static int is_supported(enum AVCodecID id)
{
switch(id) {
+ case AV_CODEC_ID_H261:
case AV_CODEC_ID_H263:
case AV_CODEC_ID_H263P:
case AV_CODEC_ID_H264:
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
- int n;
+ int n, ret = AVERROR(EINVAL);
AVStream *st;
if (s1->nb_streams != 1) {
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);
+ if (!is_supported(st->codecpar->codec_id)) {
+ av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codecpar->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);
+ st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
s->payload_type = st->id;
} else {
}
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;
- }
+ if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
+ avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
+ } else {
+ avpriv_set_pts_info(st, 32, 1, 90000);
}
-
- avpriv_set_pts_info(st, 32, 1, 90000);
- switch(st->codec->codec_id) {
+ s->buf_ptr = s->buf;
+ switch(st->codecpar->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
+ avpriv_set_pts_info(st, 32, 1, 90000);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
- s->buf_ptr = s->buf;
+ break;
+ case AV_CODEC_ID_H261:
+ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
+ av_log(s, AV_LOG_ERROR,
+ "Packetizing H.261 is experimental and produces incorrect "
+ "packetization for cases where GOBs don't fit into packets "
+ "(even though most receivers may handle it just fine). "
+ "Please set -f_strict experimental in order to enable it.\n");
+ ret = AVERROR_EXPERIMENTAL;
+ goto fail;
+ }
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;
+ if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
+ s->nal_length_size = (st->codecpar->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
+ * things simple and similar to the avcC/H.264 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;
+ if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
+ s->nal_length_size = (st->codecpar->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;
+ s->max_frames_per_packet = 15;
+ break;
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) {
+ if (st->codecpar->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
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) {
+ if (st->codecpar->block_align != 38 && st->codecpar->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;
+ s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
+ break;
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)
+ s->max_frames_per_packet = 50;
+ if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
- if (st->codec->channels != 1) {
+ if (st->codecpar->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
+ break;
case AV_CODEC_ID_AAC:
- s->num_frames = 0;
+ s->max_frames_per_packet = 50;
+ break;
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;
}
fail:
av_freep(&s->buf);
- return AVERROR(EINVAL);
+ return ret;
}
/* send an rtcp sender report packet */
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
- av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
+ av_log(s1, AV_LOG_TRACE, "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},
{
RTPMuxContext *s = s1->priv_data;
- av_dlog(s1, "rtp_send_data size=%d\n", len);
+ av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
/* build the RTP header */
avio_w8(s1->pb, RTP_VERSION << 6);
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)
{
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 frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
+ int frame_size = st->codecpar->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 > 0 &&
+ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
+ s1->max_delay, AV_TIME_BASE_Q) >= 0) {
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+ s->num_frames = 0;
+ }
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;
+ memcpy(s->buf_ptr, buf, frame_size);
+ frames--;
+ s->num_frames++;
+ s->buf_ptr += frame_size;
+ buf += frame_size;
+ s->cur_timestamp += frame_duration;
if (s->num_frames == s->max_frames_per_packet) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
int rtcp_bytes;
int size= pkt->size;
- av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
+ av_log(s1, AV_LOG_TRACE, "%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;
}
s->cur_timestamp = s->base_timestamp + pkt->pts;
- switch(st->codec->codec_id) {
+ switch(st->codecpar->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);
+ return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->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);
+ return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->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);
+ return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
case AV_CODEC_ID_ADPCM_G726:
return rtp_send_samples(s1, pkt->data, size,
- st->codec->bits_per_coded_sample * st->codec->channels);
+ st->codecpar->bits_per_coded_sample * st->codecpar->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);
rtp_send_mpegts_raw(s1, pkt->data, size);
break;
case AV_CODEC_ID_H264:
- ff_rtp_send_h264(s1, pkt->data, size);
+ ff_rtp_send_h264_hevc(s1, pkt->data, size);
+ break;
+ case AV_CODEC_ID_H261:
+ ff_rtp_send_h261(s1, pkt->data, size);
break;
case AV_CODEC_ID_H263:
if (s->flags & FF_RTP_FLAG_RFC2190) {
ff_rtp_send_h263(s1, pkt->data, size);
break;
case AV_CODEC_ID_HEVC:
- ff_rtp_send_hevc(s1, pkt->data, size);
+ ff_rtp_send_h264_hevc(s1, pkt->data, size);
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
.write_packet = rtp_write_packet,
.write_trailer = rtp_write_trailer,
.priv_class = &rtp_muxer_class,
+ .flags = AVFMT_TS_NONSTRICT,
};
projects / ffmpeg / blobdiff