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rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
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>
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@ -149,33 +149,6 @@ static int rtp_write_header(AVFormatContext *s1)
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}
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s->max_payload_size = s1->packet_size - 12;
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s->max_frames_per_packet = 0;
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if (s1->max_delay > 0) {
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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int frame_size = av_get_audio_frame_duration(st->codec, 0);
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if (!frame_size)
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frame_size = st->codec->frame_size;
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if (frame_size == 0) {
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av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
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} else {
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s->max_frames_per_packet =
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av_rescale_q_rnd(s1->max_delay,
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AV_TIME_BASE_Q,
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(AVRational){ frame_size, st->codec->sample_rate },
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AV_ROUND_DOWN);
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}
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}
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if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
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/* FIXME: We should round down here... */
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if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
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s->max_frames_per_packet = av_rescale_q(s1->max_delay,
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(AVRational){1, 1000000},
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av_inv_q(st->avg_frame_rate));
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} else
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s->max_frames_per_packet = 1;
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}
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}
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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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@ -225,9 +198,7 @@ static int rtp_write_header(AVFormatContext *s1)
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break;
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case AV_CODEC_ID_VORBIS:
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case AV_CODEC_ID_THEORA:
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if (!s->max_frames_per_packet)
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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->max_frames_per_packet = 15;
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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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@ -249,15 +220,11 @@ static int rtp_write_header(AVFormatContext *s1)
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av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
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goto fail;
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}
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if (!s->max_frames_per_packet)
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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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s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
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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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s->max_frames_per_packet = 12;
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s->max_frames_per_packet = 50;
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if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
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n = 31;
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else
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@ -273,8 +240,7 @@ static int rtp_write_header(AVFormatContext *s1)
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}
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break;
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case AV_CODEC_ID_AAC:
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if (!s->max_frames_per_packet)
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s->max_frames_per_packet = 5;
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s->max_frames_per_packet = 50;
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break;
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default:
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break;
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@ -493,18 +459,23 @@ static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
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int frames = size / frame_size;
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while (frames > 0) {
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int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
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if (s->num_frames > 0 &&
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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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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
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s->num_frames = 0;
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}
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if (!s->num_frames) {
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s->buf_ptr = s->buf;
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s->timestamp = s->cur_timestamp;
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}
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memcpy(s->buf_ptr, buf, n * frame_size);
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frames -= n;
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s->num_frames += n;
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s->buf_ptr += n * frame_size;
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buf += n * frame_size;
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s->cur_timestamp += n * frame_duration;
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memcpy(s->buf_ptr, buf, frame_size);
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frames--;
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s->num_frames++;
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s->buf_ptr += frame_size;
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buf += frame_size;
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s->cur_timestamp += frame_duration;
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if (s->num_frames == s->max_frames_per_packet) {
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
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@ -27,6 +27,7 @@
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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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@ -41,7 +42,9 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
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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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(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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@ -30,6 +30,7 @@
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void ff_rtp_send_amr(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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int max_header_toc_size = 1 + s->max_frames_per_packet;
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uint8_t *p;
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int len;
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@ -38,7 +39,9 @@ void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)
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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 - 1 > s->max_payload_size)) {
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len + size - 1 > 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 header_size = s->num_frames + 1;
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p = s->buf + max_header_toc_size - header_size;
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if (p != s->buf)
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@ -32,6 +32,7 @@
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void ff_rtp_send_xiph(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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int max_pkt_size, xdt, frag;
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uint8_t *q;
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@ -77,8 +78,10 @@ void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
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assert(s->num_frames <= s->max_frames_per_packet);
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if (s->num_frames > 0 &&
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(remaining < 0 ||
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s->num_frames == s->max_frames_per_packet)) {
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// send previous packets now; no room for new data
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s->num_frames == s->max_frames_per_packet ||
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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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// send previous packets now; no room for new data, or too much delay
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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s->num_frames = 0;
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}
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