X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=b17c4651b6bfe74728b58f1d21ae4c9403a040bc;hb=eea003814cc5afaea546a6d229690350bd7481af;hp=d173cf3eb3f5378f1d509ebf71e3a9b2992bf015;hpb=6774247a9d7d15033c2b80118c03cb0cb10027df;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index d173cf3eb3f..b17c4651b6b 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -32,8 +32,8 @@ 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 }, - { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM }, + { "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 }, { NULL }, }; @@ -46,35 +46,38 @@ static const AVClass rtp_muxer_class = { #define RTCP_SR_SIZE 28 -static int is_supported(enum CodecID id) +static int is_supported(enum AVCodecID id) { switch(id) { - case CODEC_ID_H263: - case CODEC_ID_H263P: - case CODEC_ID_H264: - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: - case CODEC_ID_MPEG4: - case CODEC_ID_AAC: - case CODEC_ID_MP2: - case CODEC_ID_MP3: - case CODEC_ID_PCM_ALAW: - case CODEC_ID_PCM_MULAW: - case CODEC_ID_PCM_S8: - case CODEC_ID_PCM_S16BE: - case CODEC_ID_PCM_S16LE: - case CODEC_ID_PCM_U16BE: - case CODEC_ID_PCM_U16LE: - case CODEC_ID_PCM_U8: - case CODEC_ID_MPEG2TS: - case CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: - case CODEC_ID_VORBIS: - case CODEC_ID_THEORA: - case CODEC_ID_VP8: - case CODEC_ID_ADPCM_G722: - case CODEC_ID_ADPCM_G726: - case CODEC_ID_ILBC: + case AV_CODEC_ID_H263: + case AV_CODEC_ID_H263P: + case AV_CODEC_ID_H264: + 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; @@ -152,43 +155,49 @@ static int rtp_write_header(AVFormatContext *s1) avpriv_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { - case CODEC_ID_MP2: - case CODEC_ID_MP3: + case AV_CODEC_ID_MP2: + case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; break; - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: break; - case CODEC_ID_MPEG2TS: + 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 CODEC_ID_H264: + 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 CODEC_ID_VORBIS: - case CODEC_ID_THEORA: + 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 CODEC_ID_VP8: - av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is " - "incompatible with the latest spec drafts.\n"); - break; - case CODEC_ID_ADPCM_G722: + 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 CODEC_ID_ILBC: + 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; @@ -198,11 +207,11 @@ static int rtp_write_header(AVFormatContext *s1) s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, s->max_payload_size / st->codec->block_align); goto defaultcase; - case CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: + 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 == CODEC_ID_AMR_NB) + if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) n = 31; else n = 61; @@ -215,7 +224,7 @@ static int rtp_write_header(AVFormatContext *s1) av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); goto fail; } - case CODEC_ID_AAC: + case AV_CODEC_ID_AAC: s->num_frames = 0; default: defaultcase: @@ -281,8 +290,8 @@ 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_bits) +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; @@ -292,7 +301,7 @@ static void rtp_send_samples(AVFormatContext *s1, 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(); + return AVERROR(EINVAL); n = 0; while (size > 0) { s->buf_ptr = s->buf; @@ -307,6 +316,7 @@ static void rtp_send_samples(AVFormatContext *s1, 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, @@ -457,54 +467,50 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) s->cur_timestamp = s->base_timestamp + pkt->pts; switch(st->codec->codec_id) { - case CODEC_ID_PCM_MULAW: - case CODEC_ID_PCM_ALAW: - case CODEC_ID_PCM_U8: - case CODEC_ID_PCM_S8: - 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, 16 * st->codec->channels); - break; - case CODEC_ID_ADPCM_G722: + 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. */ - 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: + 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 CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: ff_rtp_send_mpegvideo(s1, pkt->data, size); break; - case CODEC_ID_AAC: + 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 CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: + case AV_CODEC_ID_AMR_NB: + case AV_CODEC_ID_AMR_WB: ff_rtp_send_amr(s1, pkt->data, size); break; - case CODEC_ID_MPEG2TS: + case AV_CODEC_ID_MPEG2TS: rtp_send_mpegts_raw(s1, pkt->data, size); break; - case CODEC_ID_H264: + case AV_CODEC_ID_H264: ff_rtp_send_h264(s1, pkt->data, size); break; - case CODEC_ID_H263: + case AV_CODEC_ID_H263: if (s->flags & FF_RTP_FLAG_RFC2190) { int mb_info_size = 0; const uint8_t *mb_info = @@ -514,19 +520,30 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) break; } /* Fallthrough */ - case CODEC_ID_H263P: + case AV_CODEC_ID_H263P: ff_rtp_send_h263(s1, pkt->data, size); break; - case CODEC_ID_VORBIS: - case CODEC_ID_THEORA: + case AV_CODEC_ID_VORBIS: + case AV_CODEC_ID_THEORA: ff_rtp_send_xiph(s1, pkt->data, size); break; - case CODEC_ID_VP8: + case AV_CODEC_ID_VP8: ff_rtp_send_vp8(s1, pkt->data, size); break; - case CODEC_ID_ILBC: + 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); @@ -548,8 +565,8 @@ AVOutputFormat ff_rtp_muxer = { .name = "rtp", .long_name = NULL_IF_CONFIG_SMALL("RTP output"), .priv_data_size = sizeof(RTPMuxContext), - .audio_codec = CODEC_ID_PCM_MULAW, - .video_codec = CODEC_ID_MPEG4, + .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,