#include "libavutil/intreadwrite.h"
#include "libavutil/time.h"
+#include "libavcodec/bytestream.h"
+
#include "avformat.h"
#include "network.h"
#include "srtp.h"
#include "url.h"
#include "rtpdec.h"
#include "rtpdec_formats.h"
+#include "internal.h"
#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
-static RTPDynamicProtocolHandler l24_dynamic_handler = {
+static const RTPDynamicProtocolHandler l24_dynamic_handler = {
.enc_name = "L24",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_PCM_S24BE,
};
-static RTPDynamicProtocolHandler gsm_dynamic_handler = {
+static const RTPDynamicProtocolHandler gsm_dynamic_handler = {
.enc_name = "GSM",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_GSM,
};
-static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
+static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_MP3ADU,
};
-static RTPDynamicProtocolHandler speex_dynamic_handler = {
+static const RTPDynamicProtocolHandler speex_dynamic_handler = {
.enc_name = "speex",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_SPEEX,
};
-static RTPDynamicProtocolHandler opus_dynamic_handler = {
+static const RTPDynamicProtocolHandler opus_dynamic_handler = {
.enc_name = "opus",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_OPUS,
};
-static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
+static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
.enc_name = "t140",
.codec_type = AVMEDIA_TYPE_SUBTITLE,
.codec_id = AV_CODEC_ID_TEXT,
};
-extern RTPDynamicProtocolHandler ff_rdt_video_handler;
-extern RTPDynamicProtocolHandler ff_rdt_audio_handler;
-extern RTPDynamicProtocolHandler ff_rdt_live_video_handler;
-extern RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_video_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_audio_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler;
+extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
-static const RTPDynamicProtocolHandler *rtp_dynamic_protocol_handler_list[] = {
+static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = {
/* rtp */
&ff_ac3_dynamic_handler,
&ff_amr_nb_dynamic_handler,
void ff_rtp_send_punch_packets(URLContext *rtp_handle)
{
- AVIOContext *pb;
- uint8_t *buf;
- int len;
+ uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf;
/* Send a small RTP packet */
- if (avio_open_dyn_buf(&pb) < 0)
- return;
- avio_w8(pb, (RTP_VERSION << 6));
- avio_w8(pb, 0); /* Payload type */
- avio_wb16(pb, 0); /* Seq */
- avio_wb32(pb, 0); /* Timestamp */
- avio_wb32(pb, 0); /* SSRC */
+ bytestream_put_byte(&ptr, (RTP_VERSION << 6));
+ bytestream_put_byte(&ptr, 0); /* Payload type */
+ bytestream_put_be16(&ptr, 0); /* Seq */
+ bytestream_put_be32(&ptr, 0); /* Timestamp */
+ bytestream_put_be32(&ptr, 0); /* SSRC */
- avio_flush(pb);
- len = avio_close_dyn_buf(pb, &buf);
- if ((len > 0) && buf)
- ffurl_write(rtp_handle, buf, len);
- av_free(buf);
+ ffurl_write(rtp_handle, buf, ptr - buf);
/* Send a minimal RTCP RR */
- if (avio_open_dyn_buf(&pb) < 0)
- return;
-
- avio_w8(pb, (RTP_VERSION << 6));
- avio_w8(pb, RTCP_RR); /* receiver report */
- avio_wb16(pb, 1); /* length in words - 1 */
- avio_wb32(pb, 0); /* our own SSRC */
+ ptr = buf;
+ bytestream_put_byte(&ptr, (RTP_VERSION << 6));
+ bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */
+ bytestream_put_be16(&ptr, 1); /* length in words - 1 */
+ bytestream_put_be32(&ptr, 0); /* our own SSRC */
- avio_flush(pb);
- len = avio_close_dyn_buf(pb, &buf);
- if ((len > 0) && buf)
- ffurl_write(rtp_handle, buf, len);
- av_free(buf);
+ ffurl_write(rtp_handle, buf, ptr - buf);
}
static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
return 0;
}
+static int opus_write_extradata(AVCodecParameters *codecpar)
+{
+ uint8_t *bs;
+ int ret;
+
+ /* This function writes an extradata with a channel mapping family of 0.
+ * This mapping family only supports mono and stereo layouts. And RFC7587
+ * specifies that the number of channels in the SDP must be 2.
+ */
+ if (codecpar->channels > 2) {
+ return AVERROR_INVALIDDATA;
+ }
+
+ ret = ff_alloc_extradata(codecpar, 19);
+ if (ret < 0)
+ return ret;
+
+ bs = (uint8_t *)codecpar->extradata;
+
+ /* Opus magic */
+ bytestream_put_buffer(&bs, "OpusHead", 8);
+ /* Version */
+ bytestream_put_byte (&bs, 0x1);
+ /* Channel count */
+ bytestream_put_byte (&bs, codecpar->channels);
+ /* Pre skip */
+ bytestream_put_le16 (&bs, 0);
+ /* Input sample rate */
+ bytestream_put_le32 (&bs, 48000);
+ /* Output gain */
+ bytestream_put_le16 (&bs, 0x0);
+ /* Mapping family */
+ bytestream_put_byte (&bs, 0x0);
+
+ return 0;
+}
+
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG-2 TS streams.
int payload_type, int queue_size)
{
RTPDemuxContext *s;
+ int ret;
s = av_mallocz(sizeof(RTPDemuxContext));
if (!s)
if (st->codecpar->sample_rate == 8000)
st->codecpar->sample_rate = 16000;
break;
+ case AV_CODEC_ID_OPUS:
+ ret = opus_write_extradata(st->codecpar);
+ if (ret < 0) {
+ av_log(s1, AV_LOG_ERROR,
+ "Error creating opus extradata: %s\n",
+ av_err2str(ret));
+ av_free(s);
+ return NULL;
+ }
+ break;
default:
break;
}
s->srtp_enabled = 1;
}
+static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
+ int64_t rtcp_time, delta_timestamp, delta_time;
+
+ AVProducerReferenceTime *prft =
+ (AVProducerReferenceTime *) av_packet_new_side_data(
+ pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
+ if (!prft)
+ return AVERROR(ENOMEM);
+
+ rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US;
+ delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
+ delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
+
+ prft->wallclock = rtcp_time + delta_time;
+ prft->flags = 24;
+ return 0;
+}
+
/**
* This was the second switch in rtp_parse packet.
* Normalizes time, if required, sets stream_index, etc.
if (timestamp == RTP_NOTS_VALUE)
return;
+ if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+ if (rtp_set_prft(s, pkt, timestamp) < 0) {
+ av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
+ }
+ }
+
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
int64_t addend;
int delta_timestamp;
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
{
int ret;
- av_init_packet(pkt);
+ av_packet_unref(pkt);
pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
pkt->stream_index = stream_idx;
projects / ffmpeg / blobdiff