#include <limits.h>
-#include "libavutil/audioconvert.h"
+#include "libavutil/channel_layout.h"
#include "libavutil/crc.h"
#include "avcodec.h"
#include "internal.h"
int got_streaminfo; ///< indicates if the STREAMINFO has been read
int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
+ uint8_t *decoded_buffer;
+ unsigned int decoded_buffer_size;
FLACDSPContext dsp;
} FLACContext;
-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 void allocate_buffers(FLACContext *s);
+static int allocate_buffers(FLACContext *s);
static void flac_set_bps(FLACContext *s)
{
{
enum FLACExtradataFormat format;
uint8_t *streaminfo;
+ int ret;
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
/* initialize based on the demuxer-supplied streamdata header */
avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
- allocate_buffers(s);
+ ret = allocate_buffers(s);
+ if (ret < 0)
+ return ret;
flac_set_bps(s);
ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
- if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))
- avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];
-
return 0;
}
av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps);
}
-static void allocate_buffers(FLACContext *s)
+static int allocate_buffers(FLACContext *s)
{
- int i;
+ int buf_size;
assert(s->max_blocksize);
- for (i = 0; i < s->channels; i++) {
- s->decoded[i] = av_malloc(sizeof(int32_t)*s->max_blocksize);
- }
+ buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize,
+ AV_SAMPLE_FMT_S32P, 0);
+ if (buf_size < 0)
+ return buf_size;
+
+ av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size);
+ if (!s->decoded_buffer)
+ return AVERROR(ENOMEM);
+
+ return av_samples_fill_arrays((uint8_t **)s->decoded, NULL,
+ s->decoded_buffer, s->channels,
+ s->max_blocksize, AV_SAMPLE_FMT_S32P, 0);
}
/**
*/
static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
{
- int metadata_type, metadata_size;
+ int metadata_type, metadata_size, ret;
if (buf_size < FLAC_STREAMINFO_SIZE+8) {
/* need more data */
return AVERROR_INVALIDDATA;
}
avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
- allocate_buffers(s);
+ ret = allocate_buffers(s);
+ if (ret < 0)
+ return ret;
flac_set_bps(s);
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;
bps -= wasted;
}
if (bps > 32) {
- av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
- return -1;
+ av_log_missing_feature(s->avctx, "Decorrelated bit depth > 32", 0);
+ return AVERROR_PATCHWELCOME;
}
//FIXME use av_log2 for types
static int decode_frame(FLACContext *s)
{
- int i;
+ int i, ret;
GetBitContext *gb = &s->gb;
FLACFrameInfo fi;
return -1;
}
- if (s->channels && fi.channels != s->channels) {
- av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "
- "is not supported\n");
- return -1;
+ if (s->channels && fi.channels != s->channels && s->got_streaminfo) {
+ s->channels = s->avctx->channels = fi.channels;
+ ff_flac_set_channel_layout(s->avctx);
+ ret = allocate_buffers(s);
+ if (ret < 0)
+ return ret;
}
s->channels = s->avctx->channels = fi.channels;
+ if (!s->avctx->channel_layout)
+ ff_flac_set_channel_layout(s->avctx);
s->ch_mode = fi.ch_mode;
if (!s->bps && !fi.bps) {
" or frame header\n");
return -1;
}
- if (fi.samplerate == 0) {
+ if (fi.samplerate == 0)
fi.samplerate = s->samplerate;
- } else if (s->samplerate && fi.samplerate != s->samplerate) {
- av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n",
- s->samplerate, fi.samplerate);
- }
s->samplerate = s->avctx->sample_rate = fi.samplerate;
if (!s->got_streaminfo) {
- allocate_buffers(s);
+ ret = allocate_buffers(s);
+ if (ret < 0)
+ return ret;
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;
dump_headers(s->avctx, (FLACStreaminfo *)s);
/* get output buffer */
s->frame.nb_samples = s->blocksize;
- if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
static av_cold int flac_decode_close(AVCodecContext *avctx)
{
FLACContext *s = avctx->priv_data;
- int i;
- for (i = 0; i < s->channels; i++) {
- av_freep(&s->decoded[i]);
- }
+ av_freep(&s->decoded_buffer);
return 0;
}