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-rw-r--r--libavcodec/8svx.c48
-rw-r--r--libavcodec/adpcm.c60
-rw-r--r--libavcodec/apedec.c39
-rw-r--r--libavcodec/atrac3.c44
-rw-r--r--libavcodec/cook.c40
-rw-r--r--libavcodec/dcadec.c128
-rw-r--r--libavcodec/imc.c26
-rw-r--r--libavcodec/mpc.c14
-rw-r--r--libavcodec/mpc.h2
-rw-r--r--libavcodec/mpc7.c6
-rw-r--r--libavcodec/mpc8.c7
-rw-r--r--libavcodec/ralf.c35
-rw-r--r--libavcodec/twinvq.c31
13 files changed, 210 insertions, 270 deletions
diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c
index f41b19fd58..0da48986f0 100644
--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -59,25 +59,6 @@ static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0,
#define MAX_FRAME_SIZE 2048
/**
- * Interleave samples in buffer containing all left channel samples
- * at the beginning, and right channel samples at the end.
- * Each sample is assumed to be in signed 8-bit format.
- *
- * @param size the size in bytes of the dst and src buffer
- */
-static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
-{
- uint8_t *dst_end = dst + size;
- size = size>>1;
-
- while (dst < dst_end) {
- *dst++ = *src;
- *dst++ = *(src+size);
- src++;
- }
-}
-
-/**
* Delta decode the compressed values in src, and put the resulting
* decoded n samples in dst.
*
@@ -107,7 +88,8 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
- int n, out_data_size, ret;
+ int n, out_data_size;
+ int ch, ret;
uint8_t *src, *dst;
/* decode and interleave the first packet */
@@ -152,10 +134,7 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
deinterleaved_samples = avpkt->data;
}
- if (avctx->channels == 2)
- interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
- else
- memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
+ memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
av_freep(&p);
}
@@ -170,11 +149,14 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
- dst = esc->frame.data[0];
- src = esc->samples + esc->samples_idx;
- out_data_size = esc->frame.nb_samples * avctx->channels;
- for (n = out_data_size; n > 0; n--)
- *dst++ = *src++ + 128;
+ out_data_size = esc->frame.nb_samples;
+ for (ch = 0; ch<avctx->channels; ch++) {
+ dst = esc->frame.data[ch];
+ src = esc->samples + esc->samples_idx / avctx->channels + ch * esc->samples_size / avctx->channels;
+ for (n = out_data_size; n > 0; n--)
+ *dst++ = *src++ + 128;
+ }
+ out_data_size *= avctx->channels;
esc->samples_idx += out_data_size;
return esc->table ?
@@ -200,7 +182,7 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
return AVERROR_INVALIDDATA;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_U8;
+ avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
avcodec_get_frame_defaults(&esc->frame);
avctx->coded_frame = &esc->frame;
@@ -230,6 +212,8 @@ AVCodec ff_eightsvx_fib_decoder = {
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_EIGHTSVX_EXP_DECODER
@@ -243,6 +227,8 @@ AVCodec ff_eightsvx_exp_decoder = {
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_PCM_S8_PLANAR_DECODER
@@ -256,5 +242,7 @@ AVCodec ff_pcm_s8_planar_decoder = {
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_NONE },
};
#endif
diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c
index 852549638b..054fa0a73d 100644
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -1268,7 +1268,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
}
-#define ADPCM_DECODER(id_, name_, long_name_) \
+static const enum AVSampleFormat sample_fmts_s16[] = { AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE };
+
+#define ADPCM_DECODER(id_, sample_fmts_, name_, long_name_) \
AVCodec ff_ ## name_ ## _decoder = { \
.name = #name_, \
.type = AVMEDIA_TYPE_AUDIO, \
@@ -1278,33 +1281,34 @@ AVCodec ff_ ## name_ ## _decoder = { \
.decode = adpcm_decode_frame, \
.capabilities = CODEC_CAP_DR1, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+ .sample_fmts = sample_fmts_, \
}
/* Note: Do not forget to add new entries to the Makefile as well. */
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_CT, adpcm_ct, "ADPCM Creative Technology");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA, adpcm_ea, "ADPCM Electronic Arts");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1, "ADPCM Electronic Arts R1");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_APC, adpcm_ima_apc, "ADPCM IMA CRYO APC");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
-ADPCM_DECODER(AV_CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, sample_fmts_s16, adpcm_4xm, "ADPCM 4X Movie");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_CT, sample_fmts_s16, adpcm_ct, "ADPCM Creative Technology");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA, sample_fmts_s16, adpcm_ea, "ADPCM Electronic Arts");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_MAXIS_XA, sample_fmts_s16, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, sample_fmts_s16, adpcm_ea_r1, "ADPCM Electronic Arts R1");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, sample_fmts_s16, adpcm_ea_r2, "ADPCM Electronic Arts R2");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, sample_fmts_s16, adpcm_ea_r3, "ADPCM Electronic Arts R3");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, sample_fmts_s16, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_AMV, sample_fmts_s16, adpcm_ima_amv, "ADPCM IMA AMV");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_APC, sample_fmts_s16, adpcm_ima_apc, "ADPCM IMA CRYO APC");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK3, sample_fmts_s16, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK4, sample_fmts_s16, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_EACS, sample_fmts_s16, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_SEAD, sample_fmts_s16, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_ISS, sample_fmts_s16, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, sample_fmts_s16, adpcm_ima_qt, "ADPCM IMA QuickTime");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_SMJPEG, sample_fmts_s16, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, sample_fmts_s16, adpcm_ima_wav, "ADPCM IMA WAV");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, sample_fmts_s16, adpcm_ima_ws, "ADPCM IMA Westwood");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_MS, sample_fmts_s16, adpcm_ms, "ADPCM Microsoft");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_2, sample_fmts_s16, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_3, sample_fmts_s16, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_4, sample_fmts_s16, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_SWF, sample_fmts_s16, adpcm_swf, "ADPCM Shockwave Flash");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16, adpcm_thp, "ADPCM Nintendo Gamecube THP");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16, adpcm_xa, "ADPCM CDROM XA");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_YAMAHA, sample_fmts_s16, adpcm_yamaha, "ADPCM Yamaha");
diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c
index 21080902ee..cba3dc0354 100644
--- a/libavcodec/apedec.c
+++ b/libavcodec/apedec.c
@@ -196,13 +196,13 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
s->bps = avctx->bits_per_coded_sample;
switch (s->bps) {
case 8:
- avctx->sample_fmt = AV_SAMPLE_FMT_U8;
+ avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
break;
case 16:
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
case 24:
- avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
break;
default:
av_log_ask_for_sample(avctx, "Unsupported bits per coded sample %d\n",
@@ -830,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
uint8_t *sample8;
int16_t *sample16;
int32_t *sample24;
- int i, ret;
+ int i, ch, ret;
int blockstodecode;
int bytes_used = 0;
@@ -930,27 +930,24 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
switch (s->bps) {
case 8:
- sample8 = (uint8_t *)s->frame.data[0];
- for (i = 0; i < blockstodecode; i++) {
- *sample8++ = (s->decoded[0][i] + 0x80) & 0xff;
- if (s->channels == 2)
- *sample8++ = (s->decoded[1][i] + 0x80) & 0xff;
+ for (ch = 0; ch < s->channels; ch++) {
+ sample8 = (uint8_t *)s->frame.data[ch];
+ for (i = 0; i < blockstodecode; i++)
+ *sample8++ = (s->decoded[ch][i] + 0x80) & 0xff;
}
break;
case 16:
- sample16 = (int16_t *)s->frame.data[0];
- for (i = 0; i < blockstodecode; i++) {
- *sample16++ = s->decoded[0][i];
- if (s->channels == 2)
- *sample16++ = s->decoded[1][i];
+ for (ch = 0; ch < s->channels; ch++) {
+ sample16 = (int16_t *)s->frame.data[ch];
+ for (i = 0; i < blockstodecode; i++)
+ *sample16++ = s->decoded[ch][i];
}
break;
case 24:
- sample24 = (int32_t *)s->frame.data[0];
- for (i = 0; i < blockstodecode; i++) {
- *sample24++ = s->decoded[0][i] << 8;
- if (s->channels == 2)
- *sample24++ = s->decoded[1][i] << 8;
+ for (ch = 0; ch < s->channels; ch++) {
+ sample24 = (int32_t *)s->frame.data[ch];
+ for (i = 0; i < blockstodecode; i++)
+ *sample24++ = s->decoded[ch][i] << 8;
}
break;
}
@@ -995,5 +992,9 @@ AVCodec ff_ape_decoder = {
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
.flush = ape_flush,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_NONE },
.priv_class = &ape_decoder_class,
};
diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c
index 7d076bed99..b900882a7a 100644
--- a/libavcodec/atrac3.c
+++ b/libavcodec/atrac3.c
@@ -112,7 +112,6 @@ typedef struct {
//@}
//@{
/** data buffers */
- float *outSamples[2];
uint8_t* decoded_bytes_buffer;
float tempBuf[1070];
//@}
@@ -198,7 +197,7 @@ static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
}
-static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
+static av_cold int init_atrac3_transforms(ATRAC3Context *q) {
float enc_window[256];
int i;
@@ -214,7 +213,7 @@ static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
}
/* Initialize the MDCT transform. */
- return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0);
+ return ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768);
}
/**
@@ -227,7 +226,6 @@ static av_cold int atrac3_decode_close(AVCodecContext *avctx)
av_free(q->pUnits);
av_free(q->decoded_bytes_buffer);
- av_freep(&q->outSamples[0]);
ff_mdct_end(&q->mdct_ctx);
@@ -838,8 +836,6 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
ATRAC3Context *q = avctx->priv_data;
int result;
const uint8_t* databuf;
- float *samples_flt;
- int16_t *samples_s16;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
@@ -853,8 +849,6 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return result;
}
- samples_flt = (float *)q->frame.data[0];
- samples_s16 = (int16_t *)q->frame.data[0];
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
@@ -864,27 +858,13 @@ static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
databuf = buf;
}
- if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
- result = decodeFrame(q, databuf, &samples_flt);
- else
- result = decodeFrame(q, databuf, q->outSamples);
+ result = decodeFrame(q, databuf, (float **)q->frame.extended_data);
if (result != 0) {
av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
return result;
}
- /* interleave */
- if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
- q->fmt_conv.float_interleave(samples_flt,
- (const float **)q->outSamples,
- SAMPLES_PER_FRAME, 2);
- } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
- q->fmt_conv.float_to_int16_interleave(samples_s16,
- (const float **)q->outSamples,
- SAMPLES_PER_FRAME, q->channels);
- }
-
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
@@ -1006,12 +986,9 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
vlcs_initialized = 1;
}
- if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT)
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- else
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
- if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) {
+ if ((ret = init_atrac3_transforms(q))) {
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
av_freep(&q->decoded_bytes_buffer);
return ret;
@@ -1049,15 +1026,6 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
return AVERROR(ENOMEM);
}
- if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
- q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0]));
- q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME;
- if (!q->outSamples[0]) {
- atrac3_decode_close(avctx);
- return AVERROR(ENOMEM);
- }
- }
-
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
@@ -1076,4 +1044,6 @@ AVCodec ff_atrac3_decoder =
.decode = atrac3_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index cca97e3046..b4d1a37ca1 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -119,9 +119,10 @@ typedef struct cook {
void (*interpolate)(struct cook *q, float *buffer,
int gain_index, int gain_index_next);
- void (*saturate_output)(struct cook *q, int chan, float *out);
+ void (*saturate_output)(struct cook *q, float *out);
AVCodecContext* avctx;
+ DSPContext dsp;
AVFrame frame;
GetBitContext gb;
/* stream data */
@@ -887,18 +888,15 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
* Saturate the output signal and interleave.
*
* @param q pointer to the COOKContext
- * @param chan channel to saturate
* @param out pointer to the output vector
*/
-static void saturate_output_float(COOKContext *q, int chan, float *out)
+static void saturate_output_float(COOKContext *q, float *out)
{
- int j;
- float *output = q->mono_mdct_output + q->samples_per_channel;
- for (j = 0; j < q->samples_per_channel; j++) {
- out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
- }
+ q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
+ -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
}
+
/**
* Final part of subpacket decoding:
* Apply modulated lapped transform, gain compensation,
@@ -909,15 +907,14 @@ static void saturate_output_float(COOKContext *q, int chan, float *out)
* @param gains_ptr array of current/prev gain pointers
* @param previous_buffer pointer to the previous buffer to be used for overlapping
* @param out pointer to the output buffer
- * @param chan 0: left or single channel, 1: right channel
*/
static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
cook_gains *gains_ptr, float *previous_buffer,
- float *out, int chan)
+ float *out)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
if (out)
- q->saturate_output(q, chan, out);
+ q->saturate_output(q, out);
}
@@ -930,7 +927,7 @@ static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
* @param outbuffer pointer to the outbuffer
*/
static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
- const uint8_t *inbuffer, float *outbuffer)
+ const uint8_t *inbuffer, float **outbuffer)
{
int sub_packet_size = p->size;
int res;
@@ -953,15 +950,18 @@ static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
}
mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
- p->mono_previous_buffer1, outbuffer, p->ch_idx);
+ p->mono_previous_buffer1,
+ outbuffer ? outbuffer[p->ch_idx] : NULL);
if (p->num_channels == 2)
if (p->joint_stereo)
mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
- p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
+ p->mono_previous_buffer2,
+ outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
else
mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
- p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
+ p->mono_previous_buffer2,
+ outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
return 0;
}
@@ -978,7 +978,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
- float *samples = NULL;
+ float **samples = NULL;
int i, ret;
int offset = 0;
int chidx = 0;
@@ -993,7 +993,7 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples = (float *) q->frame.data[0];
+ samples = (float **)q->frame.extended_data;
}
/* estimate subpacket sizes */
@@ -1110,6 +1110,8 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
/* Initialize RNG. */
av_lfg_init(&q->random_state, 0);
+ ff_dsputil_init(&q->dsp, avctx);
+
while (edata_ptr < edata_ptr_end) {
/* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */
@@ -1290,7 +1292,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
return AVERROR_PATCHWELCOME;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (channel_mask)
avctx->channel_layout = channel_mask;
else
@@ -1315,4 +1317,6 @@ AVCodec ff_cook_decoder = {
.decode = cook_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
index 4e9f957660..4d5e1152f2 100644
--- a/libavcodec/dcadec.c
+++ b/libavcodec/dcadec.c
@@ -417,11 +417,9 @@ typedef struct {
DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
- float scale_bias; ///< output scale
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
- const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
+ float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
@@ -1169,20 +1167,20 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
}
/* downmixing routines */
-#define MIX_REAR1(samples, si1, rs, coef) \
- samples[i] += samples[si1] * coef[rs][0]; \
- samples[i+256] += samples[si1] * coef[rs][1];
+#define MIX_REAR1(samples, s1, rs, coef) \
+ samples[0][i] += samples[s1][i] * coef[rs][0]; \
+ samples[1][i] += samples[s1][i] * coef[rs][1];
-#define MIX_REAR2(samples, si1, si2, rs, coef) \
- samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \
- samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];
+#define MIX_REAR2(samples, s1, s2, rs, coef) \
+ samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
+ samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
#define MIX_FRONT3(samples, coef) \
- t = samples[i + c]; \
- u = samples[i + l]; \
- v = samples[i + r]; \
- samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
- samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
+ t = samples[c][i]; \
+ u = samples[l][i]; \
+ v = samples[r][i]; \
+ samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
+ samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for (i = 0; i < 256; i++) { \
@@ -1190,7 +1188,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
op2 \
}
-static void dca_downmix(float *samples, int srcfmt,
+static void dca_downmix(float **samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
const int8_t *channel_mapping)
{
@@ -1215,36 +1213,36 @@ static void dca_downmix(float *samples, int srcfmt,
case DCA_STEREO:
break;
case DCA_3F:
- c = channel_mapping[0] * 256;
- l = channel_mapping[1] * 256;
- r = channel_mapping[2] * 256;
+ c = channel_mapping[0];
+ l = channel_mapping[1];
+ r = channel_mapping[2];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
break;
case DCA_2F1R:
- s = channel_mapping[2] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
+ s = channel_mapping[2];
+ DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
break;
case DCA_3F1R:
- c = channel_mapping[0] * 256;
- l = channel_mapping[1] * 256;
- r = channel_mapping[2] * 256;
- s = channel_mapping[3] * 256;
+ c = channel_mapping[0];
+ l = channel_mapping[1];
+ r = channel_mapping[2];
+ s = channel_mapping[3];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR1(samples, i + s, 3, coef));
+ MIX_REAR1(samples, s, 3, coef));
break;
case DCA_2F2R:
- sl = channel_mapping[2] * 256;
- sr = channel_mapping[3] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
+ sl = channel_mapping[2];
+ sr = channel_mapping[3];
+ DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
break;
case DCA_3F2R:
- c = channel_mapping[0] * 256;
- l = channel_mapping[1] * 256;
- r = channel_mapping[2] * 256;
- sl = channel_mapping[3] * 256;
- sr = channel_mapping[4] * 256;
+ c = channel_mapping[0];
+ l = channel_mapping[1];
+ r = channel_mapping[2];
+ sl = channel_mapping[3];
+ sr = channel_mapping[4];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR2(samples, i + sl, i + sr, 3, coef));
+ MIX_REAR2(samples, sl, sr, 3, coef));
break;
}
}
@@ -1441,21 +1439,21 @@ static int dca_filter_channels(DCAContext *s, int block_index)
/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
0, 8388608.0, 8388608.0 };*/
qmf_32_subbands(s, k, subband_samples[k],
- &s->samples[256 * s->channel_order_tab[k]],
- M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);
+ s->samples_chanptr[s->channel_order_tab[k]],
+ M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);
}
/* Down mixing */
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
- dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
+ dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
- &s->samples[256 * s->lfe_index],
- (1.0 / 256.0) * s->scale_bias);
+ s->samples_chanptr[s->lfe_index],
+ 1.0 / (256.0 * 32768.0));
/* Outputs 20bits pcm samples */
}
@@ -2067,10 +2065,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
int lfe_samples;
int num_core_channels = 0;
int i, ret;
- float *samples_flt;
+ float **samples_flt;
float *src_chan;
float *dst_chan;
- int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int core_ss_end;
int channels;
@@ -2081,7 +2078,6 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
int lavc;
int posn;
int j, k;
- int ch;
int endch;
s->xch_present = 0;
@@ -2342,19 +2338,23 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples_flt = (float *) s->frame.data[0];
- samples_s16 = (int16_t *) s->frame.data[0];
+ samples_flt = (float **) s->frame.extended_data;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
+ int ch;
+
+ for (ch = 0; ch < channels; ch++)
+ s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
+
dca_filter_channels(s, i);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
if ((s->source_pcm_res & 1) && s->xch_present) {
- float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
- float *lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
- float *rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
+ float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
+ float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
+ float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
@@ -2370,12 +2370,12 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
/* undo downmix */
for (j = ch; j < endch; j++) {
if (mask & (1 << j)) { /* this channel has been mixed-out */
- src_chan = s->samples + s->channel_order_tab[j] * 256;
+ src_chan = s->samples_chanptr[s->channel_order_tab[j]];
for (k = 0; k < endch; k++) {
achan = s->channel_order_tab[k];
scale = s->xxch_dmix_coeff[j][k];
if (scale != 0.0) {
- dst_chan = s->samples + achan * 256;
+ dst_chan = s->samples_chanptr[achan];
s->fdsp.vector_fmac_scalar(dst_chan, src_chan,
-scale, 256);
}
@@ -2388,14 +2388,14 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
scale = s->xxch_dmix_sf[chset];
for (j = 0; j < ch; j++) {
- src_chan = s->samples + s->channel_order_tab[j] * 256;
+ src_chan = s->samples_chanptr[s->channel_order_tab[j]];
for (k = 0; k < 256; k++)
src_chan[k] *= scale;
}
/* LFE channel is always part of core, scale if it exists */
if (s->lfe) {
- src_chan = s->samples + s->lfe_index * 256;
+ src_chan = s->samples_chanptr[s->lfe_index];
for (k = 0; k < 256; k++)
src_chan[k] *= scale;
}
@@ -2405,17 +2405,6 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
}
}
-
- if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
- s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
- channels);
- samples_flt += 256 * channels;
- } else {
- s->fmt_conv.float_to_int16_interleave(samples_s16,
- s->samples_chanptr, 256,
- channels);
- samples_s16 += 256 * channels;
- }
}
/* update lfe history */
@@ -2440,7 +2429,6 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
- int i;
s->avctx = avctx;
dca_init_vlcs();
@@ -2451,16 +2439,7 @@ static av_cold int dca_decode_init(AVCodecContext *avctx)
ff_dcadsp_init(&s->dcadsp);
ff_fmt_convert_init(&s->fmt_conv, avctx);
- for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
- s->samples_chanptr[i] = s->samples + i * 256;
-
- if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
- s->scale_bias = 1.0 / 32768.0;
- } else {
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- s->scale_bias = 1.0;
- }
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
@@ -2500,8 +2479,7 @@ AVCodec ff_dca_decoder = {
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};
diff --git a/libavcodec/imc.c b/libavcodec/imc.c
index d12c7551e9..9da769ba38 100644
--- a/libavcodec/imc.c
+++ b/libavcodec/imc.c
@@ -242,7 +242,7 @@ static av_cold int imc_decode_init(AVCodecContext *avctx)
return ret;
}
ff_dsputil_init(&q->dsp, avctx);
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO
: AV_CH_LAYOUT_STEREO;
@@ -662,7 +662,7 @@ static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
int i;
float re, im;
float *dst1 = q->out_samples;
- float *dst2 = q->out_samples + (COEFFS - 1) * channels;
+ float *dst2 = q->out_samples + (COEFFS - 1);
/* prerotation */
for (i = 0; i < COEFFS / 2; i++) {
@@ -684,8 +684,8 @@ static void imc_imdct256(IMCContext *q, IMCChannel *chctx, int channels)
+ (q->mdct_sine_window[i * 2] * re);
*dst2 = (q->mdct_sine_window[i * 2] * chctx->last_fft_im[i])
- (q->mdct_sine_window[COEFFS - 1 - i * 2] * re);
- dst1 += channels * 2;
- dst2 -= channels * 2;
+ dst1 += 2;
+ dst2 -= 2;
chctx->last_fft_im[i] = im;
}
}
@@ -786,7 +786,6 @@ static int imc_decode_block(AVCodecContext *avctx, IMCContext *q, int ch)
chctx->decoder_reset = 1;
if (chctx->decoder_reset) {
- memset(q->out_samples, 0, COEFFS * sizeof(*q->out_samples));
for (i = 0; i < BANDS; i++)
chctx->old_floor[i] = 1.0;
for (i = 0; i < COEFFS; i++)
@@ -945,7 +944,7 @@ static int imc_decode_frame(AVCodecContext *avctx, void *data,
}
for (i = 0; i < avctx->channels; i++) {
- q->out_samples = (float*)q->frame.data[0] + i;
+ q->out_samples = (float *)q->frame.extended_data[i];
q->dsp.bswap16_buf(buf16, (const uint16_t*)buf, IMC_BLOCK_SIZE / 2);
@@ -958,15 +957,8 @@ static int imc_decode_frame(AVCodecContext *avctx, void *data,
}
if (avctx->channels == 2) {
- float *src = (float*)q->frame.data[0], t1, t2;
-
- for (i = 0; i < COEFFS; i++) {
- t1 = src[0];
- t2 = src[1];
- src[0] = t1 + t2;
- src[1] = t1 - t2;
- src += 2;
- }
+ q->dsp.butterflies_float((float *)q->frame.extended_data[0],
+ (float *)q->frame.extended_data[1], COEFFS);
}
*got_frame_ptr = 1;
@@ -996,6 +988,8 @@ AVCodec ff_imc_decoder = {
.decode = imc_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("IMC (Intel Music Coder)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_IAC_DECODER
@@ -1009,5 +1003,7 @@ AVCodec ff_iac_decoder = {
.decode = imc_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("IAC (Indeo Audio Coder)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
diff --git a/libavcodec/mpc.c b/libavcodec/mpc.c
index 7ef437c4dc..d064924b91 100644
--- a/libavcodec/mpc.c
+++ b/libavcodec/mpc.c
@@ -43,28 +43,24 @@ void ff_mpc_init(void)
/**
* Process decoded Musepack data and produce PCM
*/
-static void mpc_synth(MPCContext *c, int16_t *out, int channels)
+static void mpc_synth(MPCContext *c, int16_t **out, int channels)
{
int dither_state = 0;
int i, ch;
- OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
for(ch = 0; ch < channels; ch++){
- samples_ptr = samples + ch;
for(i = 0; i < SAMPLES_PER_BAND; i++) {
ff_mpa_synth_filter_fixed(&c->mpadsp,
c->synth_buf[ch], &(c->synth_buf_offset[ch]),
ff_mpa_synth_window_fixed, &dither_state,
- samples_ptr, channels,
+ out[ch] + 32 * i, 1,
c->sb_samples[ch][i]);
- samples_ptr += 32 * channels;
}
}
- for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
- *out++=samples[i];
}
-void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
+void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
+ int channels)
{
int i, j, ch;
Band *bands = c->bands;
@@ -100,5 +96,5 @@ void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int ch
}
}
- mpc_synth(c, data, channels);
+ mpc_synth(c, out, channels);
}
diff --git a/libavcodec/mpc.h b/libavcodec/mpc.h
index 8b4deef689..34ef263b40 100644
--- a/libavcodec/mpc.h
+++ b/libavcodec/mpc.h
@@ -73,6 +73,6 @@ typedef struct {
} MPCContext;
void ff_mpc_init(void);
-void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, void *dst, int channels);
+void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels);
#endif /* AVCODEC_MPC_H */
diff --git a/libavcodec/mpc7.c b/libavcodec/mpc7.c
index cab4eee03d..9f3cf6c828 100644
--- a/libavcodec/mpc7.c
+++ b/libavcodec/mpc7.c
@@ -90,7 +90,7 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx)
c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
c->frames_to_skip = 0;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
avcodec_get_frame_defaults(&c->frame);
@@ -293,7 +293,7 @@ static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
for(ch = 0; ch < 2; ch++)
idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
- ff_mpc_dequantize_and_synth(c, mb, c->frame.data[0], 2);
+ ff_mpc_dequantize_and_synth(c, mb, (int16_t **)c->frame.extended_data, 2);
if(last_frame)
c->frame.nb_samples = c->lastframelen;
@@ -342,4 +342,6 @@ AVCodec ff_mpc7_decoder = {
.flush = mpc7_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/mpc8.c b/libavcodec/mpc8.c
index cb30b2c28b..64d19193b0 100644
--- a/libavcodec/mpc8.c
+++ b/libavcodec/mpc8.c
@@ -139,7 +139,7 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx)
c->MSS = get_bits1(&gb);
c->frames = 1 << (get_bits(&gb, 3) * 2);
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = (channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avctx->channels = channels;
@@ -413,7 +413,8 @@ static int mpc8_decode_frame(AVCodecContext * avctx, void *data,
}
}
- ff_mpc_dequantize_and_synth(c, maxband - 1, c->frame.data[0],
+ ff_mpc_dequantize_and_synth(c, maxband - 1,
+ (int16_t **)c->frame.extended_data,
avctx->channels);
c->cur_frame++;
@@ -446,4 +447,6 @@ AVCodec ff_mpc8_decoder = {
.flush = mpc8_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/ralf.c b/libavcodec/ralf.c
index 6866f79609..3244424f74 100644
--- a/libavcodec/ralf.c
+++ b/libavcodec/ralf.c
@@ -149,7 +149,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
avctx->sample_rate, avctx->channels);
return AVERROR_INVALIDDATA;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO;
@@ -338,7 +338,8 @@ static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)
}
}
-static int decode_block(AVCodecContext *avctx, GetBitContext *gb, int16_t *dst)
+static int decode_block(AVCodecContext *avctx, GetBitContext *gb,
+ int16_t *dst0, int16_t *dst1)
{
RALFContext *ctx = avctx->priv_data;
int len, ch, ret;
@@ -382,35 +383,35 @@ static int decode_block(AVCodecContext *avctx, GetBitContext *gb, int16_t *dst)
switch (dmode) {
case 0:
for (i = 0; i < len; i++)
- *dst++ = ch0[i] + ctx->bias[0];
+ dst0[i] = ch0[i] + ctx->bias[0];
break;
case 1:
for (i = 0; i < len; i++) {
- *dst++ = ch0[i] + ctx->bias[0];
- *dst++ = ch1[i] + ctx->bias[1];
+ dst0[i] = ch0[i] + ctx->bias[0];
+ dst1[i] = ch1[i] + ctx->bias[1];
}
break;
case 2:
for (i = 0; i < len; i++) {
ch0[i] += ctx->bias[0];
- *dst++ = ch0[i];
- *dst++ = ch0[i] - (ch1[i] + ctx->bias[1]);
+ dst0[i] = ch0[i];
+ dst1[i] = ch0[i] - (ch1[i] + ctx->bias[1]);
}
break;
case 3:
for (i = 0; i < len; i++) {
t = ch0[i] + ctx->bias[0];
t2 = ch1[i] + ctx->bias[1];
- *dst++ = t + t2;
- *dst++ = t;
+ dst0[i] = t + t2;
+ dst1[i] = t;
}
break;
case 4:
for (i = 0; i < len; i++) {
t = ch1[i] + ctx->bias[1];
t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);
- *dst++ = (t2 + t) / 2;
- *dst++ = (t2 - t) / 2;
+ dst0[i] = (t2 + t) / 2;
+ dst1[i] = (t2 - t) / 2;
}
break;
}
@@ -424,7 +425,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
RALFContext *ctx = avctx->priv_data;
- int16_t *samples;
+ int16_t *samples0;
+ int16_t *samples1;
int ret;
GetBitContext gb;
int table_size, table_bytes, i;
@@ -465,7 +467,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
av_log(avctx, AV_LOG_ERROR, "Me fail get_buffer()? That's unpossible!\n");
return ret;
}
- samples = (int16_t*)ctx->frame.data[0];
+ samples0 = (int16_t *)ctx->frame.data[0];
+ samples1 = (int16_t *)ctx->frame.data[1];
if (src_size < 5) {
av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");
@@ -498,8 +501,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
break;
}
init_get_bits(&gb, block_pointer, ctx->block_size[i] * 8);
- if (decode_block(avctx, &gb, samples + ctx->sample_offset
- * avctx->channels) < 0) {
+ if (decode_block(avctx, &gb, samples0 + ctx->sample_offset,
+ samples1 + ctx->sample_offset) < 0) {
av_log(avctx, AV_LOG_ERROR, "Sir, I got carsick in your office. Not decoding the rest of packet.\n");
break;
}
@@ -533,4 +536,6 @@ AVCodec ff_ralf_decoder = {
.flush = decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c
index 070f498cae..f47f4c30f1 100644
--- a/libavcodec/twinvq.c
+++ b/libavcodec/twinvq.c
@@ -666,7 +666,7 @@ static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
}
static void imdct_output(TwinContext *tctx, enum FrameType ftype, int wtype,
- float *out)
+ float **out)
{
const ModeTab *mtab = tctx->mtab;
int size1, size2;
@@ -685,24 +685,15 @@ static void imdct_output(TwinContext *tctx, enum FrameType ftype, int wtype,
size2 = tctx->last_block_pos[0];
size1 = mtab->size - size2;
- if (tctx->avctx->channels == 2) {
- tctx->dsp.butterflies_float_interleave(out, prev_buf,
- &prev_buf[2*mtab->size],
- size1);
-
- out += 2 * size1;
-
- tctx->dsp.butterflies_float_interleave(out, tctx->curr_frame,
- &tctx->curr_frame[2*mtab->size],
- size2);
- } else {
- memcpy(out, prev_buf, size1 * sizeof(*out));
- out += size1;
+ memcpy(&out[0][0 ], prev_buf, size1 * sizeof(out[0][0]));
+ memcpy(&out[0][size1], tctx->curr_frame, size2 * sizeof(out[0][0]));
- memcpy(out, tctx->curr_frame, size2 * sizeof(*out));
+ if (tctx->avctx->channels == 2) {
+ memcpy(&out[1][0], &prev_buf[2*mtab->size], size1 * sizeof(out[1][0]));
+ memcpy(&out[1][size1], &tctx->curr_frame[2*mtab->size], size2 * sizeof(out[1][0]));
+ tctx->dsp.butterflies_float(out[0], out[1], mtab->size);
}
-
}
static void dec_bark_env(TwinContext *tctx, const uint8_t *in, int use_hist,
@@ -825,7 +816,7 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data,
TwinContext *tctx = avctx->priv_data;
GetBitContext gb;
const ModeTab *mtab = tctx->mtab;
- float *out = NULL;
+ float **out = NULL;
enum FrameType ftype;
int window_type, ret;
static const enum FrameType wtype_to_ftype_table[] = {
@@ -846,7 +837,7 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- out = (float *)tctx->frame.data[0];
+ out = (float **)tctx->frame.extended_data;
}
init_get_bits(&gb, buf, buf_size * 8);
@@ -1119,7 +1110,7 @@ static av_cold int twin_decode_init(AVCodecContext *avctx)
int isampf, ibps;
tctx->avctx = avctx;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!avctx->extradata || avctx->extradata_size < 12) {
av_log(avctx, AV_LOG_ERROR, "Missing or incomplete extradata\n");
@@ -1184,4 +1175,6 @@ AVCodec ff_twinvq_decoder = {
.decode = twin_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("VQF TwinVQ"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-15 15:06:35 +0000