audio encoders.
This is purely for clarity/documentation purposes.
Originally committed as revision 24481 to svn://svn.ffmpeg.org/ffmpeg/trunk
unsigned char *frame, int buf_size, void *data)
{
AC3EncodeContext *s = avctx->priv_data;
- int16_t *samples = data;
+ const int16_t *samples = data;
int i, j, k, v, ch;
int16_t input_samples[N];
int32_t mdct_coef[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
int ich = s->channel_map[ch];
/* fixed mdct to the six sub blocks & exponent computation */
for(i=0;i<NB_BLOCKS;i++) {
- int16_t *sptr;
+ const int16_t *sptr;
int sinc;
/* compute input samples */
} AlacEncodeContext;
-static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
+static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples)
{
int ch, i;
for(ch=0;ch<s->avctx->channels;ch++) {
- int16_t *sptr = input_samples + ch;
+ const int16_t *sptr = input_samples + ch;
for(i=0;i<s->avctx->frame_size;i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
if((s->compression_level == 0) || verbatim_flag) {
// Verbatim mode
- int16_t *samples = data;
+ const int16_t *samples = data;
write_frame_header(s, 1);
for(i=0; i<avctx->frame_size*avctx->channels; i++) {
put_sbits(pb, 16, *samples++);
/**
* Copy channel-interleaved input samples into separate subframes
*/
-static void copy_samples(FlacEncodeContext *s, int16_t *samples)
+static void copy_samples(FlacEncodeContext *s, const int16_t *samples)
{
int i, j, ch;
FlacFrame *frame;
flush_put_bits(&s->pb);
}
-static void update_md5_sum(FlacEncodeContext *s, int16_t *samples)
+static void update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
{
#if HAVE_BIGENDIAN
int i;
av_md5_update(s->md5ctx, (uint8_t *)&smp, 2);
}
#else
- av_md5_update(s->md5ctx, (uint8_t *)samples, s->frame.blocksize*s->channels*2);
+ av_md5_update(s->md5ctx, (const uint8_t *)samples, s->frame.blocksize*s->channels*2);
#endif
}
{
int ch;
FlacEncodeContext *s;
- int16_t *samples = data;
+ const int16_t *samples = data;
int out_bytes;
int reencoded=0;
uint8_t *dst, int buf_size, void *data)
{
G726Context *c = avctx->priv_data;
- short *samples = data;
+ const short *samples = data;
PutBitContext pb;
init_put_bits(&pb, dst, 1024*1024);
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
-static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
+static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
{
short *p, *q;
int sum, offset, i, j;
unsigned char *frame, int buf_size, void *data)
{
MpegAudioContext *s = avctx->priv_data;
- short *samples = data;
+ const short *samples = data;
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
int padding, i;
static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data)
{
NellyMoserEncodeContext *s = avctx->priv_data;
- int16_t *samples = data;
+ const int16_t *samples = data;
int i;
if (s->last_frame)
unsigned char *frame, int buf_size, void *data)
{
int n, sample_size, v;
- short *samples;
+ const short *samples;
unsigned char *dst;
- uint8_t *srcu8;
- int16_t *samples_int16_t;
- int32_t *samples_int32_t;
- int64_t *samples_int64_t;
- uint16_t *samples_uint16_t;
- uint32_t *samples_uint32_t;
+ const uint8_t *srcu8;
+ const int16_t *samples_int16_t;
+ const int32_t *samples_int32_t;
+ const int64_t *samples_int64_t;
+ const uint16_t *samples_uint16_t;
+ const uint32_t *samples_uint32_t;
sample_size = av_get_bits_per_sample(avctx->codec->id)/8;
n = buf_size / sample_size;
unsigned char *frame, int buf_size, void *data)
{
int i, samples, stereo, ch;
- short *in;
+ const short *in;
unsigned char *out;
ROQDPCMContext *context = avctx->priv_data;
}
}
-static int apply_window_and_mdct(vorbis_enc_context *venc, signed short *audio,
+static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *audio,
int samples)
{
int i, j, channel;
int buf_size, void *data)
{
vorbis_enc_context *venc = avccontext->priv_data;
- signed short *audio = data;
+ const signed short *audio = data;
int samples = data ? avccontext->frame_size : 0;
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
}
-static void apply_window_and_mdct(AVCodecContext * avctx, signed short * audio, int len) {
+static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
WMACodecContext *s = avctx->priv_data;
int window_index= s->frame_len_bits - s->block_len_bits;
int i, j, channel;
static int encode_superframe(AVCodecContext *avctx,
unsigned char *buf, int buf_size, void *data){
WMACodecContext *s = avctx->priv_data;
- short *samples = data;
+ const short *samples = data;
int i, total_gain;
s->block_len_bits= s->frame_len_bits; //required by non variable block len