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https://gitee.com/openharmony/third_party_ffmpeg
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Merge commit 'f8c01257f93ceda3e03bc4e540a51022d1e2bff2'
* commit 'f8c01257f93ceda3e03bc4e540a51022d1e2bff2': rtpenc: Always do the default initialization regardless of codecs Merged-by: Michael Niedermayer <michaelni@gmx.at>
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commit
a2c4c30b5f
@ -179,11 +179,17 @@ static int rtp_write_header(AVFormatContext *s1)
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}
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}
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avpriv_set_pts_info(st, 32, 1, 90000);
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
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} else {
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avpriv_set_pts_info(st, 32, 1, 90000);
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}
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s->buf_ptr = s->buf;
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switch(st->codec->codec_id) {
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case AV_CODEC_ID_MP2:
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case AV_CODEC_ID_MP3:
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s->buf_ptr = s->buf + 4;
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avpriv_set_pts_info(st, 32, 1, 90000);
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break;
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case AV_CODEC_ID_MPEG1VIDEO:
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case AV_CODEC_ID_MPEG2VIDEO:
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@ -227,7 +233,7 @@ static int rtp_write_header(AVFormatContext *s1)
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s->max_frames_per_packet = 15;
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s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
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s->num_frames = 0;
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goto defaultcase;
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break;
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case AV_CODEC_ID_ADPCM_G722:
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/* Due to a historical error, the clock rate for G722 in RTP is
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* 8000, even if the sample rate is 16000. See RFC 3551. */
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@ -252,7 +258,7 @@ static int rtp_write_header(AVFormatContext *s1)
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s->max_frames_per_packet = 1;
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s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
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s->max_payload_size / st->codec->block_align);
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goto defaultcase;
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break;
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case AV_CODEC_ID_AMR_NB:
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case AV_CODEC_ID_AMR_WB:
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if (!s->max_frames_per_packet)
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@ -271,18 +277,13 @@ static int rtp_write_header(AVFormatContext *s1)
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goto fail;
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}
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s->num_frames = 0;
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goto defaultcase;
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break;
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case AV_CODEC_ID_AAC:
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s->num_frames = 0;
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if (!s->max_frames_per_packet)
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s->max_frames_per_packet = 5;
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goto defaultcase;
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break;
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default:
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defaultcase:
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
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}
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s->buf_ptr = s->buf;
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break;
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}
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