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Diffstat (limited to 'libavformat/rtpenc.c')
-rw-r--r--libavformat/rtpenc.c44
1 files changed, 22 insertions, 22 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 6158934900..c7733a6862 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 %x\n", st->codec->codec_id);
+ if (!is_supported(st->codecpar->codec_id)) {
+ av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", 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 {
@@ -149,13 +149,13 @@ 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;
@@ -183,8 +183,8 @@ 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:
@@ -192,8 +192,8 @@ 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:
@@ -206,7 +206,7 @@ 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;
}
@@ -216,16 +216,16 @@ 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;
@@ -234,7 +234,7 @@ 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;
}
@@ -454,8 +454,8 @@ 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) {
@@ -505,26 +505,26 @@ 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);
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-03 08:36:26 +0000