* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/opt.h"
+
#include "avcodec.h"
#include "put_bits.h"
#include "bytestream.h"
} TrellisNode;
typedef struct ADPCMEncodeContext {
+ AVClass *class;
+ int block_size;
+
ADPCMChannelStatus status[6];
TrellisPath *paths;
TrellisNode *node_buf;
#define FREEZE_INTERVAL 128
-static av_cold int adpcm_encode_close(AVCodecContext *avctx);
-
static av_cold int adpcm_encode_init(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
uint8_t *extradata;
int i;
- int ret = AVERROR(ENOMEM);
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
return AVERROR(EINVAL);
}
- if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
- av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
+ if (s->block_size & (s->block_size - 1)) {
+ av_log(avctx, AV_LOG_ERROR, "block size must be power of 2\n");
return AVERROR(EINVAL);
}
- if (avctx->trellis && avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_SSI) {
- /*
- * The current trellis implementation doesn't work for extended
- * runs of samples without periodic resets. Disallow it.
- */
- av_log(avctx, AV_LOG_ERROR, "trellis not supported\n");
- return AVERROR_PATCHWELCOME;
- }
-
if (avctx->trellis) {
- int frontier = 1 << avctx->trellis;
- int max_paths = frontier * FREEZE_INTERVAL;
- FF_ALLOC_OR_GOTO(avctx, s->paths,
- max_paths * sizeof(*s->paths), error);
- FF_ALLOC_OR_GOTO(avctx, s->node_buf,
- 2 * frontier * sizeof(*s->node_buf), error);
- FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
- 2 * frontier * sizeof(*s->nodep_buf), error);
- FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
- 65536 * sizeof(*s->trellis_hash), error);
+ int frontier, max_paths;
+
+ if ((unsigned)avctx->trellis > 16U) {
+ av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_SSI ||
+ avctx->codec->id == AV_CODEC_ID_ADPCM_IMA_APM ||
+ avctx->codec->id == AV_CODEC_ID_ADPCM_ARGO) {
+ /*
+ * The current trellis implementation doesn't work for extended
+ * runs of samples without periodic resets. Disallow it.
+ */
+ av_log(avctx, AV_LOG_ERROR, "trellis not supported\n");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ frontier = 1 << avctx->trellis;
+ max_paths = frontier * FREEZE_INTERVAL;
+ if (!FF_ALLOC_TYPED_ARRAY(s->paths, max_paths) ||
+ !FF_ALLOC_TYPED_ARRAY(s->node_buf, 2 * frontier) ||
+ !FF_ALLOC_TYPED_ARRAY(s->nodep_buf, 2 * frontier) ||
+ !FF_ALLOC_TYPED_ARRAY(s->trellis_hash, 65536))
+ return AVERROR(ENOMEM);
}
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
case AV_CODEC_ID_ADPCM_MS:
/* each 16 bits sample gives one nibble
and we have 7 bytes per channel overhead */
- avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
+ avctx->frame_size = (s->block_size - 7 * avctx->channels) * 2 / avctx->channels + 2;
avctx->bits_per_coded_sample = 4;
- avctx->block_align = BLKSIZE;
+ avctx->block_align = s->block_size;
if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
- goto error;
+ return AVERROR(ENOMEM);
avctx->extradata_size = 32;
extradata = avctx->extradata;
bytestream_put_le16(&extradata, avctx->frame_size);
avctx->sample_rate != 44100) {
av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
"22050 or 44100\n");
- ret = AVERROR(EINVAL);
- goto error;
+ return AVERROR(EINVAL);
}
avctx->frame_size = 512 * (avctx->sample_rate / 11025);
break;
avctx->frame_size = BLKSIZE * 2 / avctx->channels;
avctx->block_align = BLKSIZE;
break;
+ case AV_CODEC_ID_ADPCM_IMA_APM:
+ avctx->frame_size = BLKSIZE * 2 / avctx->channels;
+ avctx->block_align = BLKSIZE;
+
+ if (!(avctx->extradata = av_mallocz(28 + AV_INPUT_BUFFER_PADDING_SIZE)))
+ return AVERROR(ENOMEM);
+ avctx->extradata_size = 28;
+ break;
+ case AV_CODEC_ID_ADPCM_ARGO:
+ avctx->frame_size = 32;
+ avctx->block_align = 17 * avctx->channels;
+ break;
default:
- ret = AVERROR(EINVAL);
- goto error;
+ return AVERROR(EINVAL);
}
return 0;
-error:
- return ret;
}
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
c->idelta = nodes[0]->step;
}
+static inline int adpcm_argo_compress_nibble(const ADPCMChannelStatus *cs, int16_t s,
+ int shift, int flag)
+{
+ int nibble;
+
+ if (flag)
+ nibble = 4 * s - 8 * cs->sample1 + 4 * cs->sample2;
+ else
+ nibble = 4 * s - 4 * cs->sample1;
+
+ return (nibble >> shift) & 0x0F;
+}
+
+static int64_t adpcm_argo_compress_block(ADPCMChannelStatus *cs, PutBitContext *pb,
+ const int16_t *samples, int nsamples,
+ int shift, int flag)
+{
+ int64_t error = 0;
+
+ if (pb) {
+ put_bits(pb, 4, shift - 2);
+ put_bits(pb, 1, 0);
+ put_bits(pb, 1, !!flag);
+ put_bits(pb, 2, 0);
+ }
+
+ for (int n = 0; n < nsamples; n++) {
+ /* Compress the nibble, then expand it to see how much precision we've lost. */
+ int nibble = adpcm_argo_compress_nibble(cs, samples[n], shift, flag);
+ int16_t sample = ff_adpcm_argo_expand_nibble(cs, nibble, shift, flag);
+
+ error += abs(samples[n] - sample);
+
+ if (pb)
+ put_bits(pb, 4, nibble);
+ }
+
+ return error;
+}
+
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
- else if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_SSI)
+ else if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_SSI ||
+ avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_APM)
pkt_size = (frame->nb_samples * avctx->channels) / 2;
else
pkt_size = avctx->block_align;
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
if (avctx->trellis > 0) {
- FF_ALLOC_ARRAY_OR_GOTO(avctx, buf, avctx->channels, blocks * 8, error);
+ if (!FF_ALLOC_TYPED_ARRAY(buf, avctx->channels * blocks * 8))
+ return AVERROR(ENOMEM);
for (ch = 0; ch < avctx->channels; ch++) {
adpcm_compress_trellis(avctx, &samples_p[ch][1],
buf + ch * blocks * 8, &c->status[ch],
}
if (avctx->trellis > 0) {
- FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
+ if (!(buf = av_malloc(2 * n)))
+ return AVERROR(ENOMEM);
adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
&c->status[0], n, avctx->channels);
if (avctx->channels == 2)
if (avctx->trellis > 0) {
n = avctx->block_align - 7 * avctx->channels;
- FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
+ if (!(buf = av_malloc(2 * n)))
+ return AVERROR(ENOMEM);
if (avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
avctx->channels);
case AV_CODEC_ID_ADPCM_YAMAHA:
n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
- FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
+ if (!(buf = av_malloc(2 * n * 2)))
+ return AVERROR(ENOMEM);
n *= 2;
if (avctx->channels == 1) {
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
*dst++ = nibble;
}
break;
+ case AV_CODEC_ID_ADPCM_IMA_APM:
+ {
+ PutBitContext pb;
+ init_put_bits(&pb, dst, pkt_size);
+
+ av_assert0(avctx->trellis == 0);
+
+ for (n = frame->nb_samples / 2; n > 0; n--) {
+ for (ch = 0; ch < avctx->channels; ch++) {
+ put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, *samples++));
+ put_bits(&pb, 4, adpcm_ima_qt_compress_sample(c->status + ch, samples[st]));
+ }
+ samples += avctx->channels;
+ }
+
+ flush_put_bits(&pb);
+ break;
+ }
+ case AV_CODEC_ID_ADPCM_ARGO:
+ {
+ PutBitContext pb;
+ init_put_bits(&pb, dst, pkt_size);
+
+ av_assert0(frame->nb_samples == 32);
+
+ for (ch = 0; ch < avctx->channels; ch++) {
+ int64_t error = INT64_MAX, tmperr = INT64_MAX;
+ int shift = 2, flag = 0;
+ int saved1 = c->status[ch].sample1;
+ int saved2 = c->status[ch].sample2;
+
+ /* Find the optimal coefficients, bail early if we find a perfect result. */
+ for (int s = 2; s < 18 && tmperr != 0; s++) {
+ for (int f = 0; f < 2 && tmperr != 0; f++) {
+ c->status[ch].sample1 = saved1;
+ c->status[ch].sample2 = saved2;
+ tmperr = adpcm_argo_compress_block(c->status + ch, NULL, samples_p[ch],
+ frame->nb_samples, s, f);
+ if (tmperr < error) {
+ shift = s;
+ flag = f;
+ error = tmperr;
+ }
+ }
+ }
+
+ /* Now actually do the encode. */
+ c->status[ch].sample1 = saved1;
+ c->status[ch].sample2 = saved2;
+ adpcm_argo_compress_block(c->status + ch, &pb, samples_p[ch],
+ frame->nb_samples, shift, flag);
+ }
+
+ flush_put_bits(&pb);
+ break;
+ }
default:
return AVERROR(EINVAL);
}
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
-error:
- return AVERROR(ENOMEM);
}
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
};
+static const AVOption options[] = {
+ {
+ .name = "block_size",
+ .help = "set the block size",
+ .offset = offsetof(ADPCMEncodeContext, block_size),
+ .type = AV_OPT_TYPE_INT,
+ .default_val = {.i64 = BLKSIZE},
+ .min = 32,
+ .max = 8192, /* Is this a reasonable upper limit? */
+ .flags = AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+ },
+ { NULL }
+};
+
+static const AVClass adpcm_encoder_class = {
+ .class_name = "ADPCM Encoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
#define ADPCM_ENCODER(id_, name_, sample_fmts_, capabilities_, long_name_) \
AVCodec ff_ ## name_ ## _encoder = { \
.name = #name_, \
.sample_fmts = sample_fmts_, \
.capabilities = capabilities_, \
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP, \
+ .priv_class = &adpcm_encoder_class, \
}
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, 0, "ADPCM IMA QuickTime");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_ARGO, adpcm_argo, sample_fmts_p, 0, "ADPCM Argonaut Games");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_APM, adpcm_ima_apm, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Ubisoft APM");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, 0, "ADPCM IMA QuickTime");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_SSI, adpcm_ima_ssi, sample_fmts, AV_CODEC_CAP_SMALL_LAST_FRAME, "ADPCM IMA Simon & Schuster Interactive");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, 0, "ADPCM IMA WAV");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, 0, "ADPCM Microsoft");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, 0, "ADPCM Shockwave Flash");
-ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, 0, "ADPCM Yamaha");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, 0, "ADPCM IMA WAV");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, 0, "ADPCM Microsoft");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, 0, "ADPCM Shockwave Flash");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, 0, "ADPCM Yamaha");
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