Mercurial > mplayer.hg
changeset 24589:9118be6575da
demux_audio.c: Fix timestamp handling
The code calculated the pts values of audio packets by adding the length
of the current packet to the pts of the previous one. The length of the
previous packet should be added instead. This broke WAV timestamps near
the end of the stream where a short packet occurs.
Change the code to store the pts of the next packet instead of the last
one. This fixes the WAV timestamps and allows some simplifications.
MP3 timestamps are not affected as packets are always treated as
constant decoded length, and FLAC timestamps still have worse problems
(FLAC is treated as as if it was constant bitrate even though it isn't).
Also store the timestamps as double instead of float.
author | uau |
---|---|
date | Mon, 24 Sep 2007 21:49:56 +0000 |
parents | 8eb1ef462d29 |
children | 2c238fa777ff |
files | libmpdemux/demux_audio.c |
diffstat | 1 files changed, 15 insertions(+), 14 deletions(-) [+] |
line wrap: on
line diff
--- a/libmpdemux/demux_audio.c Mon Sep 24 21:49:53 2007 +0000 +++ b/libmpdemux/demux_audio.c Mon Sep 24 21:49:56 2007 +0000 @@ -26,7 +26,7 @@ typedef struct da_priv { int frmt; - float last_pts; + double next_pts; } da_priv_t; //! rather arbitrary value for maximum length of wav-format headers @@ -521,7 +521,7 @@ priv = malloc(sizeof(da_priv_t)); priv->frmt = frmt; - priv->last_pts = -1; + priv->next_pts = 0; demuxer->priv = priv; demuxer->audio->id = 0; demuxer->audio->sh = sh_audio; @@ -570,6 +570,8 @@ if(s->eof) return 0; + double this_pts = priv->next_pts; + switch(priv->frmt) { case MP3 : while(1) { @@ -590,7 +592,7 @@ free_demux_packet(dp); return 0; } - priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + sh_audio->audio.dwScale/(float)sh_audio->samplerate; + priv->next_pts += sh_audio->audio.dwScale/(double)sh_audio->samplerate; break; } } break; @@ -606,14 +608,15 @@ l = (l + align - 1) / align * align; dp = new_demux_packet(l); l = stream_read(s,dp->buffer,l); - priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + l/(float)sh_audio->i_bps; + priv->next_pts += l/(double)sh_audio->i_bps; break; } case fLaC: { l = 65535; dp = new_demux_packet(l); l = stream_read(s,dp->buffer,l); - priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + l/(float)sh_audio->i_bps; + /* FLAC is not a constant-bitrate codec. These values will be wrong. */ + priv->next_pts += l/(double)sh_audio->i_bps; break; } default: @@ -622,7 +625,7 @@ } resize_demux_packet(dp, l); - dp->pts = priv->last_pts; + dp->pts = this_pts; ds_add_packet(ds, dp); return 1; } @@ -642,7 +645,7 @@ continue; } stream_skip(demuxer->stream,len-4); - priv->last_pts += sh->audio.dwScale/(float)sh->samplerate; + priv->next_pts += sh->audio.dwScale/(double)sh->samplerate; nf--; } } @@ -660,11 +663,11 @@ priv = demuxer->priv; if(priv->frmt == MP3 && hr_mp3_seek && !(flags & 2)) { - len = (flags & 1) ? rel_seek_secs - priv->last_pts : rel_seek_secs; + len = (flags & 1) ? rel_seek_secs - priv->next_pts : rel_seek_secs; if(len < 0) { stream_seek(s,demuxer->movi_start); - len = priv->last_pts + len; - priv->last_pts = 0; + len = priv->next_pts + len; + priv->next_pts = 0; } if(len > 0) high_res_mp3_seek(demuxer,len); @@ -682,15 +685,13 @@ } else if(pos < demuxer->movi_start) pos = demuxer->movi_start; - priv->last_pts = (pos-demuxer->movi_start)/(float)sh_audio->i_bps; + priv->next_pts = (pos-demuxer->movi_start)/(double)sh_audio->i_bps; switch(priv->frmt) { case WAV: pos -= (pos - demuxer->movi_start) % (sh_audio->wf->nBlockAlign ? sh_audio->wf->nBlockAlign : (sh_audio->channels * sh_audio->samplesize)); - // We need to decrease the pts by one step to make it the "last one" - priv->last_pts -= sh_audio->wf->nAvgBytesPerSec/(float)sh_audio->i_bps; break; } @@ -719,7 +720,7 @@ case DEMUXER_CTRL_GET_PERCENT_POS: if (audio_length<=0) return DEMUXER_CTRL_DONTKNOW; - *((int *)arg)=(int)( (priv->last_pts*100) / audio_length); + *((int *)arg)=(int)( (priv->next_pts*100) / audio_length); return DEMUXER_CTRL_OK; default: