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adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
This commit is contained in:
Justin Ruggles
parent
d9a13c2f56
commit
095be4fbf6
@ -271,12 +271,11 @@ static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
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static void adpcm_compress_trellis(AVCodecContext *avctx,
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const int16_t *samples, uint8_t *dst,
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ADPCMChannelStatus *c, int n)
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ADPCMChannelStatus *c, int n, int stride)
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{
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//FIXME 6% faster if frontier is a compile-time constant
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ADPCMEncodeContext *s = avctx->priv_data;
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const int frontier = 1 << avctx->trellis;
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const int stride = avctx->channels;
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const int version = avctx->codec->id;
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TrellisPath *paths = s->paths, *p;
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TrellisNode *node_buf = s->node_buf;
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@ -480,11 +479,13 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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{
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int n, i, ch, st, pkt_size, ret;
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const int16_t *samples;
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int16_t **samples_p;
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uint8_t *dst;
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ADPCMEncodeContext *c = avctx->priv_data;
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uint8_t *buf;
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samples = (const int16_t *)frame->data[0];
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samples_p = (int16_t **)frame->extended_data;
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st = avctx->channels == 2;
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if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)
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@ -506,7 +507,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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for (ch = 0; ch < avctx->channels; ch++) {
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ADPCMChannelStatus *status = &c->status[ch];
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status->prev_sample = samples[ch];
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status->prev_sample = samples_p[ch][0];
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/* status->step_index = 0;
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XXX: not sure how to init the state machine */
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bytestream_put_le16(&dst, status->prev_sample);
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@ -518,9 +519,9 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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if (avctx->trellis > 0) {
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FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
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for (ch = 0; ch < avctx->channels; ch++) {
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adpcm_compress_trellis(avctx, &samples[avctx->channels + ch],
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adpcm_compress_trellis(avctx, &samples_p[ch][1],
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buf + ch * blocks * 8, &c->status[ch],
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blocks * 8);
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blocks * 8, 1);
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}
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for (i = 0; i < blocks; i++) {
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for (ch = 0; ch < avctx->channels; ch++) {
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@ -534,10 +535,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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for (i = 0; i < blocks; i++) {
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for (ch = 0; ch < avctx->channels; ch++) {
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ADPCMChannelStatus *status = &c->status[ch];
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const int16_t *smp = &samples[avctx->channels * (1 + i * 8) + ch];
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const int16_t *smp = &samples_p[ch][1 + i * 8];
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for (j = 0; j < 8; j += 2) {
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*dst++ = adpcm_ima_compress_sample(status, smp[avctx->channels * j ]) |
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(adpcm_ima_compress_sample(status, smp[avctx->channels * (j + 1)]) << 4);
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*dst++ = adpcm_ima_compress_sample(status, smp[j ]) |
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(adpcm_ima_compress_sample(status, smp[j + 1]) << 4);
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}
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}
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}
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@ -550,20 +551,20 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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init_put_bits(&pb, dst, pkt_size * 8);
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for (ch = 0; ch < avctx->channels; ch++) {
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put_bits(&pb, 9, (c->status[ch].prev_sample & 0xFFFF) >> 7);
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put_bits(&pb, 7, c->status[ch].step_index);
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ADPCMChannelStatus *status = &c->status[ch];
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put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
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put_bits(&pb, 7, status->step_index);
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if (avctx->trellis > 0) {
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uint8_t buf[64];
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adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
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adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
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64, 1);
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for (i = 0; i < 64; i++)
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put_bits(&pb, 4, buf[i ^ 1]);
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} else {
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for (i = 0; i < 64; i += 2) {
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int t1, t2;
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t1 = adpcm_ima_qt_compress_sample(&c->status[ch],
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samples[avctx->channels * (i + 0) + ch]);
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t2 = adpcm_ima_qt_compress_sample(&c->status[ch],
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samples[avctx->channels * (i + 1) + ch]);
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t1 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i ]);
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t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);
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put_bits(&pb, 4, t2);
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put_bits(&pb, 4, t1);
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}
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@ -595,10 +596,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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if (avctx->trellis > 0) {
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FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
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adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
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&c->status[0], n);
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&c->status[0], n, avctx->channels);
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if (avctx->channels == 2)
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adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
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buf + n, &c->status[1], n);
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buf + n, &c->status[1], n,
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avctx->channels);
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for (i = 0; i < n; i++) {
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put_bits(&pb, 4, buf[i]);
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if (avctx->channels == 2)
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@ -642,12 +644,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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n = avctx->block_align - 7 * avctx->channels;
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FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
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if (avctx->channels == 1) {
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
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avctx->channels);
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for (i = 0; i < n; i += 2)
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*dst++ = (buf[i] << 4) | buf[i + 1];
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} else {
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
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adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
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adpcm_compress_trellis(avctx, samples, buf,
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&c->status[0], n, avctx->channels);
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adpcm_compress_trellis(avctx, samples + 1, buf + n,
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&c->status[1], n, avctx->channels);
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for (i = 0; i < n; i++)
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*dst++ = (buf[i] << 4) | buf[n + i];
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}
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@ -667,12 +672,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
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n *= 2;
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if (avctx->channels == 1) {
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n,
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avctx->channels);
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for (i = 0; i < n; i += 2)
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*dst++ = buf[i] | (buf[i + 1] << 4);
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} else {
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
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adpcm_compress_trellis(avctx, samples + 1, buf + n, &c->status[1], n);
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adpcm_compress_trellis(avctx, samples, buf,
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&c->status[0], n, avctx->channels);
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adpcm_compress_trellis(avctx, samples + 1, buf + n,
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&c->status[1], n, avctx->channels);
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for (i = 0; i < n; i++)
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*dst++ = buf[i] | (buf[n + i] << 4);
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}
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@ -700,7 +708,11 @@ static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
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};
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#define ADPCM_ENCODER(id_, name_, long_name_) \
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static const enum AVSampleFormat sample_fmts_p[] = {
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AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
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};
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#define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
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AVCodec ff_ ## name_ ## _encoder = { \
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.name = #name_, \
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.type = AVMEDIA_TYPE_AUDIO, \
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@ -709,12 +721,12 @@ AVCodec ff_ ## name_ ## _encoder = { \
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.init = adpcm_encode_init, \
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.encode2 = adpcm_encode_frame, \
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.close = adpcm_encode_close, \
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.sample_fmts = sample_fmts, \
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.sample_fmts = sample_fmts_, \
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.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
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}
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, sample_fmts_p, "ADPCM IMA QuickTime");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, sample_fmts, "ADPCM Microsoft");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, sample_fmts, "ADPCM Shockwave Flash");
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ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, sample_fmts, "ADPCM Yamaha");
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