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authorMichael Niedermayer <michaelni@gmx.at>2012-11-02 14:15:28 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-11-02 14:20:33 +0100
commit6788350281c418f0f395a8279eee82f7abe7c63b (patch)
tree69cd76f699eff929f5b13f76b42eabc7f25f9355 /libavcodec/flac.c
parent00aa7fa786e41b5fc8404732453869aa3c14e33a (diff)
parent50a65e7a540ce6747f81d6dbf6a602ad35be77ff (diff)
downloadffmpeg-6788350281c418f0f395a8279eee82f7abe7c63b.tar.gz
Merge commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff'
* commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff': (24 commits) vmdaudio: set channel layout twinvq: validate sample rate code twinvq: set channel layout twinvq: validate that channels is not <= 0 truespeech: set channel layout sipr: set channel layout shorten: validate that the channel count in the header is not <= 0 ra288dec: set channel layout ra144dec: set channel layout qdm2: remove unneeded checks for channel count qdm2: make sure channels is not <= 0 and set channel layout qcelpdec: set channel layout nellymoserdec: set channels to 1 libopencore-amr: set channel layout for amr-nb or if not set by the user libilbc: set channel layout dpcm: use AVCodecContext.channels instead of keeping a private copy imc: set channels to 1 instead of validating it gsmdec: always set channel layout and sample rate at initialization libgsmdec: always set channel layout and sample rate at initialization g726dec: do not validate sample rate ... Conflicts: libavcodec/dpcm.c libavcodec/qdm2.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/flac.c')
-rw-r--r--libavcodec/flac.c19
1 files changed, 19 insertions, 0 deletions
diff --git a/libavcodec/flac.c b/libavcodec/flac.c
index b07e4f8705..4c1bc58408 100644
--- a/libavcodec/flac.c
+++ b/libavcodec/flac.c
@@ -19,6 +19,7 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/audioconvert.h"
#include "libavutil/crc.h"
#include "libavutil/log.h"
#include "bytestream.h"
@@ -28,6 +29,15 @@
static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
+static const int64_t flac_channel_layouts[6] = {
+ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_QUAD,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_5POINT1
+};
+
static int64_t get_utf8(GetBitContext *gb)
{
int64_t val;
@@ -181,6 +191,14 @@ int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
return 1;
}
+void ff_flac_set_channel_layout(AVCodecContext *avctx)
+{
+ if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))
+ avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];
+ else
+ avctx->channel_layout = 0;
+}
+
void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
const uint8_t *buffer)
{
@@ -205,6 +223,7 @@ void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *
avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
avctx->bits_per_raw_sample = s->bps;
+ ff_flac_set_channel_layout(avctx);
s->samples = get_bits_longlong(&gb, 36);