generic-library/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt

Browse Source
This commit is contained in:
Justin Ruggles
2012-08-25 09:13:50 -04:00
parent d9a13c2f56
commit 095be4fbf6
1 changed files with 42 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

72
libavcodec/adpcmenc.c
View File

@@ -271,12 +271,11 @@ static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
static void adpcm_compress_trellis(AVCodecContext *avctx,
const int16_t *samples, uint8_t *dst,
ADPCMChannelStatus *c, int n)
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;
@@ -480,11 +479,13 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
{
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;
samples = (const int16_t *)frame->data[0];
samples_p = (int16_t **)frame->extended_data;
st = avctx->channels == 2;
if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
@@ -506,7 +507,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
for (ch = 0; ch < avctx->channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
status->prev_sample = samples[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);
@@ -518,9 +519,9 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
for (ch = 0; ch < avctx->channels; ch++) {
adpcm_compress_trellis(avctx, &samples[avctx->channels + ch],
adpcm_compress_trellis(avctx, &samples_p[ch][1],
buf + ch * blocks * 8, &c->status[ch],
blocks * 8);
blocks * 8, 1);
}
for (i = 0; i < blocks; i++) {
for (ch = 0; ch < avctx->channels; ch++) {
@@ -534,10 +535,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
for (i = 0; i < blocks; i++) {
for (ch = 0; ch < avctx->channels; ch++) {
ADPCMChannelStatus *status = &c->status[ch];
const int16_t *smp = &samples[avctx->channels * (1 + i * 8) + ch];
const int16_t *smp = &samples_p[ch][1 + i * 8];
for (j = 0; j < 8; j += 2) {
*dst++ = adpcm_ima_compress_sample(status, smp[avctx->channels * j ]) |
(adpcm_ima_compress_sample(status, smp[avctx->channels * (j + 1)]) << 4);
*dst++ = adpcm_ima_compress_sample(status, smp[j ]) |
(adpcm_ima_compress_sample(status, smp[j + 1]) << 4);
}
}
}
@@ -550,20 +551,20 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
put_bits(&pb, 9, (c->status[ch].prev_sample & 0xFFFF) >> 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][1], buf, status,
64, 1);
for (i = 0; i < 64; i++)
put_bits(&pb, 4, buf[i ^ 1]);
} 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);
}
@@ -595,10 +596,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
&c->status[0], n);
&c->status[0], n, avctx->channels);
if (avctx->channels == 2)
adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
buf + n, &c->status[1], n);
buf + n, &c->status[1], n,
avctx->channels);
for (i = 0; i < n; i++) {
put_bits(&pb, 4, buf[i]);
if (avctx->channels == 2)
@@ -642,12 +644,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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];
}
@@ -667,12 +672,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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);
}
@@ -700,7 +708,11 @@ static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
};
#define ADPCM_ENCODER(id_, name_, long_name_) \
static const enum AVSampleFormat sample_fmts_p[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
};
#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
AVCodec ff_ ## name_ ## _encoder = { \
.name = #name_, \
.type = AVMEDIA_TYPE_AUDIO, \
@@ -709,12 +721,12 @@ AVCodec ff_ ## name_ ## _encoder = { \
.init = adpcm_encode_init, \
.encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
.sample_fmts = sample_fmts, \
.sample_fmts = sample_fmts_, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
}
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_ENCODER(AV_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");