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authorJustin Ruggles <justin.ruggles@gmail.com>2012-03-23 17:42:17 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2012-04-24 21:28:27 -0400
commitc8af852b97447491823ff9b91413e32415e2babf (patch)
tree6c02f850cf954612c7077f266a75d663bb9cde57 /libavresample/audio_convert.c
parentc5671aeb77abb18a5a10ace314ab49e8fd3d0cb3 (diff)
downloadffmpeg-streaming-c8af852b97447491823ff9b91413e32415e2babf.zip
ffmpeg-streaming-c8af852b97447491823ff9b91413e32415e2babf.tar.gz
Add libavresample
This is a new library for audio sample format, channel layout, and sample rate conversion.
Diffstat (limited to 'libavresample/audio_convert.c')
-rw-r--r--libavresample/audio_convert.c334
1 files changed, 334 insertions, 0 deletions
diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c
new file mode 100644
index 0000000..200eb10
--- /dev/null
+++ b/libavresample/audio_convert.c
@@ -0,0 +1,334 @@
+/*
+ * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "config.h"
+#include "libavutil/libm.h"
+#include "libavutil/log.h"
+#include "libavutil/mem.h"
+#include "libavutil/samplefmt.h"
+#include "audio_convert.h"
+#include "audio_data.h"
+
+enum ConvFuncType {
+ CONV_FUNC_TYPE_FLAT,
+ CONV_FUNC_TYPE_INTERLEAVE,
+ CONV_FUNC_TYPE_DEINTERLEAVE,
+};
+
+typedef void (conv_func_flat)(uint8_t *out, const uint8_t *in, int len);
+
+typedef void (conv_func_interleave)(uint8_t *out, uint8_t *const *in,
+ int len, int channels);
+
+typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len,
+ int channels);
+
+struct AudioConvert {
+ AVAudioResampleContext *avr;
+ enum AVSampleFormat in_fmt;
+ enum AVSampleFormat out_fmt;
+ int channels;
+ int planes;
+ int ptr_align;
+ int samples_align;
+ int has_optimized_func;
+ const char *func_descr;
+ const char *func_descr_generic;
+ enum ConvFuncType func_type;
+ conv_func_flat *conv_flat;
+ conv_func_flat *conv_flat_generic;
+ conv_func_interleave *conv_interleave;
+ conv_func_interleave *conv_interleave_generic;
+ conv_func_deinterleave *conv_deinterleave;
+ conv_func_deinterleave *conv_deinterleave_generic;
+};
+
+void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt,
+ enum AVSampleFormat in_fmt, int channels,
+ int ptr_align, int samples_align,
+ const char *descr, void *conv)
+{
+ int found = 0;
+
+ switch (ac->func_type) {
+ case CONV_FUNC_TYPE_FLAT:
+ if (av_get_packed_sample_fmt(ac->in_fmt) == in_fmt &&
+ av_get_packed_sample_fmt(ac->out_fmt) == out_fmt) {
+ ac->conv_flat = conv;
+ ac->func_descr = descr;
+ ac->ptr_align = ptr_align;
+ ac->samples_align = samples_align;
+ if (ptr_align == 1 && samples_align == 1) {
+ ac->conv_flat_generic = conv;
+ ac->func_descr_generic = descr;
+ } else {
+ ac->has_optimized_func = 1;
+ }
+ found = 1;
+ }
+ break;
+ case CONV_FUNC_TYPE_INTERLEAVE:
+ if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
+ (!channels || ac->channels == channels)) {
+ ac->conv_interleave = conv;
+ ac->func_descr = descr;
+ ac->ptr_align = ptr_align;
+ ac->samples_align = samples_align;
+ if (ptr_align == 1 && samples_align == 1) {
+ ac->conv_interleave_generic = conv;
+ ac->func_descr_generic = descr;
+ } else {
+ ac->has_optimized_func = 1;
+ }
+ found = 1;
+ }
+ break;
+ case CONV_FUNC_TYPE_DEINTERLEAVE:
+ if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt &&
+ (!channels || ac->channels == channels)) {
+ ac->conv_deinterleave = conv;
+ ac->func_descr = descr;
+ ac->ptr_align = ptr_align;
+ ac->samples_align = samples_align;
+ if (ptr_align == 1 && samples_align == 1) {
+ ac->conv_deinterleave_generic = conv;
+ ac->func_descr_generic = descr;
+ } else {
+ ac->has_optimized_func = 1;
+ }
+ found = 1;
+ }
+ break;
+ }
+ if (found) {
+ av_log(ac->avr, AV_LOG_DEBUG, "audio_convert: found function: %-4s "
+ "to %-4s (%s)\n", av_get_sample_fmt_name(ac->in_fmt),
+ av_get_sample_fmt_name(ac->out_fmt), descr);
+ }
+}
+
+#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt
+
+#define CONV_LOOP(otype, expr) \
+ do { \
+ *(otype *)po = expr; \
+ pi += is; \
+ po += os; \
+ } while (po < end); \
+
+#define CONV_FUNC_FLAT(ofmt, otype, ifmt, itype, expr) \
+static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t *in, \
+ int len) \
+{ \
+ int is = sizeof(itype); \
+ int os = sizeof(otype); \
+ const uint8_t *pi = in; \
+ uint8_t *po = out; \
+ uint8_t *end = out + os * len; \
+ CONV_LOOP(otype, expr) \
+}
+
+#define CONV_FUNC_INTERLEAVE(ofmt, otype, ifmt, itype, expr) \
+static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t **in, \
+ int len, int channels) \
+{ \
+ int ch; \
+ int out_bps = sizeof(otype); \
+ int is = sizeof(itype); \
+ int os = channels * out_bps; \
+ for (ch = 0; ch < channels; ch++) { \
+ const uint8_t *pi = in[ch]; \
+ uint8_t *po = out + ch * out_bps; \
+ uint8_t *end = po + os * len; \
+ CONV_LOOP(otype, expr) \
+ } \
+}
+
+#define CONV_FUNC_DEINTERLEAVE(ofmt, otype, ifmt, itype, expr) \
+static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t **out, const uint8_t *in, \
+ int len, int channels) \
+{ \
+ int ch; \
+ int in_bps = sizeof(itype); \
+ int is = channels * in_bps; \
+ int os = sizeof(otype); \
+ for (ch = 0; ch < channels; ch++) { \
+ const uint8_t *pi = in + ch * in_bps; \
+ uint8_t *po = out[ch]; \
+ uint8_t *end = po + os * len; \
+ CONV_LOOP(otype, expr) \
+ } \
+}
+
+#define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr) \
+CONV_FUNC_FLAT( ofmt, otype, ifmt, itype, expr) \
+CONV_FUNC_INTERLEAVE( ofmt, otype, ifmt ## P, itype, expr) \
+CONV_FUNC_DEINTERLEAVE(ofmt ## P, otype, ifmt, itype, expr)
+
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_U8, uint8_t, *(const uint8_t *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 8)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 24)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0f / (1 << 7)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0 / (1 << 7)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t, (*(const int16_t *)pi >> 8) + 0x80)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi << 16)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0f / (1 << 15)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0 / (1 << 15)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t, (*(const int32_t *)pi >> 24) + 0x80)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi >> 16)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0f / (1U << 31)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0 / (1U << 31)))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8( lrintf(*(const float *)pi * (1 << 7)) + 0x80))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16( lrintf(*(const float *)pi * (1 << 15))))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *)pi * (1U << 31))))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_FLT, float, *(const float *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8( lrint(*(const double *)pi * (1 << 7)) + 0x80))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16( lrint(*(const double *)pi * (1 << 15))))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *)pi * (1U << 31))))
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
+CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_DBL, double, *(const double *)pi)
+
+#define SET_CONV_FUNC_GROUP(ofmt, ifmt) \
+ff_audio_convert_set_func(ac, ofmt, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt)); \
+ff_audio_convert_set_func(ac, ofmt ## P, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt ## P, ifmt)); \
+ff_audio_convert_set_func(ac, ofmt, ifmt ## P, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt ## P));
+
+static void set_generic_function(AudioConvert *ac)
+{
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL)
+ SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL)
+}
+
+AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
+ enum AVSampleFormat out_fmt,
+ enum AVSampleFormat in_fmt,
+ int channels)
+{
+ AudioConvert *ac;
+ int in_planar, out_planar;
+
+ ac = av_mallocz(sizeof(*ac));
+ if (!ac)
+ return NULL;
+
+ ac->avr = avr;
+ ac->out_fmt = out_fmt;
+ ac->in_fmt = in_fmt;
+ ac->channels = channels;
+
+ in_planar = av_sample_fmt_is_planar(in_fmt);
+ out_planar = av_sample_fmt_is_planar(out_fmt);
+
+ if (in_planar == out_planar) {
+ ac->func_type = CONV_FUNC_TYPE_FLAT;
+ ac->planes = in_planar ? ac->channels : 1;
+ } else if (in_planar)
+ ac->func_type = CONV_FUNC_TYPE_INTERLEAVE;
+ else
+ ac->func_type = CONV_FUNC_TYPE_DEINTERLEAVE;
+
+ set_generic_function(ac);
+
+ if (ARCH_X86)
+ ff_audio_convert_init_x86(ac);
+
+ return ac;
+}
+
+int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len)
+{
+ int use_generic = 1;
+
+ /* determine whether to use the optimized function based on pointer and
+ samples alignment in both the input and output */
+ if (ac->has_optimized_func) {
+ int ptr_align = FFMIN(in->ptr_align, out->ptr_align);
+ int samples_align = FFMIN(in->samples_align, out->samples_align);
+ int aligned_len = FFALIGN(len, ac->samples_align);
+ if (!(ptr_align % ac->ptr_align) && samples_align >= aligned_len) {
+ len = aligned_len;
+ use_generic = 0;
+ }
+ }
+ av_dlog(ac->avr, "%d samples - audio_convert: %s to %s (%s)\n", len,
+ av_get_sample_fmt_name(ac->in_fmt),
+ av_get_sample_fmt_name(ac->out_fmt),
+ use_generic ? ac->func_descr_generic : ac->func_descr);
+
+ switch (ac->func_type) {
+ case CONV_FUNC_TYPE_FLAT: {
+ int p;
+ if (!in->is_planar)
+ len *= in->channels;
+ if (use_generic) {
+ for (p = 0; p < ac->planes; p++)
+ ac->conv_flat_generic(out->data[p], in->data[p], len);
+ } else {
+ for (p = 0; p < ac->planes; p++)
+ ac->conv_flat(out->data[p], in->data[p], len);
+ }
+ break;
+ }
+ case CONV_FUNC_TYPE_INTERLEAVE:
+ if (use_generic)
+ ac->conv_interleave_generic(out->data[0], in->data, len, ac->channels);
+ else
+ ac->conv_interleave(out->data[0], in->data, len, ac->channels);
+ break;
+ case CONV_FUNC_TYPE_DEINTERLEAVE:
+ if (use_generic)
+ ac->conv_deinterleave_generic(out->data, in->data[0], len, ac->channels);
+ else
+ ac->conv_deinterleave(out->data, in->data[0], len, ac->channels);
+ break;
+ }
+
+ out->nb_samples = in->nb_samples;
+ return 0;
+}
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