Mercurial > mplayer.hg
annotate stream/audio_in.h @ 27518:e54c9b7eb0d8
Revert bad changes to SSA/ASS subtitle packet format
The following commits are reverted partially or completely:
"a valid ASS line contains 9 ',' before actual text"
"demux_mkv: output correctly formated ASS packets"
"libass: add a new ass_process_data() to process demuxed subtitle packets"
These commits converted the internal representation of SSA/ASS
subtitle packets from the format used by Matroska to a custom format
where each packet has contents exactly matching one line in complete
SSA script files. AFAIK no files natively use such a format for muxed
subtitles. The stated reason for this change was to use a format that
could in principle be muxed into a maximal number of containers. SSA
subtitles do not have an implicit duration so both start time and
duration or end time need to be specified explicitly; the new format
moved timing information inside the codec packet data so it could be
muxed without modification into containers that can represent only
start time at the container level. However such a change is wrong from
the viewpoint of program architecture. Timing information belongs to
the demuxer level, but these commits moved not only the duration but
also the authoritative value of the start time to inside the codec
data. Additionally the new format lost the value of the Matroska
ReadOrder field which is used by MPlayer.
This commit changes the internal packet format back to that used by
Matroska and makes the internal Matroska demuxer output that format
again. Libavformat still outputs the "new" format; it could be
converted back to the Matroska format in demux_lavf.c, but I'm not
adding that code at least yet. The current lavf code has similar
problems as the reverted code in MPlayer, and it also currently fails
to provide any way to access the value of the ReadOrder field. I hope
that the lavf side will be improved; if it isn't conversion can be
added later. For now I'll make MPlayer default to the internal Matroska
demuxer instead of the lavf one in a separate commit.
author | uau |
---|---|
date | Mon, 08 Sep 2008 21:26:22 +0000 |
parents | 9d95dc936e66 |
children | 0f1b5b68af32 |
rev | line source |
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26029 | 1 #ifndef MPLAYER_AUDIO_IN_H |
2 #define MPLAYER_AUDIO_IN_H | |
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3 |
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4 #define AUDIO_IN_ALSA 1 |
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5 #define AUDIO_IN_OSS 2 |
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6 |
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7 #include "config.h" |
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8 |
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9 #ifdef CONFIG_ALSA |
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10 #include <alsa/asoundlib.h> |
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11 |
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12 typedef struct { |
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13 char *device; |
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14 |
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15 snd_pcm_t *handle; |
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16 snd_output_t *log; |
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17 int buffer_time, period_time, chunk_size; |
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18 size_t bits_per_sample, bits_per_frame; |
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19 } ai_alsa_t; |
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20 #endif |
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21 |
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22 #ifdef CONFIG_OSS_AUDIO |
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23 typedef struct { |
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24 char *device; |
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25 |
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26 int audio_fd; |
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27 } ai_oss_t; |
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28 #endif |
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29 |
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30 typedef struct |
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31 { |
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32 int type; |
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33 int setup; |
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34 |
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35 /* requested values */ |
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36 int req_channels; |
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37 int req_samplerate; |
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38 |
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39 /* real values read-only */ |
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40 int channels; |
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41 int samplerate; |
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42 int blocksize; |
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43 int bytes_per_sample; |
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44 int samplesize; |
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45 |
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46 #ifdef CONFIG_ALSA |
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47 ai_alsa_t alsa; |
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48 #endif |
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49 #ifdef CONFIG_OSS_AUDIO |
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50 ai_oss_t oss; |
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51 #endif |
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52 } audio_in_t; |
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53 |
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54 int audio_in_init(audio_in_t *ai, int type); |
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55 int audio_in_setup(audio_in_t *ai); |
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56 int audio_in_set_device(audio_in_t *ai, char *device); |
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57 int audio_in_set_samplerate(audio_in_t *ai, int rate); |
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58 int audio_in_set_channels(audio_in_t *ai, int channels); |
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59 int audio_in_uninit(audio_in_t *ai); |
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60 int audio_in_start_capture(audio_in_t *ai); |
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61 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer); |
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62 |
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63 #ifdef CONFIG_ALSA |
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64 int ai_alsa_setup(audio_in_t *ai); |
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65 int ai_alsa_init(audio_in_t *ai); |
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66 int ai_alsa_xrun(audio_in_t *ai); |
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67 #endif |
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68 |
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69 #ifdef CONFIG_OSS_AUDIO |
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70 int ai_oss_set_samplerate(audio_in_t *ai); |
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71 int ai_oss_set_channels(audio_in_t *ai); |
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72 int ai_oss_init(audio_in_t *ai); |
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73 #endif |
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74 |
26029 | 75 #endif /* MPLAYER_AUDIO_IN_H */ |