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https://gitee.com/openharmony/third_party_ffmpeg
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4f6cd883f0
Instead check the timestamps while muxing, to avoid buffering a too long timestamp range into one single packet. This makes the AMR and AAC packetization slightly less efficient, since we set a possibly unnecessarily high max_frames_per_packet. (These packetizers end up doing a memmove of the TOC bytes if sending a packet before max_frames_per_packet is achieved, and we end up setting max_frames_per_packet to a value that should be high enough for most uses.) All packetizers that use max_frames_per_packet now set it either to a default value, or to a value calculated based on other parameters, so none of them rely on the previous default setting. For iLBC, copy one frame at a time, to allow checking the timestamp range for each of them - basically doing potentially multiple loops to simplify the code instead of trying to calculate the number of frames to buffer while honoring s1->max_delay. This is in preparation for reducing the coupling between libavformat and libavcodec, by not having the muxers use the encoder field frame_size (which may not be available during e.g. stream copy). Signed-off-by: Martin Storsjö <martin@martin.st>
87 lines
2.7 KiB
C
87 lines
2.7 KiB
C
/*
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* copyright (c) 2007 Luca Abeni
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "rtpenc.h"
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void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
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{
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RTPMuxContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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const int max_au_headers_size = 2 + 2 * s->max_frames_per_packet;
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int len, max_packet_size = s->max_payload_size - max_au_headers_size;
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uint8_t *p;
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/* skip ADTS header, if present */
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if ((s1->streams[0]->codec->extradata_size) == 0) {
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size -= 7;
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buff += 7;
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}
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/* test if the packet must be sent */
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len = (s->buf_ptr - s->buf);
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if (s->num_frames &&
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(s->num_frames == s->max_frames_per_packet ||
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(len + size) > s->max_payload_size ||
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av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
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s1->max_delay, AV_TIME_BASE_Q) >= 0)) {
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int au_size = s->num_frames * 2;
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p = s->buf + max_au_headers_size - au_size - 2;
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if (p != s->buf) {
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memmove(p + 2, s->buf + 2, au_size);
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}
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/* Write the AU header size */
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AV_WB16(p, au_size * 8);
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ff_rtp_send_data(s1, p, s->buf_ptr - p, 1);
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s->num_frames = 0;
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}
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if (s->num_frames == 0) {
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s->buf_ptr = s->buf + max_au_headers_size;
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s->timestamp = s->cur_timestamp;
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}
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if (size <= max_packet_size) {
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p = s->buf + s->num_frames++ * 2 + 2;
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AV_WB16(p, size * 8);
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memcpy(s->buf_ptr, buff, size);
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s->buf_ptr += size;
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} else {
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int au_size = size;
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max_packet_size = s->max_payload_size - 4;
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p = s->buf;
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AV_WB16(p, 2 * 8);
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while (size > 0) {
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len = FFMIN(size, max_packet_size);
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AV_WB16(&p[2], au_size * 8);
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memcpy(p + 4, buff, len);
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ff_rtp_send_data(s1, p, len + 4, len == size);
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size -= len;
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buff += len;
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}
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}
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}
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