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adc214e679
FFmpeg/libavformat/rtpenc.c
Martin Storsjö adc214e679 rtpenc: Avoid brittle switch fallthroughs 
Instead explicitly jump to the default case in the cases where
it is wanted, and avoid fallthrough between different codecs,
which could easily introduce bugs if people editing the code
aren't careful.

Signed-off-by: Martin Storsjö <martin@martin.st>
2014-12-18 23:11:37 +02:00

640 lines
21 KiB
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/*
* RTP output format
* Copyright (c) 2002 Fabrice Bellard
*
* 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
*/
#include "avformat.h"
#include "mpegts.h"
#include "internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/random_seed.h"
#include "libavutil/opt.h"
#include "rtpenc.h"
static const AVOption options[] = {
FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
{ "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
{ "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
{ "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, 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
static int is_supported(enum AVCodecID id)
{
switch(id) {
case AV_CODEC_ID_H263:
case AV_CODEC_ID_H263P:
case AV_CODEC_ID_H264:
case AV_CODEC_ID_HEVC:
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
case AV_CODEC_ID_MPEG4:
case AV_CODEC_ID_AAC:
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_MULAW:
case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_MPEG2TS:
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
case AV_CODEC_ID_VP8:
case AV_CODEC_ID_ADPCM_G722:
case AV_CODEC_ID_ADPCM_G726:
case AV_CODEC_ID_ILBC:
case AV_CODEC_ID_MJPEG:
case AV_CODEC_ID_SPEEX:
case AV_CODEC_ID_OPUS:
return 1;
default:
return 0;
}
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n;
AVStream *st;
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);
return -1;
}
if (s->payload_type < 0) {
/* Re-validate non-dynamic payload types */
if (st->id < RTP_PT_PRIVATE)
st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
s->payload_type = st->id;
} else {
/* private option takes priority */
st->id = s->payload_type;
}
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
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;
// Pick a random sequence start number, but in the lower end of the
// available range, so that any wraparound doesn't happen immediately.
// (Immediate wraparound would be an issue for SRTP.)
if (s->seq < 0)
s->seq = av_get_random_seed() & 0x0fff;
else
s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
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->buf = av_malloc(s1->packet_size);
if (!s->buf) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 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_q_rnd(s1->max_delay,
AV_TIME_BASE_Q,
(AVRational){ frame_size, st->codec->sample_rate },
AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
s->max_frames_per_packet = av_rescale_q(s1->max_delay,
(AVRational){1, 1000000},
av_inv_q(st->avg_frame_rate));
} else
s->max_frames_per_packet = 1;
}
}
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case AV_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 AV_CODEC_ID_HEVC:
/* Only check for the standardized hvcC version of extradata, keeping
* things simple and similar to the avcC/H264 case above, instead
* of trying to handle the pre-standardization versions (as in
* libavcodec/hevc.c). */
if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_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 AV_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 AV_CODEC_ID_OPUS:
if (st->codec->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_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 AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == AV_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;
}
s->num_frames = 0;
goto defaultcase;
case AV_CODEC_ID_AAC:
s->num_frames = 0;
goto defaultcase;
default:
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, int bye)
{
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
s->last_rtcp_ntp_time = ntp_time;
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
s1->streams[0]->time_base) + s->base_timestamp;
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_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
avio_wb32(s1->pb, rtp_ts);
avio_wb32(s1->pb, s->packet_count);
avio_wb32(s1->pb, s->octet_count);
if (s->cname) {
int len = FFMIN(strlen(s->cname), 255);
avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
avio_w8(s1->pb, RTCP_SDES);
avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
avio_wb32(s1->pb, s->ssrc);
avio_w8(s1->pb, 0x01); /* CNAME */
avio_w8(s1->pb, len);
avio_write(s1->pb, s->cname, len);
avio_w8(s1->pb, 0); /* END */
for (len = (7 + len) % 4; len % 4; len++)
avio_w8(s1->pb, 0);
}
if (bye) {
avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
avio_w8(s1->pb, RTCP_BYE);
avio_wb16(s1->pb, 1); /* length in words - 1 */
avio_wb32(s1->pb, s->ssrc);
}
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)
{
RTPMuxContext *s = s1->priv_data;
av_dlog(s1, "rtp_send_data size=%d\n", len);
/* build the RTP header */
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);
avio_write(s1->pb, buf1, len);
avio_flush(s1->pb);
s->seq = (s->seq + 1) & 0xffff;
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 int rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size_bits)
{
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 / 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)
return AVERROR(EINVAL);
n = 0;
while (size > 0) {
s->buf_ptr = s->buf;
len = FFMIN(max_packet_size, size);
/* copy data */
memcpy(s->buf_ptr, buf1, len);
s->buf_ptr += len;
buf1 += len;
size -= len;
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);
}
return 0;
}
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
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) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->buf_ptr = s->buf + 4;
}
}
if (s->buf_ptr == s->buf + 4) {
s->timestamp = s->cur_timestamp;
}
/* 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);
ff_rtp_send_data(s1, s->buf, len + 4, 0);
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;
}
}
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size;
max_packet_size = s->max_payload_size;
while (size > 0) {
len = max_packet_size;
if (len > size)
len = size;
s->timestamp = s->cur_timestamp;
ff_rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
}
}
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
int len, out_len;
s->timestamp = s->cur_timestamp;
while (size >= TS_PACKET_SIZE) {
len = s->max_payload_size - (s->buf_ptr - s->buf);
if (len > size)
len = size;
memcpy(s->buf_ptr, buf1, len);
buf1 += len;
size -= len;
s->buf_ptr += len;
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
ff_rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
}
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)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
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) &&
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
rtcp_send_sr(s1, ff_ntp_time(), 0);
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
s->cur_timestamp = s->base_timestamp + pkt->pts;
switch(st->codec->codec_id) {
case AV_CODEC_ID_PCM_MULAW:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_S8:
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
case AV_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. */
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_ADPCM_G726:
return rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
ff_rtp_send_mpegvideo(s1, pkt->data, size);
break;
case AV_CODEC_ID_AAC:
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 AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
ff_rtp_send_amr(s1, pkt->data, size);
break;
case AV_CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, pkt->data, size);
break;
case AV_CODEC_ID_H264:
ff_rtp_send_h264(s1, pkt->data, size);
break;
case AV_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 AV_CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size);
break;
case AV_CODEC_ID_HEVC:
ff_rtp_send_hevc(s1, pkt->data, size);
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
ff_rtp_send_xiph(s1, pkt->data, size);
break;
case AV_CODEC_ID_VP8:
ff_rtp_send_vp8(s1, pkt->data, size);
break;
case AV_CODEC_ID_ILBC:
rtp_send_ilbc(s1, pkt->data, size);
break;
case AV_CODEC_ID_MJPEG:
ff_rtp_send_jpeg(s1, pkt->data, size);
break;
case AV_CODEC_ID_OPUS:
if (size > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR,
"Packet size %d too large for max RTP payload size %d\n",
size, s->max_payload_size);
return AVERROR(EINVAL);
}
/* Intentional fallthrough */
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);
break;
}
return 0;
}
static int rtp_write_trailer(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
/* If the caller closes and recreates ->pb, this might actually
* be NULL here even if it was successfully allocated at the start. */
if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
rtcp_send_sr(s1, ff_ntp_time(), 1);
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 = AV_CODEC_ID_PCM_MULAW,
.video_codec = AV_CODEC_ID_MPEG4,
.write_header = rtp_write_header,
.write_packet = rtp_write_packet,
.write_trailer = rtp_write_trailer,
.priv_class = &rtp_muxer_class,
.flags = AVFMT_TS_NONSTRICT,
};
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