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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

segafilm: Remove deplanarization hack

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This commit is contained in:
Paul B Mahol 2015-06-23 13:48:08 +01:00 committed by Vittorio Giovara
parent 5a79bf0284
commit ded5957d75
2 changed files with 3 additions and 49 deletions

View File

@ -61,10 +61,6 @@ typedef struct FilmDemuxContext {
unsigned int base_clock;
unsigned int version;
/* buffer used for interleaving stereo PCM data */
unsigned char *stereo_buffer;
int stereo_buffer_size;
} FilmDemuxContext;
static int film_probe(AVProbeData *p)
@ -80,7 +76,6 @@ static int film_read_close(AVFormatContext *s)
FilmDemuxContext *film = s->priv_data;
av_freep(&film->sample_table);
av_freep(&film->stereo_buffer);
return 0;
}
@ -96,8 +91,6 @@ static int film_read_header(AVFormatContext *s)
unsigned int audio_frame_counter;
film->sample_table = NULL;
film->stereo_buffer = NULL;
film->stereo_buffer_size = 0;
/* load the main FILM header */
if (avio_read(pb, scratch, 16) != 16)
@ -131,9 +124,9 @@ static int film_read_header(AVFormatContext *s)
film->audio_type = AV_CODEC_ID_ADPCM_ADX;
else if (film->audio_channels > 0) {
if (film->audio_bits == 8)
film->audio_type = AV_CODEC_ID_PCM_S8;
film->audio_type = AV_CODEC_ID_PCM_S8_PLANAR;
else if (film->audio_bits == 16)
film->audio_type = AV_CODEC_ID_PCM_S16BE;
film->audio_type = AV_CODEC_ID_PCM_S16BE_PLANAR;
else
film->audio_type = AV_CODEC_ID_NONE;
} else
@ -265,8 +258,6 @@ static int film_read_packet(AVFormatContext *s,
AVIOContext *pb = s->pb;
film_sample *sample;
int ret = 0;
int i;
int left, right;
if (film->current_sample >= film->sample_count)
return AVERROR(EIO);
@ -283,43 +274,6 @@ static int film_read_packet(AVFormatContext *s,
if (av_new_packet(pkt, sample->sample_size))
return AVERROR(ENOMEM);
avio_read(pb, pkt->data, sample->sample_size);
} else if ((sample->stream == film->audio_stream_index) &&
(film->audio_channels == 2) &&
(film->audio_type != AV_CODEC_ID_ADPCM_ADX)) {
/* stereo PCM needs to be interleaved */
if (av_new_packet(pkt, sample->sample_size))
return AVERROR(ENOMEM);
/* make sure the interleave buffer is large enough */
if (sample->sample_size > film->stereo_buffer_size) {
av_free(film->stereo_buffer);
film->stereo_buffer_size = sample->sample_size;
film->stereo_buffer = av_malloc(film->stereo_buffer_size);
if (!film->stereo_buffer) {
film->stereo_buffer_size = 0;
return AVERROR(ENOMEM);
}
}
pkt->pos= avio_tell(pb);
ret = avio_read(pb, film->stereo_buffer, sample->sample_size);
if (ret != sample->sample_size)
ret = AVERROR(EIO);
left = 0;
right = sample->sample_size / 2;
for (i = 0; i < sample->sample_size; ) {
if (film->audio_bits == 8) {
pkt->data[i++] = film->stereo_buffer[left++];
pkt->data[i++] = film->stereo_buffer[right++];
} else {
pkt->data[i++] = film->stereo_buffer[left++];
pkt->data[i++] = film->stereo_buffer[left++];
pkt->data[i++] = film->stereo_buffer[right++];
pkt->data[i++] = film->stereo_buffer[right++];
}
}
} else {
ret= av_get_packet(pb, pkt, sample->sample_size);
if (ret != sample->sample_size)

View File

@ -31,7 +31,7 @@
#define LIBAVFORMAT_VERSION_MAJOR 56
#define LIBAVFORMAT_VERSION_MINOR 20
#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_MICRO 1
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \