#include "avcodec.h"
#include "put_bits.h"
-#include "dsputil.h"
#include "internal.h"
#include "lpc.h"
#include "mathops.h"
+#include "alac_data.h"
#define DEFAULT_FRAME_SIZE 4096
-#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
#define ALAC_FRAME_FOOTER_SIZE 3
int max_coded_frame_size;
int write_sample_size;
int extra_bits;
- int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
+ int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
- AlacLPCContext lpc[MAX_CHANNELS];
+ AlacLPCContext lpc[2];
LPCContext lpc_ctx;
AVCodecContext *avctx;
} AlacEncodeContext;
-static void init_sample_buffers(AlacEncodeContext *s,
- uint8_t * const *samples)
+static void init_sample_buffers(AlacEncodeContext *s, int channels,
+ uint8_t const *samples[2])
{
int ch, 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++) { \
+ for (ch = 0; ch < channels; ch++) { \
int32_t *bptr = s->sample_buf[ch]; \
const type *sptr = (const type *)samples[ch]; \
for (i = 0; i < s->frame_size; i++) \
}
}
-static void write_frame_header(AlacEncodeContext *s)
+static void write_element_header(AlacEncodeContext *s,
+ enum AlacRawDataBlockType element,
+ int instance)
{
int encode_fs = 0;
if (s->frame_size < DEFAULT_FRAME_SIZE)
encode_fs = 1;
- 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, 3, element); // element type
+ put_bits(&s->pbctx, 4, instance); // element instance
+ put_bits(&s->pbctx, 12, 0); // unused header bits
put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
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
}
}
-static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
- uint8_t * const *samples)
+static void write_element(AlacEncodeContext *s,
+ enum AlacRawDataBlockType element, int instance,
+ const uint8_t *samples0, const uint8_t *samples1)
{
- int i, j;
+ uint8_t const *samples[2] = { samples0, samples1 };
+ int i, j, channels;
int prediction_type = 0;
PutBitContext *pb = &s->pbctx;
- init_put_bits(pb, avpkt->data, avpkt->size);
+ channels = element == TYPE_CPE ? 2 : 1;
if (s->verbatim) {
- write_frame_header(s);
+ write_element_header(s, element, instance);
/* samples are channel-interleaved in verbatim mode */
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;
+ int32_t const *samples_s32[2] = { (const int32_t *)samples0,
+ (const int32_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
- for (j = 0; j < s->avctx->channels; j++)
+ for (j = 0; j < 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;
+ int16_t const *samples_s16[2] = { (const int16_t *)samples0,
+ (const int16_t *)samples1 };
for (i = 0; i < s->frame_size; i++)
- for (j = 0; j < s->avctx->channels; j++)
+ for (j = 0; j < 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);
+ s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
+ channels - 1;
- if (s->avctx->channels == 2)
+ init_sample_buffers(s, channels, samples);
+ write_element_header(s, element, instance);
+
+ if (channels == 2)
alac_stereo_decorrelation(s);
+ else
+ s->interlacing_shift = s->interlacing_leftweight = 0;
put_bits(pb, 8, s->interlacing_shift);
put_bits(pb, 8, s->interlacing_leftweight);
- for (i = 0; i < s->avctx->channels; i++) {
+ for (i = 0; i < channels; i++) {
calc_predictor_params(s, i);
put_bits(pb, 4, prediction_type);
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++) {
+ for (j = 0; j < 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++) {
+ for (i = 0; i < channels; i++) {
alac_linear_predictor(s, i);
// TODO: determine when this will actually help. for now it's not used.
for (j = s->frame_size - 1; j > 0; j--)
s->predictor_buf[j] -= s->predictor_buf[j - 1];
}
-
alac_entropy_coder(s);
}
}
- put_bits(pb, 3, 7);
+}
+
+static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
+ uint8_t * const *samples)
+{
+ PutBitContext *pb = &s->pbctx;
+ const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
+ const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
+ int ch, element, sce, cpe;
+
+ init_put_bits(pb, avpkt->data, avpkt->size);
+
+ ch = element = sce = cpe = 0;
+ while (ch < s->avctx->channels) {
+ if (ch_elements[element] == TYPE_CPE) {
+ write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
+ samples[ch_map[ch + 1]]);
+ cpe++;
+ ch += 2;
+ } else {
+ write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
+ sce++;
+ ch++;
+ }
+ element++;
+ }
+
+ put_bits(pb, 3, TYPE_END);
flush_put_bits(pb);
+
return put_bits_count(pb) >> 3;
}
avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
- /* TODO: Correctly implement multi-channel ALAC.
- It is similar to multi-channel AAC, in that it has a series of
- single-channel (SCE), channel-pair (CPE), and LFE elements. */
- if (avctx->channels > 2) {
- av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n");
- 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");
goto error;
}
- avctx->coded_frame = avcodec_alloc_frame();
+ avctx->coded_frame = av_frame_alloc();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
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, frame->extended_data);
/* frame too large. use verbatim mode */
s->verbatim = 1;
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);
}
AVCodec ff_alac_encoder = {
.name = "alac",
+ .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ALAC,
.priv_data_size = sizeof(AlacEncodeContext),
.encode2 = alac_encode_frame,
.close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .channel_layouts = ff_alac_channel_layouts,
.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)"),
};
projects / ffmpeg / blobdiff