X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=78121ed3c5abaf4fbab1e265320f63d30214536f;hb=c9024a9fd7ea7980f876a65816da8da78dd9d88d;hp=7cedff382eba3f850a99c95fe6454911c206d14d;hpb=2912e87a6c9264d556734e2bf94a99c64cf9b102;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 7cedff382eb..78121ed3c5a 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -22,12 +22,27 @@ #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" //#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 }, + { 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 CodecID id) @@ -57,6 +72,7 @@ static int is_supported(enum CodecID id) case CODEC_ID_THEORA: case CODEC_ID_VP8: case CODEC_ID_ADPCM_G722: + case CODEC_ID_ADPCM_G726: return 1; default: return 0; @@ -66,11 +82,13 @@ static int is_supported(enum CodecID id) 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); @@ -78,10 +96,8 @@ static int rtp_write_header(AVFormatContext *s1) return -1; } - s->payload_type = ff_rtp_get_payload_type(st->codec); if (s->payload_type < 0) - s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); - + 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; @@ -93,22 +109,36 @@ static int rtp_write_header(AVFormatContext *s1) 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 * 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) { @@ -117,7 +147,7 @@ static int rtp_write_header(AVFormatContext *s1) } } - 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: @@ -153,7 +183,7 @@ static int rtp_write_header(AVFormatContext *s1) 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. */ - av_set_pts_info(st, 32, 1, 8000); + avpriv_set_pts_info(st, 32, 1, 8000); break; case CODEC_ID_AMR_NB: case CODEC_ID_AMR_WB: @@ -177,7 +207,7 @@ static int rtp_write_header(AVFormatContext *s1) default: defaultcase: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - av_set_pts_info(st, 32, 1, st->codec->sample_rate); + avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; break; @@ -235,14 +265,16 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) /* 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) { @@ -254,7 +286,7 @@ static void rtp_send_samples(AVFormatContext *s1, 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); } @@ -368,8 +400,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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; @@ -381,19 +414,24 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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: @@ -404,7 +442,10 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) ff_rtp_send_mpegvideo(s1, pkt->data, size); break; case CODEC_ID_AAC: - ff_rtp_send_aac(s1, pkt->data, 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: @@ -417,6 +458,15 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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; @@ -445,14 +495,13 @@ static int rtp_write_trailer(AVFormatContext *s1) } AVOutputFormat ff_rtp_muxer = { - "rtp", - NULL_IF_CONFIG_SMALL("RTP output format"), - NULL, - NULL, - sizeof(RTPMuxContext), - CODEC_ID_PCM_MULAW, - CODEC_ID_NONE, - rtp_write_header, - rtp_write_packet, - rtp_write_trailer, + .name = "rtp", + .long_name = NULL_IF_CONFIG_SMALL("RTP output format"), + .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, };