pb, s->ssrc);
- put_buffer(s1->pb, buf1, len);
- put_flush_packet(s1->pb);
+ avio_write(s1->pb, buf1, len);
+ avio_flush(s1->pb);
s->seq++;
s->octet_count += len;
/* 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)
+ const uint8_t *buf1, int size, int sample_size_bits)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
+ /* Calculate the number of bytes to get samples aligned on a byte border */
+ int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
- max_packet_size = (s->max_payload_size / sample_size) * sample_size;
- /* not needed, but who nows */
- if ((size % sample_size) != 0)
+ max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
+ /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
+ if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
av_abort();
n = 0;
while (size > 0) {
s->buf_ptr += len;
buf1 += len;
size -= len;
- s->timestamp = s->cur_timestamp + n / sample_size;
+ s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
}
-/* 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)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, count, max_packet_size;
max_packet_size = s->max_payload_size;
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, max_packet_size;
max_packet_size = s->max_payload_size;
static void rtp_send_mpegts_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, out_len;
while (size >= TS_PACKET_SIZE) {
}
}
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
+static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
+{
+ RTPMuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int frame_duration = av_get_audio_frame_duration(st->codec, 0);
+ int frame_size = st->codec->block_align;
+ int frames = size / frame_size;
+
+ while (frames > 0) {
+ int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
+
+ if (!s->num_frames) {
+ s->buf_ptr = s->buf;
+ s->timestamp = s->cur_timestamp;
+ }
+ memcpy(s->buf_ptr, buf, n * frame_size);
+ frames -= n;
+ s->num_frames += n;
+ s->buf_ptr += n * frame_size;
+ buf += n * frame_size;
+ s->cur_timestamp += n * frame_duration;
+
+ if (s->num_frames == s->max_frames_per_packet) {
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+ s->num_frames = 0;
+ }
+ }
+ return 0;
+}
+
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
- uint8_t *buf1= pkt->data;
-#ifdef DEBUG
- printf("%d: write len=%d\n", pkt->stream_index, size);
-#endif
+ av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
- /* 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 >= RTCP_SR_SIZE) &&
- (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
- rtcp_send_sr(s1, ntp_time());
+ if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
+ (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
+ !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
+ rtcp_send_sr(s1, ff_ntp_time());
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
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);
+ rtp_send_samples(s1, pkt->data, size, 8 * 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);
+ rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
+ break;
+ case CODEC_ID_ADPCM_G722:
+ /* The actual sample size is half a byte per sample, but since the
+ * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
+ * the correct parameter for send_samples_bits is 8 bits per stream
+ * clock. */
+ rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
+ break;
+ case CODEC_ID_ADPCM_G726:
+ rtp_send_samples(s1, pkt->data, size,
+ st->codec->bits_per_coded_sample * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
- rtp_send_mpegaudio(s1, buf1, size);
+ rtp_send_mpegaudio(s1, pkt->data, size);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
- ff_rtp_send_mpegvideo(s1, buf1, size);
+ ff_rtp_send_mpegvideo(s1, pkt->data, size);
break;
case CODEC_ID_AAC:
- ff_rtp_send_aac(s1, buf1, size);
+ if (s->flags & FF_RTP_FLAG_MP4A_LATM)
+ ff_rtp_send_latm(s1, pkt->data, size);
+ else
+ ff_rtp_send_aac(s1, pkt->data, size);
+ break;
+ case CODEC_ID_AMR_NB:
+ case CODEC_ID_AMR_WB:
+ ff_rtp_send_amr(s1, pkt->data, size);
break;
case CODEC_ID_MPEG2TS:
- rtp_send_mpegts_raw(s1, buf1, size);
+ rtp_send_mpegts_raw(s1, pkt->data, size);
break;
case CODEC_ID_H264:
- ff_rtp_send_h264(s1, buf1, size);
+ ff_rtp_send_h264(s1, pkt->data, size);
+ break;
+ case CODEC_ID_H263:
+ if (s->flags & FF_RTP_FLAG_RFC2190) {
+ int mb_info_size = 0;
+ const uint8_t *mb_info =
+ av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
+ &mb_info_size);
+ ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
+ break;
+ }
+ /* Fallthrough */
+ case CODEC_ID_H263P:
+ ff_rtp_send_h263(s1, pkt->data, size);
+ break;
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ ff_rtp_send_xiph(s1, pkt->data, size);
+ break;
+ case CODEC_ID_VP8:
+ ff_rtp_send_vp8(s1, pkt->data, size);
+ break;
+ case CODEC_ID_ILBC:
+ rtp_send_ilbc(s1, pkt->data, size);
break;
default:
/* better than nothing : send the codec raw data */
- rtp_send_raw(s1, buf1, size);
+ rtp_send_raw(s1, pkt->data, size);
break;
}
return 0;
}
-AVOutputFormat rtp_muxer = {
- "rtp",
- "RTP output format",
- NULL,
- NULL,
- sizeof(RTPDemuxContext),
- CODEC_ID_PCM_MULAW,
- CODEC_ID_NONE,
- rtp_write_header,
- rtp_write_packet,
+static int rtp_write_trailer(AVFormatContext *s1)
+{
+ RTPMuxContext *s = s1->priv_data;
+
+ av_freep(&s->buf);
+
+ return 0;
+}
+
+AVOutputFormat ff_rtp_muxer = {
+ .name = "rtp",
+ .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
+ .priv_data_size = sizeof(RTPMuxContext),
+ .audio_codec = CODEC_ID_PCM_MULAW,
+ .video_codec = CODEC_ID_MPEG4,
+ .write_header = rtp_write_header,
+ .write_packet = rtp_write_packet,
+ .write_trailer = rtp_write_trailer,
+ .priv_class = &rtp_muxer_class,
};
projects / ffmpeg / blobdiff