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ffmpeg.wasm-core
Commit 7c278d2a authored by Justin Ruggles's avatar Justin Ruggles
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alacenc: support 24-bit encoding

parent c9d0f450
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Showing with 75 additions and 26 deletions
libavcodec/alacenc.c
View file @ 7c278d2a
......@@ -27,7 +27,6 @@
#include "mathops.h"
#define DEFAULT_FRAME_SIZE 4096
#define DEFAULT_SAMPLE_SIZE 16
#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
......@@ -66,6 +65,7 @@ typedef struct AlacEncodeContext {
int max_prediction_order;
int max_coded_frame_size;
int write_sample_size;
int extra_bits;
int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
......@@ -78,16 +78,26 @@ typedef struct AlacEncodeContext {
} AlacEncodeContext;
static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples)
static void init_sample_buffers(AlacEncodeContext *s,
uint8_t * const *samples)
{
int ch, i;
for (ch = 0; ch < s->avctx->channels; ch++) {
int32_t *bptr = s->sample_buf[ch];
const int16_t *sptr = input_samples[ch];
for (i = 0; i < s->frame_size; i++)
bptr[i] = sptr[i];
}
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;
#define COPY_SAMPLES(type) do { \
for (ch = 0; ch < s->avctx->channels; ch++) { \
int32_t *bptr = s->sample_buf[ch]; \
const type *sptr = (const type *)samples[ch]; \
for (i = 0; i < s->frame_size; i++) \
bptr[i] = sptr[i] >> shift; \
} \
} while (0)
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
COPY_SAMPLES(int32_t);
else
COPY_SAMPLES(int16_t);
}
static void encode_scalar(AlacEncodeContext *s, int x,
......@@ -128,7 +138,7 @@ static void write_frame_header(AlacEncodeContext *s)
put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
if (encode_fs)
put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
......@@ -345,7 +355,8 @@ static void alac_entropy_coder(AlacEncodeContext *s)
}
}
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples)
static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
uint8_t * const *samples)
{
int i, j;
int prediction_type = 0;
......@@ -356,9 +367,20 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples)
if (s->verbatim) {
write_frame_header(s);
/* samples are channel-interleaved in verbatim mode */
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
put_sbits(pb, 16, samples[j][i]);
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
int shift = 32 - s->avctx->bits_per_raw_sample;
int32_t * const *samples_s32 = (int32_t * const *)samples;
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s32[j][i] >> shift);
} else {
int16_t * const *samples_s16 = (int16_t * const *)samples;
for (i = 0; i < s->frame_size; i++)
for (j = 0; j < s->avctx->channels; j++)
put_sbits(pb, s->avctx->bits_per_raw_sample,
samples_s16[j][i]);
}
} else {
init_sample_buffers(s, samples);
write_frame_header(s);
......@@ -381,6 +403,17 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples)
put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
}
// write extra bits if needed
if (s->extra_bits) {
uint32_t mask = (1 << s->extra_bits) - 1;
for (i = 0; i < s->frame_size; i++) {
for (j = 0; j < s->avctx->channels; j++) {
put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask);
s->sample_buf[j][i] >>= s->extra_bits;
}
}
}
// apply lpc and entropy coding to audio samples
for (i = 0; i < s->avctx->channels; i++) {
......@@ -433,6 +466,15 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
return AVERROR_PATCHWELCOME;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
if (avctx->bits_per_raw_sample != 24)
av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
avctx->bits_per_raw_sample = 24;
} else {
avctx->bits_per_raw_sample = 16;
s->extra_bits = 0;
}
// Set default compression level
if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
s->compression_level = 2;
......@@ -447,10 +489,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,
avctx->channels,
DEFAULT_SAMPLE_SIZE);
// FIXME: consider wasted_bytes
s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1;
avctx->bits_per_raw_sample);
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
......@@ -463,11 +502,11 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
AV_WB32(alac_extradata+12, avctx->frame_size);
AV_WB8 (alac_extradata+17, DEFAULT_SAMPLE_SIZE);
AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);
AV_WB8 (alac_extradata+21, avctx->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
AV_WB32(alac_extradata+28,
avctx->sample_rate * avctx->channels * DEFAULT_SAMPLE_SIZE); // average bitrate
avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate
AV_WB32(alac_extradata+32, avctx->sample_rate);
// Set relevant extradata fields
......@@ -536,13 +575,12 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
{
AlacEncodeContext *s = avctx->priv_data;
int out_bytes, max_frame_size, ret;
int16_t **samples = (int16_t **)frame->extended_data;
s->frame_size = frame->nb_samples;
if (frame->nb_samples < DEFAULT_FRAME_SIZE)
max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
DEFAULT_SAMPLE_SIZE);
avctx->bits_per_raw_sample);
else
max_frame_size = s->max_coded_frame_size;
......@@ -552,14 +590,24 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
/* use verbatim mode for compression_level 0 */
s->verbatim = !s->compression_level;
if (s->compression_level) {
s->verbatim = 0;
s->extra_bits = avctx->bits_per_raw_sample - 16;
} else {
s->verbatim = 1;
s->extra_bits = 0;
}
s->write_sample_size = avctx->bits_per_raw_sample - s->extra_bits +
avctx->channels - 1;
out_bytes = write_frame(s, avpkt, samples);
out_bytes = write_frame(s, avpkt, frame->extended_data);
if (out_bytes > max_frame_size) {
/* frame too large. use verbatim mode */
s->verbatim = 1;
out_bytes = write_frame(s, avpkt, samples);
s->extra_bits = 0;
s->write_sample_size = avctx->bits_per_raw_sample + avctx->channels - 1;
out_bytes = write_frame(s, avpkt, frame->extended_data);
}
avpkt->size = out_bytes;
......@@ -576,7 +624,8 @@ AVCodec ff_alac_encoder = {
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
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