/*
- * Copyright (c) 2001-2003 The ffmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
+ *
+ * first version by Francois Revol (revol@free.fr)
+ * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
+ * by Mike Melanson (melanson@pcisys.net)
*
* This file is part of Libav.
*
*/
#include "avcodec.h"
-#include "get_bits.h"
#include "put_bits.h"
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
+#include "internal.h"
/**
* @file
* ADPCM encoders
- * First version by Francois Revol (revol@free.fr)
- * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
- * by Mike Melanson (melanson@pcisys.net)
- *
* See ADPCM decoder reference documents for codec information.
*/
avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
switch (avctx->codec->id) {
- case CODEC_ID_ADPCM_IMA_WAV:
+ case AV_CODEC_ID_ADPCM_IMA_WAV:
/* each 16 bits sample gives one nibble
and we have 4 bytes per channel overhead */
avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
have to buffer the samples :-( */
avctx->block_align = BLKSIZE;
break;
- case CODEC_ID_ADPCM_IMA_QT:
+ case AV_CODEC_ID_ADPCM_IMA_QT:
avctx->frame_size = 64;
avctx->block_align = 34 * avctx->channels;
break;
- case CODEC_ID_ADPCM_MS:
+ 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->block_align = BLKSIZE;
+ if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
+ goto error;
avctx->extradata_size = 32;
- extradata = avctx->extradata = av_malloc(avctx->extradata_size);
- if (!extradata)
- return AVERROR(ENOMEM);
+ extradata = avctx->extradata;
bytestream_put_le16(&extradata, avctx->frame_size);
bytestream_put_le16(&extradata, 7); /* wNumCoef */
for (i = 0; i < 7; i++) {
bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
}
break;
- case CODEC_ID_ADPCM_YAMAHA:
- avctx->frame_size = BLKSIZE * avctx->channels;
+ case AV_CODEC_ID_ADPCM_YAMAHA:
+ avctx->frame_size = BLKSIZE * 2 / avctx->channels;
avctx->block_align = BLKSIZE;
break;
- case CODEC_ID_ADPCM_SWF:
+ case AV_CODEC_ID_ADPCM_SWF:
if (avctx->sample_rate != 11025 &&
avctx->sample_rate != 22050 &&
avctx->sample_rate != 44100) {
goto error;
}
- if (!(avctx->coded_frame = avcodec_alloc_frame()))
- goto error;
-
return 0;
error:
av_freep(&s->paths);
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
- av_freep(&avctx->coded_frame);
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
}
-static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c,
- short sample)
+static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
+ int16_t sample)
{
int delta = sample - c->prev_sample;
int nibble = FFMIN(7, abs(delta) * 4 /
return nibble;
}
-static inline unsigned char adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
- short sample)
+static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
+ int16_t sample)
{
int delta = sample - c->prev_sample;
int mask, step = ff_adpcm_step_table[c->step_index];
return nibble;
}
-static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c,
- short sample)
+static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
+ int16_t sample)
{
int predictor, nibble, bias;
nibble = (nibble + bias) / c->idelta;
nibble = av_clip(nibble, -8, 7) & 0x0F;
- predictor += (signed)((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
+ predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
c->sample2 = c->sample1;
c->sample1 = av_clip_int16(predictor);
- c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
+ c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
if (c->idelta < 16)
c->idelta = 16;
return nibble;
}
-static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
- short sample)
+static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
+ int16_t sample)
{
int nibble, delta;
return nibble;
}
-static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
- uint8_t *dst, ADPCMChannelStatus *c, int n)
+static void adpcm_compress_trellis(AVCodecContext *avctx,
+ const int16_t *samples, uint8_t *dst,
+ ADPCMChannelStatus *c, int n, int stride)
{
//FIXME 6% faster if frontier is a compile-time constant
ADPCMEncodeContext *s = avctx->priv_data;
const int frontier = 1 << avctx->trellis;
- const int stride = avctx->channels;
const int version = avctx->codec->id;
TrellisPath *paths = s->paths, *p;
TrellisNode *node_buf = s->node_buf;
nodes[0]->step = c->step_index;
nodes[0]->sample1 = c->sample1;
nodes[0]->sample2 = c->sample2;
- if (version == CODEC_ID_ADPCM_IMA_WAV ||
- version == CODEC_ID_ADPCM_IMA_QT ||
- version == CODEC_ID_ADPCM_SWF)
+ if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
+ version == AV_CODEC_ID_ADPCM_IMA_QT ||
+ version == AV_CODEC_ID_ADPCM_SWF)
nodes[0]->sample1 = c->prev_sample;
- if (version == CODEC_ID_ADPCM_MS)
+ if (version == AV_CODEC_ID_ADPCM_MS)
nodes[0]->step = c->idelta;
- if (version == CODEC_ID_ADPCM_YAMAHA) {
+ if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
if (c->step == 0) {
nodes[0]->step = 127;
nodes[0]->sample1 = 0;
const int range = (j < frontier / 2) ? 1 : 0;
const int step = nodes[j]->step;
int nidx;
- if (version == CODEC_ID_ADPCM_MS) {
+ if (version == AV_CODEC_ID_ADPCM_MS) {
const int predictor = ((nodes[j]->sample1 * c->coeff1) +
(nodes[j]->sample2 * c->coeff2)) / 64;
const int div = (sample - predictor) / step;
STORE_NODE(ms, FFMAX(16,
(ff_adpcm_AdaptationTable[nibble] * step) >> 8));
}
- } else if (version == CODEC_ID_ADPCM_IMA_WAV ||
- version == CODEC_ID_ADPCM_IMA_QT ||
- version == CODEC_ID_ADPCM_SWF) {
+ } else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
+ version == AV_CODEC_ID_ADPCM_IMA_QT ||
+ version == AV_CODEC_ID_ADPCM_SWF) {
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
const int predictor = nodes[j]->sample1;\
const int div = (sample - predictor) * 4 / STEP_TABLE;\
}
LOOP_NODES(ima, ff_adpcm_step_table[step],
av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
- } else { //CODEC_ID_ADPCM_YAMAHA
+ } else { //AV_CODEC_ID_ADPCM_YAMAHA
LOOP_NODES(yamaha, step,
av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
127, 24567));
c->idelta = nodes[0]->step;
}
-static int adpcm_encode_frame(AVCodecContext *avctx,
- unsigned char *frame, int buf_size, void *data)
+static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
- int n, i, st;
- short *samples;
- unsigned char *dst;
+ int n, i, ch, st, pkt_size, ret;
+ const int16_t *samples;
+ int16_t **samples_p;
+ uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
uint8_t *buf;
- dst = frame;
- samples = (short *)data;
+ samples = (const int16_t *)frame->data[0];
+ samples_p = (int16_t **)frame->extended_data;
st = avctx->channels == 2;
- /* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
+
+ if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
+ pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
+ else
+ pkt_size = avctx->block_align;
+ if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ dst = avpkt->data;
switch(avctx->codec->id) {
- case CODEC_ID_ADPCM_IMA_WAV:
- n = avctx->frame_size / 8;
- c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
- /* c->status[0].step_index = 0;
- XXX: not sure how to init the state machine */
- bytestream_put_le16(&dst, c->status[0].prev_sample);
- *dst++ = (unsigned char)c->status[0].step_index;
- *dst++ = 0; /* unknown */
- samples++;
- if (avctx->channels == 2) {
- c->status[1].prev_sample = (signed short)samples[0];
- /* c->status[1].step_index = 0; */
- bytestream_put_le16(&dst, c->status[1].prev_sample);
- *dst++ = (unsigned char)c->status[1].step_index;
- *dst++ = 0;
- samples++;
+ case AV_CODEC_ID_ADPCM_IMA_WAV:
+ {
+ int blocks, j;
+
+ blocks = (frame->nb_samples - 1) / 8;
+
+ for (ch = 0; ch < avctx->channels; ch++) {
+ ADPCMChannelStatus *status = &c->status[ch];
+ status->prev_sample = samples_p[ch][0];
+ /* status->step_index = 0;
+ XXX: not sure how to init the state machine */
+ bytestream_put_le16(&dst, status->prev_sample);
+ *dst++ = status->step_index;
+ *dst++ = 0; /* unknown */
}
- /* stereo: 4 bytes (8 samples) for left,
- 4 bytes for right, 4 bytes left, ... */
+ /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */
if (avctx->trellis > 0) {
- FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 8, error);
- adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n * 8);
- if (avctx->channels == 2)
- adpcm_compress_trellis(avctx, samples + 1, buf + n * 8,
- &c->status[1], n * 8);
- for (i = 0; i < n; i++) {
- *dst++ = buf[8 * i + 0] | (buf[8 * i + 1] << 4);
- *dst++ = buf[8 * i + 2] | (buf[8 * i + 3] << 4);
- *dst++ = buf[8 * i + 4] | (buf[8 * i + 5] << 4);
- *dst++ = buf[8 * i + 6] | (buf[8 * i + 7] << 4);
- if (avctx->channels == 2) {
- uint8_t *buf1 = buf + n * 8;
- *dst++ = buf1[8 * i + 0] | (buf1[8 * i + 1] << 4);
- *dst++ = buf1[8 * i + 2] | (buf1[8 * i + 3] << 4);
- *dst++ = buf1[8 * i + 4] | (buf1[8 * i + 5] << 4);
- *dst++ = buf1[8 * i + 6] | (buf1[8 * i + 7] << 4);
+ FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
+ for (ch = 0; ch < avctx->channels; ch++) {
+ adpcm_compress_trellis(avctx, &samples_p[ch][1],
+ buf + ch * blocks * 8, &c->status[ch],
+ blocks * 8, 1);
+ }
+ for (i = 0; i < blocks; i++) {
+ for (ch = 0; ch < avctx->channels; ch++) {
+ uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
+ for (j = 0; j < 8; j += 2)
+ *dst++ = buf1[j] | (buf1[j + 1] << 4);
}
}
av_free(buf);
} else {
- for (; n > 0; n--) {
- *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels ]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
- /* right channel */
- if (avctx->channels == 2) {
- *dst = adpcm_ima_compress_sample(&c->status[1], samples[1 ]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[3 ]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[1], samples[5 ]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[7 ]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[1], samples[9 ]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
- *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
- *dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
+ for (i = 0; i < blocks; i++) {
+ for (ch = 0; ch < avctx->channels; ch++) {
+ ADPCMChannelStatus *status = &c->status[ch];
+ const int16_t *smp = &samples_p[ch][1 + i * 8];
+ for (j = 0; j < 8; j += 2) {
+ uint8_t v = adpcm_ima_compress_sample(status, smp[j ]);
+ v |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;
+ *dst++ = v;
+ }
}
- samples += 8 * avctx->channels;
}
}
break;
- case CODEC_ID_ADPCM_IMA_QT:
+ }
+ case AV_CODEC_ID_ADPCM_IMA_QT:
{
- int ch, i;
PutBitContext pb;
- init_put_bits(&pb, dst, buf_size * 8);
+ init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
- put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
- put_bits(&pb, 7, c->status[ch].step_index);
+ ADPCMChannelStatus *status = &c->status[ch];
+ put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
+ put_bits(&pb, 7, status->step_index);
if (avctx->trellis > 0) {
uint8_t buf[64];
- adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
+ adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
+ 64, 1);
for (i = 0; i < 64; i++)
put_bits(&pb, 4, buf[i ^ 1]);
+ status->prev_sample = status->predictor;
} else {
for (i = 0; i < 64; i += 2) {
int t1, t2;
- t1 = adpcm_ima_qt_compress_sample(&c->status[ch],
- samples[avctx->channels * (i + 0) + ch]);
- t2 = adpcm_ima_qt_compress_sample(&c->status[ch],
- samples[avctx->channels * (i + 1) + ch]);
+ t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
+ t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
put_bits(&pb, 4, t2);
put_bits(&pb, 4, t1);
}
}
flush_put_bits(&pb);
- dst += put_bits_count(&pb) >> 3;
break;
}
- case CODEC_ID_ADPCM_SWF:
+ case AV_CODEC_ID_ADPCM_SWF:
{
- int i;
PutBitContext pb;
- init_put_bits(&pb, dst, buf_size * 8);
+ init_put_bits(&pb, dst, pkt_size * 8);
- n = avctx->frame_size - 1;
+ n = frame->nb_samples - 1;
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
put_sbits(&pb, 16, samples[i]);
put_bits(&pb, 6, c->status[i].step_index);
- c->status[i].prev_sample = (signed short)samples[i];
+ c->status[i].prev_sample = samples[i];
}
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
- adpcm_compress_trellis(avctx, samples + 2, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
+ &c->status[0], n, avctx->channels);
if (avctx->channels == 2)
- adpcm_compress_trellis(avctx, samples + 3, buf + n,
- &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
+ buf + n, &c->status[1], n,
+ avctx->channels);
for (i = 0; i < n; i++) {
put_bits(&pb, 4, buf[i]);
if (avctx->channels == 2)
}
av_free(buf);
} else {
- for (i = 1; i < avctx->frame_size; i++) {
+ for (i = 1; i < frame->nb_samples; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
samples[avctx->channels * i]));
if (avctx->channels == 2)
}
}
flush_put_bits(&pb);
- dst += put_bits_count(&pb) >> 3;
break;
}
- case CODEC_ID_ADPCM_MS:
+ case AV_CODEC_ID_ADPCM_MS:
for (i = 0; i < avctx->channels; i++) {
int predictor = 0;
*dst++ = predictor;
bytestream_put_le16(&dst, c->status[i].sample2);
if (avctx->trellis > 0) {
- int n = avctx->block_align - 7 * avctx->channels;
+ n = avctx->block_align - 7 * avctx->channels;
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
if (avctx->channels == 1) {
- adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
+ avctx->channels);
for (i = 0; i < n; i += 2)
*dst++ = (buf[i] << 4) | buf[i + 1];
} else {
- adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
- adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples, buf,
+ &c->status[0], n, avctx->channels);
+ adpcm_compress_trellis(avctx, samples + 1, buf + n,
+ &c->status[1], n, avctx->channels);
for (i = 0; i < n; i++)
*dst++ = (buf[i] << 4) | buf[n + i];
}
}
}
break;
- case CODEC_ID_ADPCM_YAMAHA:
- n = avctx->frame_size / 2;
+ 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);
n *= 2;
if (avctx->channels == 1) {
- adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
+ avctx->channels);
for (i = 0; i < n; i += 2)
*dst++ = buf[i] | (buf[i + 1] << 4);
} else {
- adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
- adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
+ adpcm_compress_trellis(avctx, samples, buf,
+ &c->status[0], n, avctx->channels);
+ adpcm_compress_trellis(avctx, samples + 1, buf + n,
+ &c->status[1], n, avctx->channels);
for (i = 0; i < n; i++)
*dst++ = buf[i] | (buf[n + i] << 4);
}
default:
return AVERROR(EINVAL);
}
- return dst - frame;
+
+ avpkt->size = pkt_size;
+ *got_packet_ptr = 1;
+ return 0;
error:
return AVERROR(ENOMEM);
}
+static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+};
+
+static const enum AVSampleFormat sample_fmts_p[] = {
+ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
+};
-#define ADPCM_ENCODER(id_, name_, long_name_) \
+#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
AVCodec ff_ ## name_ ## _encoder = { \
.name = #name_, \
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
.type = AVMEDIA_TYPE_AUDIO, \
.id = id_, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
- .encode = adpcm_encode_frame, \
+ .encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, \
- AV_SAMPLE_FMT_NONE}, \
- .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+ .sample_fmts = sample_fmts_, \
}
-ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
-ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
-ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
-ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
-ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
+ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");