Mercurial > libavformat.hg
changeset 532:772247018ade libavformat
* experimental dynamic audio stream allocation for DV demuxer. This
should make Nathan Kurz and if I don't hear too much complaints
about it -- that's the way it will be from now on.
* updating regressions
author | romansh |
---|---|
date | Mon, 27 Sep 2004 22:53:27 +0000 |
parents | 2af447b91329 |
children | 59da52e5f5a5 |
files | dv.c |
diffstat | 1 files changed, 19 insertions(+), 32 deletions(-) [+] |
line wrap: on
line diff
--- a/dv.c Mon Sep 27 22:46:36 2004 +0000 +++ b/dv.c Mon Sep 27 22:53:27 2004 +0000 @@ -538,7 +538,7 @@ { const uint8_t* as_pack; const DVprofile* sys; - int freq, smpls, quant, i; + int freq, smpls, quant, i, ach; sys = dv_frame_profile(frame); as_pack = dv_extract_pack(frame, dv_audio_source); @@ -550,23 +550,24 @@ smpls = as_pack[1] & 0x3f; /* samples in this frame - min. samples */ freq = (as_pack[4] >> 3) & 0x07; /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */ quant = as_pack[4] & 0x07; /* 0 - 16bit linear, 1 - 12bit nonlinear */ - c->ach = (quant && freq == 2) ? 2 : 1; + ach = (quant && freq == 2) ? 2 : 1; - /* The second stereo channel could appear in IEC 61834 stream only */ - if (c->ach == 2 && !c->ast[1]) { - c->ast[1] = av_new_stream(c->fctx, 0); - if (c->ast[1]) { - av_set_pts_info(c->ast[1], 64, 1, 30000); - c->ast[1]->codec.codec_type = CODEC_TYPE_AUDIO; - c->ast[1]->codec.codec_id = CODEC_ID_PCM_S16LE; - } else - c->ach = 1; - } - for (i=0; i<c->ach; i++) { + /* Dynamic handling of the audio streams in DV */ + for (i=0; i<ach; i++) { + if (!c->ast[i]) { + c->ast[i] = av_new_stream(c->fctx, 0); + if (!c->ast[i]) + break; + av_set_pts_info(c->ast[i], 64, 1, 30000); + c->ast[i]->codec.codec_type = CODEC_TYPE_AUDIO; + c->ast[i]->codec.codec_id = CODEC_ID_PCM_S16LE; + } c->ast[i]->codec.sample_rate = dv_audio_frequency[freq]; c->ast[i]->codec.channels = 2; c->ast[i]->codec.bit_rate = 2 * dv_audio_frequency[freq] * 16; + c->ast[i]->start_time = 0; } + c->ach = i; return (sys->audio_min_samples[freq] + smpls) * 4; /* 2ch, 2bytes */; } @@ -732,14 +733,14 @@ return NULL; c->vst = av_new_stream(s, 0); - c->ast[0] = av_new_stream(s, 0); - if (!c->vst || !c->ast[0]) - goto fail; + if (!c->vst) { + av_free(c); + return NULL; + } av_set_pts_info(c->vst, 64, 1, 30000); - av_set_pts_info(c->ast[0], 64, 1, 30000); c->fctx = s; - c->ast[1] = NULL; + c->ast[0] = c->ast[1] = NULL; c->ach = 0; c->frames = 0; c->abytes = 0; @@ -751,21 +752,7 @@ c->vst->codec.bit_rate = 25000000; c->vst->start_time = 0; - c->ast[0]->codec.codec_type = CODEC_TYPE_AUDIO; - c->ast[0]->codec.codec_id = CODEC_ID_PCM_S16LE; - c->ast[0]->codec.sample_rate = 48000; - c->ast[0]->codec.channels = 2; - c->ast[0]->start_time = 0; - return c; - -fail: - if (c->vst) - av_free(c->vst); - if (c->ast[0]) - av_free(c->ast[0]); - av_free(c); - return NULL; } static void __destruct_pkt(struct AVPacket *pkt)