[FFmpeg-devel] [PATCH 05/10] avcodec/dsd: rework codecs to use AV_SAMPLE_FMT_DSD
Peter Ross
pross at xvid.org
Tue May 6 14:38:36 CEST 2014
Signed-off-by: Peter Ross <pross at xvid.org>
---
libavcodec/Makefile | 8 +-
libavcodec/allcodecs.c | 8 +-
libavcodec/codec_desc.c | 8 +-
libavcodec/dsd_tablegen.c | 38 ---------
libavcodec/dsd_tablegen.h | 95 ----------------------
libavcodec/dsddec.c | 198 ++++++++++++++++++++++------------------------
libavcodec/utils.c | 2 +
7 files changed, 110 insertions(+), 247 deletions(-)
delete mode 100644 libavcodec/dsd_tablegen.c
delete mode 100644 libavcodec/dsd_tablegen.h
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index d239924..7c475fb 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -180,9 +180,13 @@ OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o
OBJS-$(CONFIG_DPX_DECODER) += dpx.o
OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.o
OBJS-$(CONFIG_DSD_LSBF_DECODER) += dsddec.o
+OBJS-$(CONFIG_DSD_LSBF_ENCODER) += dsddec.o
OBJS-$(CONFIG_DSD_MSBF_DECODER) += dsddec.o
+OBJS-$(CONFIG_DSD_MSBF_ENCODER) += dsddec.o
OBJS-$(CONFIG_DSD_LSBF_PLANAR_DECODER) += dsddec.o
+OBJS-$(CONFIG_DSD_LSBF_PLANAR_ENCODER) += dsddec.o
OBJS-$(CONFIG_DSD_MSBF_PLANAR_DECODER) += dsddec.o
+OBJS-$(CONFIG_DSD_MSBF_PLANAR_ENCODER) += dsddec.o
OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinav.o
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
@@ -853,7 +857,6 @@ HOSTPROGS = aac_tablegen \
aacps_tablegen \
cbrt_tablegen \
cos_tablegen \
- dsd_tablegen \
dv_tablegen \
motionpixels_tablegen \
mpegaudio_tablegen \
@@ -878,7 +881,7 @@ else
$(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
endif
-GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dsd_tables.h dv_tables.h \
+GEN_HEADERS = cbrt_tables.h aacps_tables.h aac_tables.h dv_tables.h \
sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
pcm_tables.h qdm2_tables.h
GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))
@@ -890,7 +893,6 @@ ifdef CONFIG_HARDCODED_TABLES
$(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
$(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
-$(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
$(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
$(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
$(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index a1cd44e..eb6b0c3 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -337,10 +337,10 @@ void avcodec_register_all(void)
REGISTER_DECODER(BMV_AUDIO, bmv_audio);
REGISTER_DECODER(COOK, cook);
REGISTER_ENCDEC (DCA, dca);
- REGISTER_DECODER(DSD_LSBF, dsd_lsbf);
- REGISTER_DECODER(DSD_MSBF, dsd_msbf);
- REGISTER_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar);
- REGISTER_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar);
+ REGISTER_ENCDEC (DSD_LSBF, dsd_lsbf);
+ REGISTER_ENCDEC (DSD_MSBF, dsd_msbf);
+ REGISTER_ENCDEC (DSD_LSBF_PLANAR, dsd_lsbf_planar);
+ REGISTER_ENCDEC (DSD_MSBF_PLANAR, dsd_msbf_planar);
REGISTER_DECODER(DSICINAUDIO, dsicinaudio);
REGISTER_ENCDEC (EAC3, eac3);
REGISTER_DECODER(EVRC, evrc);
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 4906a0a..057c6a3 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -2485,28 +2485,28 @@ static const AVCodecDescriptor codec_descriptors[] = {
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_lsbf",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first"),
- .props = AV_CODEC_PROP_LOSSY,
+ .props = AV_CODEC_PROP_LOSSLESS,
},
{
.id = AV_CODEC_ID_DSD_MSBF,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_msbf",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first"),
- .props = AV_CODEC_PROP_LOSSY,
+ .props = AV_CODEC_PROP_LOSSLESS,
},
{
.id = AV_CODEC_ID_DSD_LSBF_PLANAR,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_lsbf_planar",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), least significant bit first, planar"),
- .props = AV_CODEC_PROP_LOSSY,
+ .props = AV_CODEC_PROP_LOSSLESS,
},
{
.id = AV_CODEC_ID_DSD_MSBF_PLANAR,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dsd_msbf_planar",
.long_name = NULL_IF_CONFIG_SMALL("DSD (Direct Stream Digital), most significant bit first, planar"),
- .props = AV_CODEC_PROP_LOSSY,
+ .props = AV_CODEC_PROP_LOSSLESS,
},
/* subtitle codecs */
diff --git a/libavcodec/dsd_tablegen.c b/libavcodec/dsd_tablegen.c
deleted file mode 100644
index dbeb9fe..0000000
--- a/libavcodec/dsd_tablegen.c
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Generate a header file for hardcoded DSD tables
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdlib.h>
-#define CONFIG_HARDCODED_TABLES 0
-#include "dsd_tablegen.h"
-#include "tableprint.h"
-#include <inttypes.h>
-
-int main(void)
-{
- dsd_ctables_tableinit();
-
- write_fileheader();
-
- printf("static const double ctables[CTABLES][256] = {\n");
- write_float_2d_array(ctables, CTABLES, 256);
- printf("};\n");
-
- return 0;
-}
diff --git a/libavcodec/dsd_tablegen.h b/libavcodec/dsd_tablegen.h
deleted file mode 100644
index 6afb416..0000000
--- a/libavcodec/dsd_tablegen.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Header file for hardcoded DSD tables
- * based on BSD licensed dsd2pcm by Sebastian Gesemann
- * Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_DSD_TABLEGEN_H
-#define AVCODEC_DSD_TABLEGEN_H
-
-#include <stdint.h>
-#include "libavutil/attributes.h"
-
-#define HTAPS 48 /** number of FIR constants */
-#define CTABLES ((HTAPS + 7) / 8) /** number of "8 MACs" lookup tables */
-
-#if CONFIG_HARDCODED_TABLES
-#define dsd_ctables_tableinit()
-#include "libavcodec/dsd_tables.h"
-#else
-#include "libavutil/common.h"
-
-/*
- * Properties of this 96-tap lowpass filter when applied on a signal
- * with sampling rate of 44100*64 Hz:
- *
- * () has a delay of 17 microseconds.
- *
- * () flat response up to 48 kHz
- *
- * () if you downsample afterwards by a factor of 8, the
- * spectrum below 70 kHz is practically alias-free.
- *
- * () stopband rejection is about 160 dB
- *
- * The coefficient tables ("ctables") take only 6 Kibi Bytes and
- * should fit into a modern processor's fast cache.
- */
-
-/**
- * The 2nd half (48 coeffs) of a 96-tap symmetric lowpass filter
- */
-static const double htaps[HTAPS] = {
- 0.09950731974056658, 0.09562845727714668, 0.08819647126516944,
- 0.07782552527068175, 0.06534876523171299, 0.05172629311427257,
- 0.0379429484910187, 0.02490921351762261, 0.0133774746265897,
- 0.003883043418804416, -0.003284703416210726, -0.008080250212687497,
- -0.01067241812471033, -0.01139427235000863, -0.0106813877974587,
- -0.009007905078766049, -0.006828859761015335, -0.004535184322001496,
- -0.002425035959059578, -0.0006922187080790708, 0.0005700762133516592,
- 0.001353838005269448, 0.001713709169690937, 0.001742046839472948,
- 0.001545601648013235, 0.001226696225277855, 0.0008704322683580222,
- 0.0005381636200535649, 0.000266446345425276, 7.002968738383528e-05,
- -5.279407053811266e-05, -0.0001140625650874684, -0.0001304796361231895,
- -0.0001189970287491285, -9.396247155265073e-05, -6.577634378272832e-05,
- -4.07492895872535e-05, -2.17407957554587e-05, -9.163058931391722e-06,
- -2.017460145032201e-06, 1.249721855219005e-06, 2.166655190537392e-06,
- 1.930520892991082e-06, 1.319400334374195e-06, 7.410039764949091e-07,
- 3.423230509967409e-07, 1.244182214744588e-07, 3.130441005359396e-08
-};
-
-static float ctables[CTABLES][256];
-
-static av_cold void dsd_ctables_tableinit(void)
-{
- int t, e, m, k;
- double acc;
- for (t = 0; t < CTABLES; ++t) {
- k = FFMIN(HTAPS - t * 8, 8);
- for (e = 0; e < 256; ++e) {
- acc = 0.0;
- for (m = 0; m < k; ++m)
- acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m];
- ctables[CTABLES - 1 - t][e] = (float)acc;
- }
- }
-}
-#endif /* CONFIG_HARDCODED_TABLES */
-
-#endif /* AVCODEC_DSD_TABLEGEN_H */
diff --git a/libavcodec/dsddec.c b/libavcodec/dsddec.c
index f1dfd4b..acfe27e 100644
--- a/libavcodec/dsddec.c
+++ b/libavcodec/dsddec.c
@@ -29,127 +29,58 @@
#include "libavcodec/internal.h"
#include "libavcodec/mathops.h"
#include "avcodec.h"
-#include "dsd_tablegen.h"
-#define FIFOSIZE 16 /** must be a power of two */
-#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */
-
-#if FIFOSIZE * 8 < HTAPS * 2
-#error "FIFOSIZE too small"
-#endif
-
-/**
- * Per-channel buffer
- */
-typedef struct {
- unsigned char buf[FIFOSIZE];
- unsigned pos;
-} DSDContext;
-
-static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
- const unsigned char *src, ptrdiff_t src_stride,
- float *dst, ptrdiff_t dst_stride)
-{
- unsigned pos, i;
- unsigned char* p;
- double sum;
-
- pos = s->pos;
-
- while (samples-- > 0) {
- s->buf[pos] = lsbf ? ff_reverse[*src] : *src;
- src += src_stride;
-
- p = s->buf + ((pos - CTABLES) & FIFOMASK);
- *p = ff_reverse[*p];
-
- sum = 0.0;
- for (i = 0; i < CTABLES; i++) {
- unsigned char a = s->buf[(pos - i) & FIFOMASK];
- unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK];
- sum += ctables[i][a] + ctables[i][b];
- }
-
- *dst = (float)sum;
- dst += dst_stride;
-
- pos = (pos + 1) & FIFOMASK;
- }
-
- s->pos = pos;
-}
-
-static av_cold void init_static_data(void)
+static void reverse_memcpy(uint8_t *dst, const uint8_t *src, int size)
{
- static int done = 0;
- if (done)
- return;
- dsd_ctables_tableinit();
- done = 1;
+ int i;
+ for (i = 0; i < size; i++)
+ dst[i] = ff_reverse[src[i]];
}
static av_cold int decode_init(AVCodecContext *avctx)
{
- DSDContext * s;
- int i;
-
- init_static_data();
-
- s = av_malloc_array(sizeof(DSDContext), avctx->channels);
- if (!s)
- return AVERROR(ENOMEM);
-
- for (i = 0; i < avctx->channels; i++) {
- s[i].pos = 0;
- memset(s[i].buf, 0x69, sizeof(s[i].buf));
-
- /* 0x69 = 01101001
- * This pattern "on repeat" makes a low energy 352.8 kHz tone
- * and a high energy 1.0584 MHz tone which should be filtered
- * out completely by any playback system --> silence
- */
- }
-
- avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
- avctx->priv_data = s;
+ avctx->sample_fmt = avctx->codec->sample_fmts[0];
return 0;
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
- DSDContext * s = avctx->priv_data;
+ const uint8_t * src = avpkt->data;
AVFrame *frame = data;
- int ret, i;
- int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR;
- int src_next;
- int src_stride;
+ int ret, ch;
frame->nb_samples = avpkt->size / avctx->channels;
-
- if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) {
- src_next = frame->nb_samples;
- src_stride = 1;
- } else {
- src_next = 1;
- src_stride = avctx->channels;
- }
-
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- for (i = 0; i < avctx->channels; i++) {
- float * dst = ((float **)frame->extended_data)[i];
- dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
- avpkt->data + i * src_next, src_stride,
- dst, 1);
+ switch(avctx->codec_id) {
+ case AV_CODEC_ID_DSD_LSBF:
+ reverse_memcpy(frame->data[0], src, frame->nb_samples * avctx->channels);
+ break;
+ case AV_CODEC_ID_DSD_MSBF:
+ memcpy(frame->data[0], src, frame->nb_samples * avctx->channels);
+ break;
+ case AV_CODEC_ID_DSD_LSBF_PLANAR:
+ for (ch = 0; ch < avctx->channels; ch++ ) {
+ reverse_memcpy(frame->extended_data[ch], src, frame->nb_samples);
+ src += frame->nb_samples;
+ }
+ break;
+ case AV_CODEC_ID_DSD_MSBF_PLANAR:
+ for (ch = 0; ch < avctx->channels; ch++ ) {
+ memcpy(frame->extended_data[ch], src, frame->nb_samples);
+ src += frame->nb_samples;
+ }
+ break;
+ default:
+ return -1;
}
-
*got_frame_ptr = 1;
return frame->nb_samples * avctx->channels;
}
-#define DSD_DECODER(id_, name_, long_name_) \
+#define DSD_DECODER(id_, sample_fmt_, name_, long_name_) \
AVCodec ff_##name_##_decoder = { \
.name = #name_, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
@@ -157,11 +88,72 @@ AVCodec ff_##name_##_decoder = { \
.id = AV_CODEC_ID_##id_, \
.init = decode_init, \
.decode = decode_frame, \
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
+ .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
+ AV_SAMPLE_FMT_NONE }, \
+};
+
+DSD_DECODER(DSD_LSBF, AV_SAMPLE_FMT_DSD, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
+DSD_DECODER(DSD_MSBF, AV_SAMPLE_FMT_DSD, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
+DSD_DECODER(DSD_LSBF_PLANAR, AV_SAMPLE_FMT_DSDP, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
+DSD_DECODER(DSD_MSBF_PLANAR, AV_SAMPLE_FMT_DSDP, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
+
+static av_cold int encode_init(AVCodecContext *avctx)
+{
+ avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
+ avctx->block_align = avctx->channels * avctx->bits_per_coded_sample / 8;
+ avctx->bit_rate = avctx->block_align * avctx->sample_rate * 8;
+ return 0;
+}
+
+static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
+ uint8_t *dst = avpkt->data;
+ int n, ret, ch;
+ n = frame->nb_samples * avctx->channels;
+ if ((ret = ff_alloc_packet2(avctx, avpkt, n)) < 0)
+ return ret;
+
+ switch(avctx->codec_id) {
+ case AV_CODEC_ID_DSD_LSBF:
+ reverse_memcpy(dst, frame->data[0], n);
+ break;
+ case AV_CODEC_ID_DSD_MSBF:
+ memcpy(dst, frame->data[0], n);
+ break;
+ case AV_CODEC_ID_DSD_LSBF_PLANAR:
+ for (ch = 0; ch < avctx->channels; ch++ ) {
+ reverse_memcpy(dst, frame->extended_data[ch], frame->nb_samples);
+ dst += frame->nb_samples;
+ }
+ break;
+ case AV_CODEC_ID_DSD_MSBF_PLANAR:
+ for (ch = 0; ch < avctx->channels; ch++ ) {
+ memcpy(dst, frame->extended_data[ch], frame->nb_samples);
+ dst += frame->nb_samples;
+ }
+ break;
+ default:
+ return -1;
+ }
+ *got_packet_ptr = 1;
+ return 0;
+}
+
+#define DSD_ENCODER(id_, sample_fmt_, name_, long_name_) \
+AVCodec ff_##name_##_encoder = { \
+ .name = #name_, \
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+ .type = AVMEDIA_TYPE_AUDIO, \
+ .id = AV_CODEC_ID_##id_, \
+ .init = encode_init, \
+ .encode2 = encode_frame, \
+ .capabilities = CODEC_CAP_VARIABLE_FRAME_SIZE, \
+ .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
AV_SAMPLE_FMT_NONE }, \
};
-DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
-DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
-DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
-DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
+DSD_ENCODER(DSD_LSBF, AV_SAMPLE_FMT_DSD, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first")
+DSD_ENCODER(DSD_MSBF, AV_SAMPLE_FMT_DSD, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first")
+DSD_ENCODER(DSD_LSBF_PLANAR, AV_SAMPLE_FMT_DSDP, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar")
+DSD_ENCODER(DSD_MSBF_PLANAR, AV_SAMPLE_FMT_DSDP, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar")
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index b899392..8bb2579 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3038,11 +3038,13 @@ enum AVCodecID av_get_pcm_codec(enum AVSampleFormat fmt, int be)
[AV_SAMPLE_FMT_S32 ] = { AV_CODEC_ID_PCM_S32LE, AV_CODEC_ID_PCM_S32BE },
[AV_SAMPLE_FMT_FLT ] = { AV_CODEC_ID_PCM_F32LE, AV_CODEC_ID_PCM_F32BE },
[AV_SAMPLE_FMT_DBL ] = { AV_CODEC_ID_PCM_F64LE, AV_CODEC_ID_PCM_F64BE },
+ [AV_SAMPLE_FMT_DSD ] = { AV_CODEC_ID_DSD_MSBF, AV_CODEC_ID_DSD_MSBF },
[AV_SAMPLE_FMT_U8P ] = { AV_CODEC_ID_PCM_U8, AV_CODEC_ID_PCM_U8 },
[AV_SAMPLE_FMT_S16P] = { AV_CODEC_ID_PCM_S16LE, AV_CODEC_ID_PCM_S16BE },
[AV_SAMPLE_FMT_S32P] = { AV_CODEC_ID_PCM_S32LE, AV_CODEC_ID_PCM_S32BE },
[AV_SAMPLE_FMT_FLTP] = { AV_CODEC_ID_PCM_F32LE, AV_CODEC_ID_PCM_F32BE },
[AV_SAMPLE_FMT_DBLP] = { AV_CODEC_ID_PCM_F64LE, AV_CODEC_ID_PCM_F64BE },
+ [AV_SAMPLE_FMT_DSDP] = { AV_CODEC_ID_DSD_MSBF, AV_CODEC_ID_DSD_MSBF },
};
if (fmt < 0 || fmt >= AV_SAMPLE_FMT_NB)
return AV_CODEC_ID_NONE;
--
1.8.3.2
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: Digital signature
URL: <http://ffmpeg.org/pipermail/ffmpeg-devel/attachments/20140506/443d798e/attachment.asc>
More information about the ffmpeg-devel
mailing list