diff options
Diffstat (limited to 'libavformat/rtpenc.c')
| -rw-r--r-- | libavformat/rtpenc.c | 44 |
1 files changed, 22 insertions, 22 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 00b69f5765..ef51236ab3 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -97,8 +97,8 @@ static int rtp_write_header(AVFormatContext *s1) return AVERROR(EINVAL); } st = s1->streams[0]; - if (!is_supported(st->codec->codec_id)) { - av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id)); + if (!is_supported(st->codecpar->codec_id)) { + av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id)); return -1; } @@ -106,7 +106,7 @@ static int rtp_write_header(AVFormatContext *s1) 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 { @@ -152,13 +152,13 @@ static int rtp_write_header(AVFormatContext *s1) } s->max_payload_size = s1->packet_size - 12; - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); + 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); } s->buf_ptr = s->buf; - switch(st->codec->codec_id) { + switch(st->codecpar->codec_id) { case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; @@ -186,8 +186,8 @@ static int rtp_write_header(AVFormatContext *s1) 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: @@ -195,8 +195,8 @@ static int rtp_write_header(AVFormatContext *s1) * things simple and similar to the avcC/H264 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: @@ -209,7 +209,7 @@ static int rtp_write_header(AVFormatContext *s1) 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; } @@ -219,16 +219,16 @@ static int rtp_write_header(AVFormatContext *s1) 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; } - s->max_frames_per_packet = s->max_payload_size / st->codec->block_align; + 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: s->max_frames_per_packet = 50; - if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) + if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB) n = 31; else n = 61; @@ -237,7 +237,7 @@ static int rtp_write_header(AVFormatContext *s1) 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; } @@ -458,8 +458,8 @@ static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int 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) { @@ -509,26 +509,26 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) } 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); 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