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audio: confine demux_mkv audio PCM hack 

Let codec_tags.c do the messy mapping.

In theory we could simplify further by makign demux_mkv.c directly use
codec names instead of the MPlayer-inherited "internal FourCC" business,
but I'd rather not touch this - it would just break things.
This commit is contained in:
wm4 2014-09-24 23:20:19 +02:00
parent 9ac86d9e99
commit e977624d87
2 changed files with 28 additions and 56 deletions
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50
audio/decode/ad_lavc.c
View File

@ -71,50 +71,6 @@ const struct m_sub_options ad_lavc_conf = {
},
};
struct pcm_map
{
int tag;
const char *codecs[6]; // {any, 1byte, 2bytes, 3bytes, 4bytes, 8bytes}
};
// NOTE: these are needed to make rawaudio with demux_mkv work.
static const struct pcm_map tag_map[] = {
// Microsoft PCM
{0x0, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
{0x1, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
// MS PCM, Extended
{0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
// IEEE float
{0x3, {"pcm_f32le", [5] = "pcm_f64le"}},
// 'raw '
{0x20776172, {"pcm_s16be", [1] = "pcm_u8"}},
// 'twos', used by demux_mkv.c internally
{MKTAG('t', 'w', 'o', 's'),
{NULL, "pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"}},
{-1},
};
static const char *find_pcm_decoder(const struct pcm_map *map, int format,
int bits_per_sample)
{
int bytes = (bits_per_sample + 7) / 8;
if (bytes == 8)
bytes = 5; // 64 bit entry
for (int n = 0; map[n].tag != -1; n++) {
const struct pcm_map *entry = &map[n];
if (entry->tag == format) {
const char *dec = NULL;
if (bytes >= 1 && bytes <= 5)
dec = entry->codecs[bytes];
if (!dec)
dec = entry->codecs[0];
if (dec)
return dec;
}
}
return NULL;
}
static void set_data_from_avframe(struct dec_audio *da)
{
struct priv *priv = da->priv;
@ -170,11 +126,6 @@ static int init(struct dec_audio *da, const char *decoder)
struct priv *ctx = talloc_zero(NULL, struct priv);
da->priv = ctx;
if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
decoder = find_pcm_decoder(tag_map, sh->format,
sh_audio->wf->wBitsPerSample);
}
ctx->force_channel_map = sh_audio->force_channels;
lavc_codec = avcodec_find_decoder_by_name(decoder);
@ -336,7 +287,6 @@ static int decode_packet(struct dec_audio *da)
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
mp_add_decoder(list, "lavc", "pcm", "pcm", "Raw PCM");
}
const struct ad_functions ad_lavc = {

34
demux/codec_tags.c
View File

@ -97,12 +97,6 @@ static const struct mp_codec_tag mp_audio_codec_tags[] = {
{MKTAG('s', 'a', 'm', 'r'), "amr_nb"},
{MKTAG('s', 'a', 'w', 'b'), "amr_wb"},
{MKTAG('v', 'r', 'b', 's'), "vorbis"},
// Special cased in ad_lavc:
{0 , "pcm"},
{0x1 , "pcm"}, // lavf: pcm_s16le
{0x3 , "pcm"}, // lavf: pcm_f32le
{0xfffe , "pcm"},
{MKTAG('t', 'w', 'o', 's'), "pcm"},
// ------- internal mplayer FourCCs ------
{MKTAG('O', 'p', 'u', 's'), "opus"}, // demux_mkv.c
{MKTAG('a', 'L', 'a', 'C'), "alac"}, // demux_mkv.c
@ -358,6 +352,28 @@ static const char *lookup_tag(const struct mp_codec_tag *mp_table,
return id == AV_CODEC_ID_NONE ? NULL : mp_codec_from_av_codec_id(id);
}
static const char *pcm_le[] = {"pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"};
static const char *pcm_be[] = {"pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"};
static const char *map_audio_pcm_tag(uint32_t tag, int bits)
{
int bytes = (bits + 7) / 8;
switch (tag) {
case 0x0: // Microsoft PCM
case 0x1:
case 0xfffe: // MS PCM, Extended
return bytes >= 1 && bytes <= 4 ? pcm_le[bytes - 1] : NULL;
case 0x3: // IEEE float
return bits == 64 ? "pcm_f64le" : "pcm_f32le";
case 0x20776172:// 'raw '
return bits == 8 ? "pcm_u8" : "pcm_s16be";
case MKTAG('t', 'w', 'o', 's'): // demux_mkv.c internal
return bytes >= 1 && bytes <= 4 ? pcm_be[bytes - 1] : NULL;
default:
return NULL;
}
}
void mp_set_codec_from_tag(struct sh_stream *sh)
{
switch (sh->type) {
@ -370,6 +386,12 @@ void mp_set_codec_from_tag(struct sh_stream *sh)
sh->codec = lookup_tag(mp_audio_codec_tags,
avformat_get_riff_audio_tags(),
sh->format);
if (sh->audio && sh->audio->wf) {
const char *codec =
map_audio_pcm_tag(sh->format, sh->audio->wf->wBitsPerSample);
if (codec)
sh->codec = codec;
}
break;
default: ;
}