Mercurial > mplayer.hg
view libmpdemux/demux_aac.c @ 24590:2c238fa777ff
ao_alsa: Fix get_space() return values larger than buffersize
After a buffer underrun the ALSA get_space() function sometimes returned
values larger than the ao had set in ao_data.buffersize. Fix this by
replacing the old check against MAX_OUTBURST by one against
ao_data.buffersize. There should be no need for the MAX_OUTBURST check;
the current MPlayer side should no longer have any constant limit on the
amount of data an ao can buffer or request at once.
The get_space() values larger than ao_data.buffersize triggered errors
in audio decoding causing the current attempt to fill audio buffers to
be aborted. I'm not sure how often that caused behavior noticeably worse
then an underrun already is.
author | uau |
---|---|
date | Mon, 24 Sep 2007 21:49:58 +0000 |
parents | 4d81dbdf46b9 |
children | d4fe6e23283e |
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line source
#include <stdio.h> #include <stdlib.h> #include <string.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "stream/stream.h" #include "demuxer.h" #include "parse_es.h" #include "stheader.h" #include "ms_hdr.h" typedef struct { uint8_t *buf; uint64_t size; /// amount of time of data packets pushed to demuxer->audio (in bytes) float time; /// amount of time elapsed based upon samples_per_frame/sample_rate (in milliseconds) float last_pts; /// last pts seen int bitrate; /// bitrate computed as size/time } aac_priv_t; /// \param srate (out) sample rate /// \param num (out) number of audio frames in this ADTS frame /// \return size of the ADTS frame in bytes /// aac_parse_frames needs a buffer at least 8 bytes long int aac_parse_frame(uint8_t *buf, int *srate, int *num) { int i = 0, sr, fl = 0, id; static int srates[] = {96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 0, 0, 0}; if((buf[i] != 0xFF) || ((buf[i+1] & 0xF6) != 0xF0)) return 0; id = (buf[i+1] >> 3) & 0x01; //id=1 mpeg2, 0: mpeg4 sr = (buf[i+2] >> 2) & 0x0F; if(sr > 11) return 0; *srate = srates[sr]; fl = ((buf[i+3] & 0x03) << 11) | (buf[i+4] << 3) | ((buf[i+5] >> 5) & 0x07); *num = (buf[i+6] & 0x02) + 1; return fl; } static int demux_aac_init(demuxer_t *demuxer) { aac_priv_t *priv; priv = calloc(1, sizeof(aac_priv_t)); if(!priv) return 0; priv->buf = (uint8_t*) malloc(8); if(!priv->buf) { free(priv); return 0; } demuxer->priv = priv; return 1; } static void demux_close_aac(demuxer_t *demuxer) { aac_priv_t *priv = (aac_priv_t *) demuxer->priv; if(!priv) return; if(priv->buf) free(priv->buf); free(demuxer->priv); return; } /// returns DEMUXER_TYPE_AAC if it finds 8 ADTS frames in 32768 bytes, 0 otherwise static int demux_aac_probe(demuxer_t *demuxer) { int cnt = 0, c, len, srate, num; off_t init, probed; aac_priv_t *priv; if(! demux_aac_init(demuxer)) { mp_msg(MSGT_DEMUX, MSGL_ERR, "COULDN'T INIT aac_demux, exit\n"); return 0; } priv = (aac_priv_t *) demuxer->priv; init = probed = stream_tell(demuxer->stream); while(probed-init <= 32768 && cnt < 8) { c = 0; while(c != 0xFF) { c = stream_read_char(demuxer->stream); if(c < 0) goto fail; } priv->buf[0] = 0xFF; if(stream_read(demuxer->stream, &(priv->buf[1]), 7) < 7) goto fail; len = aac_parse_frame(priv->buf, &srate, &num); if(len > 0) { cnt++; stream_skip(demuxer->stream, len - 8); } probed = stream_tell(demuxer->stream); } stream_seek(demuxer->stream, init); if(cnt < 8) goto fail; mp_msg(MSGT_DEMUX, MSGL_V, "demux_aac_probe, INIT: %"PRIu64", PROBED: %"PRIu64", cnt: %d\n", init, probed, cnt); return DEMUXER_TYPE_AAC; fail: mp_msg(MSGT_DEMUX, MSGL_V, "demux_aac_probe, failed to detect an AAC stream\n"); return 0; } static demuxer_t* demux_aac_open(demuxer_t *demuxer) { sh_audio_t *sh; sh = new_sh_audio(demuxer, 0); sh->ds = demuxer->audio; sh->format = mmioFOURCC('M', 'P', '4', 'A'); demuxer->audio->sh = sh; demuxer->filepos = stream_tell(demuxer->stream); return demuxer; } static int demux_aac_fill_buffer(demuxer_t *demuxer, demux_stream_t *ds) { aac_priv_t *priv = (aac_priv_t *) demuxer->priv; demux_packet_t *dp; int c1, c2, len, srate, num; float tm = 0; if(demuxer->stream->eof || (demuxer->movi_end && stream_tell(demuxer->stream) >= demuxer->movi_end)) return 0; while(! demuxer->stream->eof) { c1 = c2 = 0; while(c1 != 0xFF) { c1 = stream_read_char(demuxer->stream); if(c1 < 0) return 0; } c2 = stream_read_char(demuxer->stream); if(c2 < 0) return 0; if((c2 & 0xF6) != 0xF0) continue; priv->buf[0] = (unsigned char) c1; priv->buf[1] = (unsigned char) c2; if(stream_read(demuxer->stream, &(priv->buf[2]), 6) < 6) return 0; len = aac_parse_frame(priv->buf, &srate, &num); if(len > 0) { dp = new_demux_packet(len); if(! dp) { mp_msg(MSGT_DEMUX, MSGL_ERR, "fill_buffer, NEW_ADD_PACKET(%d)FAILED\n", len); return 0; } memcpy(dp->buffer, priv->buf, 8); stream_read(demuxer->stream, &(dp->buffer[8]), len-8); if(srate) tm = (float) (num * 1024.0/srate); priv->last_pts += tm; dp->pts = priv->last_pts; //fprintf(stderr, "\nPTS: %.3f\n", dp->pts); ds_add_packet(demuxer->audio, dp); priv->size += len; priv->time += tm; priv->bitrate = (int) (priv->size / priv->time); demuxer->filepos = stream_tell(demuxer->stream); return len; } else stream_skip(demuxer->stream, -6); } return 0; } //This is an almost verbatim copy of high_res_mp3_seek(), from demux_audio.c static void demux_aac_seek(demuxer_t *demuxer, float rel_seek_secs, float audio_delay, int flags) { aac_priv_t *priv = (aac_priv_t *) demuxer->priv; demux_stream_t *d_audio=demuxer->audio; sh_audio_t *sh_audio=d_audio->sh; float time; ds_free_packs(d_audio); time = (flags & 1) ? rel_seek_secs - priv->last_pts : rel_seek_secs; if(time < 0) { stream_seek(demuxer->stream, demuxer->movi_start); time = priv->last_pts + time; priv->last_pts = 0; } if(time > 0) { int len, nf, srate, num; nf = time * sh_audio->samplerate/1024; while(nf > 0) { if(stream_read(demuxer->stream,priv->buf, 8) < 8) break; len = aac_parse_frame(priv->buf, &srate, &num); if(len <= 0) { stream_skip(demuxer->stream, -7); continue; } stream_skip(demuxer->stream, len - 8); priv->last_pts += (float) (num*1024.0/srate); nf -= num; } } } demuxer_desc_t demuxer_desc_aac = { "AAC demuxer", "aac", "AAC", "Nico Sabbi", "Raw AAC files ", DEMUXER_TYPE_AAC, 0, // unsafe autodetect demux_aac_probe, demux_aac_fill_buffer, demux_aac_open, demux_close_aac, demux_aac_seek, NULL };