//#define DEBUG
static const AVOption options[] = {
- FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
+ 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 },
{ NULL },
};
case CODEC_ID_THEORA:
case CODEC_ID_VP8:
case CODEC_ID_ADPCM_G722:
+ case CODEC_ID_ADPCM_G726:
return 1;
default:
return 0;
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
- int max_packet_size, n;
+ 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);
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
- max_packet_size = s1->pb->max_packet_size;
- if (max_packet_size <= 12)
+ 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(max_packet_size);
+ }
+ s->buf = av_malloc(s1->packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
- s->max_payload_size = max_packet_size - 12;
+ s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
- if (s1->max_delay) {
+ if (s1->max_delay > 0) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- if (st->codec->frame_size == 0) {
+ 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 * (int64_t)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 == AVMEDIA_TYPE_VIDEO) {
/* 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)
{
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);
}
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))) {
+ 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, pkt->data, 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, pkt->data, 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 is 1 byte per stream clock. */
- rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
+ * 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:
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;
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