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ffmpeg.wasm-core
Commit c8af852b authored by Justin Ruggles's avatar Justin Ruggles
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Add libavresample 

This is a new library for audio sample format, channel layout, and sample rate
conversion.
parent c5671aeb
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Changelog
View file @ c8af852b
......@@ -16,6 +16,7 @@ version <next>:
- RealAudio Lossless decoder
- ZeroCodec decoder
- drop support for avconv without libavfilter
- add libavresample audio conversion library
version 0.8:
......
Makefile
View file @ c8af852b
......@@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_PATH)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avutil swscale
ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil swscale
IFLAGS := -I. -I$(SRC_PATH)
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
......@@ -71,6 +71,7 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_SWSCALE) += swscale
......
configure
View file @ c8af852b
......@@ -110,6 +110,7 @@ Component options:
--disable-avformat disable libavformat build
--disable-swscale disable libswscale build
--disable-avfilter disable video filter support [no]
--disable-avresample disable libavresample build [no]
--disable-pthreads disable pthreads [auto]
--disable-w32threads disable Win32 threads [auto]
--enable-x11grab enable X11 grabbing [no]
......@@ -927,6 +928,7 @@ CONFIG_LIST="
avdevice
avfilter
avformat
avresample
avisynth
bzlib
dct
......@@ -1536,7 +1538,7 @@ avdevice_deps="avcodec avformat"
avformat_deps="avcodec"
# programs
avconv_deps="avcodec avfilter avformat swscale"
avconv_deps="avcodec avfilter avformat avresample swscale"
avplay_deps="avcodec avformat swscale sdl"
avplay_select="rdft"
avprobe_deps="avcodec avformat"
......@@ -1684,6 +1686,7 @@ enable avcodec
enable avdevice
enable avfilter
enable avformat
enable avresample
enable avutil
enable swscale
......@@ -3385,6 +3388,7 @@ get_version LIBAVCODEC libavcodec/version.h
get_version LIBAVDEVICE libavdevice/avdevice.h
get_version LIBAVFILTER libavfilter/version.h
get_version LIBAVFORMAT libavformat/version.h
get_version LIBAVRESAMPLE libavresample/version.h
get_version LIBAVUTIL libavutil/avutil.h
get_version LIBSWSCALE libswscale/swscale.h
......@@ -3504,4 +3508,5 @@ pkgconfig_generate libavcodec "Libav codec library" "$LIBAVCODEC_VERSION" "$extr
pkgconfig_generate libavformat "Libav container format library" "$LIBAVFORMAT_VERSION" "$extralibs" "libavcodec = $LIBAVCODEC_VERSION"
pkgconfig_generate libavdevice "Libav device handling library" "$LIBAVDEVICE_VERSION" "$extralibs" "libavformat = $LIBAVFORMAT_VERSION"
pkgconfig_generate libavfilter "Libav video filtering library" "$LIBAVFILTER_VERSION" "$extralibs"
pkgconfig_generate libavresample "Libav audio resampling library" "$LIBAVRESAMPLE_VERSION" "$extralibs"
pkgconfig_generate libswscale "Libav image rescaling library" "$LIBSWSCALE_VERSION" "$LIBM" "libavutil = $LIBAVUTIL_VERSION"
doc/APIchanges
View file @ c8af852b
......@@ -2,16 +2,20 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase.
The last version increases were:
libavcodec: 2012-01-27
libavdevice: 2011-04-18
libavfilter: 2011-04-18
libavformat: 2012-01-27
libswscale: 2011-06-20
libavutil: 2011-04-18
libavcodec: 2012-01-27
libavdevice: 2011-04-18
libavfilter: 2011-04-18
libavformat: 2012-01-27
libavresample: 2012-xx-xx
libswscale: 2011-06-20
libavutil: 2011-04-18
API changes, most recent first:
2012-xx-xx - xxxxxxx - lavr 0.0.0
Add libavresample audio conversion library
2012-xx-xx - xxxxxxx - lavu 51.28.0 - audio_fifo.h
Add audio FIFO functions:
av_audio_fifo_free()
......
libavresample/Makefile 0 → 100644
View file @ c8af852b
NAME = avresample
FFLIBS = avutil
HEADERS = avresample.h \
version.h
OBJS = audio_convert.o \
audio_data.o \
audio_mix.o \
audio_mix_matrix.o \
options.o \
resample.o \
utils.o
TESTPROGS = avresample
libavresample/audio_convert.c 0 → 100644
View file @ c8af852b
/*
* 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;
}
libavresample/audio_convert.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_AUDIO_CONVERT_H
#define AVRESAMPLE_AUDIO_CONVERT_H
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "audio_data.h"
typedef struct AudioConvert AudioConvert;
/**
* Set conversion function if the parameters match.
*
* This compares the parameters of the conversion function to the parameters
* in the AudioConvert context. If the parameters do not match, no changes are
* made to the active functions. If the parameters do match and the alignment
* is not constrained, the function is set as the generic conversion function.
* If the parameters match and the alignment is constrained, the function is
* set as the optimized conversion function.
*
* @param ac AudioConvert context
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels, or 0 for any number of channels
* @param ptr_align buffer pointer alignment, in bytes
* @param sample_align buffer size alignment, in samples
* @param descr function type description (e.g. "C" or "SSE")
* @param conv conversion function pointer
*/
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);
/**
* Allocate and initialize AudioConvert context for sample format conversion.
*
* @param avr AVAudioResampleContext
* @param out_fmt output sample format
* @param in_fmt input sample format
* @param channels number of channels
* @return newly-allocated AudioConvert context
*/
AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr,
enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels);
/**
* Convert audio data from one sample format to another.
*
* For each call, the alignment of the input and output AudioData buffers are
* examined to determine whether to use the generic or optimized conversion
* function (when available).
*
* @param ac AudioConvert context
* @param out output audio data
* @param in input audio data
* @param len number of samples to convert
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in, int len);
/* arch-specific initialization functions */
void ff_audio_convert_init_x86(AudioConvert *ac);
#endif /* AVRESAMPLE_AUDIO_CONVERT_H */
libavresample/audio_data.c 0 → 100644
View file @ c8af852b
/*
* 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 "libavutil/mem.h"
#include "audio_data.h"
static const AVClass audio_data_class = {
.class_name = "AudioData",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
};
/*
* Calculate alignment for data pointers.
*/
static void calc_ptr_alignment(AudioData *a)
{
int p;
int min_align = 128;
for (p = 0; p < a->planes; p++) {
int cur_align = 128;
while ((intptr_t)a->data[p] % cur_align)
cur_align >>= 1;
if (cur_align < min_align)
min_align = cur_align;
}
a->ptr_align = min_align;
}
int ff_audio_data_set_channels(AudioData *a, int channels)
{
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
channels > a->allocated_channels)
return AVERROR(EINVAL);
a->channels = channels;
a->planes = a->is_planar ? channels : 1;
calc_ptr_alignment(a);
return 0;
}
int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
int nb_samples, enum AVSampleFormat sample_fmt,
int read_only, const char *name)
{
int p;
memset(a, 0, sizeof(*a));
a->class = &audio_data_class;
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
return AVERROR(EINVAL);
}
a->sample_size = av_get_bytes_per_sample(sample_fmt);
if (!a->sample_size) {
av_log(a, AV_LOG_ERROR, "invalid sample format\n");
return AVERROR(EINVAL);
}
a->is_planar = av_sample_fmt_is_planar(sample_fmt);
a->planes = a->is_planar ? channels : 1;
a->stride = a->sample_size * (a->is_planar ? 1 : channels);
for (p = 0; p < (a->is_planar ? channels : 1); p++) {
if (!src[p]) {
av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
return AVERROR(EINVAL);
}
a->data[p] = src[p];
}
a->allocated_samples = nb_samples * !read_only;
a->nb_samples = nb_samples;
a->sample_fmt = sample_fmt;
a->channels = channels;
a->allocated_channels = channels;
a->read_only = read_only;
a->allow_realloc = 0;
a->name = name ? name : "{no name}";
calc_ptr_alignment(a);
a->samples_align = plane_size / a->stride;
return 0;
}
AudioData *ff_audio_data_alloc(int channels, int nb_samples,
enum AVSampleFormat sample_fmt, const char *name)
{
AudioData *a;
int ret;
if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
return NULL;
a = av_mallocz(sizeof(*a));
if (!a)
return NULL;
a->sample_size = av_get_bytes_per_sample(sample_fmt);
if (!a->sample_size) {
av_free(a);
return NULL;
}
a->is_planar = av_sample_fmt_is_planar(sample_fmt);
a->planes = a->is_planar ? channels : 1;
a->stride = a->sample_size * (a->is_planar ? 1 : channels);
a->class = &audio_data_class;
a->sample_fmt = sample_fmt;
a->channels = channels;
a->allocated_channels = channels;
a->read_only = 0;
a->allow_realloc = 1;
a->name = name ? name : "{no name}";
if (nb_samples > 0) {
ret = ff_audio_data_realloc(a, nb_samples);
if (ret < 0) {
av_free(a);
return NULL;
}
return a;
} else {
calc_ptr_alignment(a);
return a;
}
}
int ff_audio_data_realloc(AudioData *a, int nb_samples)
{
int ret, new_buf_size, plane_size, p;
/* check if buffer is already large enough */
if (a->allocated_samples >= nb_samples)
return 0;
/* validate that the output is not read-only and realloc is allowed */
if (a->read_only || !a->allow_realloc)
return AVERROR(EINVAL);
new_buf_size = av_samples_get_buffer_size(&plane_size,
a->allocated_channels, nb_samples,
a->sample_fmt, 0);
if (new_buf_size < 0)
return new_buf_size;
/* if there is already data in the buffer and the sample format is planar,
allocate a new buffer and copy the data, otherwise just realloc the
internal buffer and set new data pointers */
if (a->nb_samples > 0 && a->is_planar) {
uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
nb_samples, a->sample_fmt, 0);
if (ret < 0)
return ret;
for (p = 0; p < a->planes; p++)
memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
av_freep(&a->buffer);
memcpy(a->data, new_data, sizeof(new_data));
a->buffer = a->data[0];
} else {
av_freep(&a->buffer);
a->buffer = av_malloc(new_buf_size);
if (!a->buffer)
return AVERROR(ENOMEM);
ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
a->allocated_channels, nb_samples,
a->sample_fmt, 0);
if (ret < 0)
return ret;
}
a->buffer_size = new_buf_size;
a->allocated_samples = nb_samples;
calc_ptr_alignment(a);
a->samples_align = plane_size / a->stride;
return 0;
}
void ff_audio_data_free(AudioData **a)
{
if (!*a)
return;
av_free((*a)->buffer);
av_freep(a);
}
int ff_audio_data_copy(AudioData *dst, AudioData *src)
{
int ret, p;
/* validate input/output compatibility */
if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
return AVERROR(EINVAL);
/* if the input is empty, just empty the output */
if (!src->nb_samples) {
dst->nb_samples = 0;
return 0;
}
/* reallocate output if necessary */
ret = ff_audio_data_realloc(dst, src->nb_samples);
if (ret < 0)
return ret;
/* copy data */
for (p = 0; p < src->planes; p++)
memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
dst->nb_samples = src->nb_samples;
return 0;
}
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
int src_offset, int nb_samples)
{
int ret, p, dst_offset2, dst_move_size;
/* validate input/output compatibility */
if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
return AVERROR(EINVAL);
}
/* validate offsets are within the buffer bounds */
if (dst_offset < 0 || dst_offset > dst->nb_samples ||
src_offset < 0 || src_offset > src->nb_samples) {
av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
src_offset, dst_offset);
return AVERROR(EINVAL);
}
/* check offsets and sizes to see if we can just do nothing and return */
if (nb_samples > src->nb_samples - src_offset)
nb_samples = src->nb_samples - src_offset;
if (nb_samples <= 0)
return 0;
/* validate that the output is not read-only */
if (dst->read_only) {
av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
return AVERROR(EINVAL);
}
/* reallocate output if necessary */
ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
if (ret < 0) {
av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
return ret;
}
dst_offset2 = dst_offset + nb_samples;
dst_move_size = dst->nb_samples - dst_offset;
for (p = 0; p < src->planes; p++) {
if (dst_move_size > 0) {
memmove(dst->data[p] + dst_offset2 * dst->stride,
dst->data[p] + dst_offset * dst->stride,
dst_move_size * dst->stride);
}
memcpy(dst->data[p] + dst_offset * dst->stride,
src->data[p] + src_offset * src->stride,
nb_samples * src->stride);
}
dst->nb_samples += nb_samples;
return 0;
}
void ff_audio_data_drain(AudioData *a, int nb_samples)
{
if (a->nb_samples <= nb_samples) {
/* drain the whole buffer */
a->nb_samples = 0;
} else {
int p;
int move_offset = a->stride * nb_samples;
int move_size = a->stride * (a->nb_samples - nb_samples);
for (p = 0; p < a->planes; p++)
memmove(a->data[p], a->data[p] + move_offset, move_size);
a->nb_samples -= nb_samples;
}
}
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
int nb_samples)
{
uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
int offset_size, p;
if (offset >= a->nb_samples)
return 0;
offset_size = offset * a->stride;
for (p = 0; p < a->planes; p++)
offset_data[p] = a->data[p] + offset_size;
return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
}
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
{
int ret;
if (a->read_only)
return AVERROR(EINVAL);
ret = ff_audio_data_realloc(a, nb_samples);
if (ret < 0)
return ret;
ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
if (ret >= 0)
a->nb_samples = ret;
return ret;
}
libavresample/audio_data.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_AUDIO_DATA_H
#define AVRESAMPLE_AUDIO_DATA_H
#include <stdint.h>
#include "libavutil/audio_fifo.h"
#include "libavutil/log.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
/**
* Audio buffer used for intermediate storage between conversion phases.
*/
typedef struct AudioData {
const AVClass *class; /**< AVClass for logging */
uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
uint8_t *buffer; /**< data buffer */
unsigned int buffer_size; /**< allocated buffer size */
int allocated_samples; /**< number of samples the buffer can hold */
int nb_samples; /**< current number of samples */
enum AVSampleFormat sample_fmt; /**< sample format */
int channels; /**< channel count */
int allocated_channels; /**< allocated channel count */
int is_planar; /**< sample format is planar */
int planes; /**< number of data planes */
int sample_size; /**< bytes per sample */
int stride; /**< sample byte offset within a plane */
int read_only; /**< data is read-only */
int allow_realloc; /**< realloc is allowed */
int ptr_align; /**< minimum data pointer alignment */
int samples_align; /**< allocated samples alignment */
const char *name; /**< name for debug logging */
} AudioData;
int ff_audio_data_set_channels(AudioData *a, int channels);
/**
* Initialize AudioData using a given source.
*
* This does not allocate an internal buffer. It only sets the data pointers
* and audio parameters.
*
* @param a AudioData struct
* @param src source data pointers
* @param plane_size plane size, in bytes.
* This can be 0 if unknown, but that will lead to
* optimized functions not being used in many cases,
* which could slow down some conversions.
* @param channels channel count
* @param nb_samples number of samples in the source data
* @param sample_fmt sample format
* @param read_only indicates if buffer is read only or read/write
* @param name name for debug logging (can be NULL)
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
int nb_samples, enum AVSampleFormat sample_fmt,
int read_only, const char *name);
/**
* Allocate AudioData.
*
* This allocates an internal buffer and sets audio parameters.
*
* @param channels channel count
* @param nb_samples number of samples to allocate space for
* @param sample_fmt sample format
* @param name name for debug logging (can be NULL)
* @return newly allocated AudioData struct, or NULL on error
*/
AudioData *ff_audio_data_alloc(int channels, int nb_samples,
enum AVSampleFormat sample_fmt,
const char *name);
/**
* Reallocate AudioData.
*
* The AudioData must have been previously allocated with ff_audio_data_alloc().
*
* @param a AudioData struct
* @param nb_samples number of samples to allocate space for
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_realloc(AudioData *a, int nb_samples);
/**
* Free AudioData.
*
* The AudioData must have been previously allocated with ff_audio_data_alloc().
*
* @param a AudioData struct
*/
void ff_audio_data_free(AudioData **a);
/**
* Copy data from one AudioData to another.
*
* @param out output AudioData
* @param in input AudioData
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_copy(AudioData *out, AudioData *in);
/**
* Append data from one AudioData to the end of another.
*
* @param dst destination AudioData
* @param dst_offset offset, in samples, to start writing, relative to the
* start of dst
* @param src source AudioData
* @param src_offset offset, in samples, to start copying, relative to the
* start of the src
* @param nb_samples number of samples to copy
* @return 0 on success, negative AVERROR value on error
*/
int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
int src_offset, int nb_samples);
/**
* Drain samples from the start of the AudioData.
*
* Remaining samples are shifted to the start of the AudioData.
*
* @param a AudioData struct
* @param nb_samples number of samples to drain
*/
void ff_audio_data_drain(AudioData *a, int nb_samples);
/**
* Add samples in AudioData to an AVAudioFifo.
*
* @param af Audio FIFO Buffer
* @param a AudioData struct
* @param offset number of samples to skip from the start of the data
* @param nb_samples number of samples to add to the FIFO
* @return number of samples actually added to the FIFO, or
* negative AVERROR code on error
*/
int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
int nb_samples);
/**
* Read samples from an AVAudioFifo to AudioData.
*
* @param af Audio FIFO Buffer
* @param a AudioData struct
* @param nb_samples number of samples to read from the FIFO
* @return number of samples actually read from the FIFO, or
* negative AVERROR code on error
*/
int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
#endif /* AVRESAMPLE_AUDIO_DATA_H */
libavresample/audio_mix.c 0 → 100644
View file @ c8af852b
/*
* 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 "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"
static const char *coeff_type_names[] = { "q6", "q15", "flt" };
void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
enum AVMixCoeffType coeff_type, int in_channels,
int out_channels, int ptr_align, int samples_align,
const char *descr, void *mix_func)
{
if (fmt == am->fmt && coeff_type == am->coeff_type &&
( in_channels == am->in_channels || in_channels == 0) &&
(out_channels == am->out_channels || out_channels == 0)) {
char chan_str[16];
am->mix = mix_func;
am->func_descr = descr;
am->ptr_align = ptr_align;
am->samples_align = samples_align;
if (ptr_align == 1 && samples_align == 1) {
am->mix_generic = mix_func;
am->func_descr_generic = descr;
} else {
am->has_optimized_func = 1;
}
if (in_channels) {
if (out_channels)
snprintf(chan_str, sizeof(chan_str), "[%d to %d] ",
in_channels, out_channels);
else
snprintf(chan_str, sizeof(chan_str), "[%d to any] ",
in_channels);
} else if (out_channels) {
snprintf(chan_str, sizeof(chan_str), "[any to %d] ",
out_channels);
}
av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] "
"[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt),
coeff_type_names[coeff_type],
(in_channels || out_channels) ? chan_str : "", descr);
}
}
#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c
#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \
static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
int len, int out_ch, int in_ch) \
{ \
int i, in, out; \
stype temp[AVRESAMPLE_MAX_CHANNELS]; \
for (i = 0; i < len; i++) { \
for (out = 0; out < out_ch; out++) { \
sumtype sum = 0; \
for (in = 0; in < in_ch; in++) \
sum += samples[in][i] * matrix[out][in]; \
temp[out] = expr; \
} \
for (out = 0; out < out_ch; out++) \
samples[out][i] = temp[out]; \
} \
}
MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum)
MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15))
MIX_FUNC_GENERIC(S16P, Q6, int16_t, int16_t, int32_t, av_clip_int16(sum >> 6))
/* TODO: templatize the channel-specific C functions */
static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float *src0 = samples[0];
float *src1 = samples[1];
float *dst = src0;
float m0 = matrix[0][0];
float m1 = matrix[0][1];
while (len > 4) {
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
*dst++ = *src0++ * m0 + *src1++ * m1;
len -= 4;
}
while (len > 0) {
*dst++ = *src0++ * m0 + *src1++ * m1;
len--;
}
}
static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *src = dst0;
float m0 = matrix[0][0];
float m1 = matrix[1][0];
while (len > 4) {
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
len -= 4;
}
while (len > 0) {
v = *src++;
*dst0++ = v * m1;
*dst1++ = v * m0;
len--;
}
}
static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *src0 = samples[0];
float *src1 = samples[1];
float *src2 = samples[2];
float *src3 = samples[3];
float *src4 = samples[4];
float *src5 = samples[5];
float *dst0 = src0;
float *dst1 = src1;
float *m0 = matrix[0];
float *m1 = matrix[1];
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * m0[0] +
v1 * m0[1] +
*src2 * m0[2] +
*src3 * m0[3] +
*src4 * m0[4] +
*src5 * m0[5];
*dst1++ = v0 * m1[0] +
v1 * m1[1] +
*src2++ * m1[2] +
*src3++ * m1[3] +
*src4++ * m1[4] +
*src5++ * m1[5];
len--;
}
}
static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len,
int out_ch, int in_ch)
{
float v0, v1;
float *dst0 = samples[0];
float *dst1 = samples[1];
float *dst2 = samples[2];
float *dst3 = samples[3];
float *dst4 = samples[4];
float *dst5 = samples[5];
float *src0 = dst0;
float *src1 = dst1;
while (len > 0) {
v0 = *src0++;
v1 = *src1++;
*dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1];
*dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1];
*dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1];
*dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1];
*dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1];
*dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1];
len--;
}
}
static int mix_function_init(AudioMix *am)
{
/* any-to-any C versions */
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15));
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q6,
0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q6));
/* channel-specific C versions */
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c);
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c);
if (ARCH_X86)
ff_audio_mix_init_x86(am);
if (!am->mix) {
av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] "
"[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt),
coeff_type_names[am->coeff_type], am->in_channels,
am->out_channels);
return AVERROR_PATCHWELCOME;
}
return 0;
}
int ff_audio_mix_init(AVAudioResampleContext *avr)
{
int ret;
/* build matrix if the user did not already set one */
if (!avr->am->matrix) {
int i, j;
char in_layout_name[128];
char out_layout_name[128];
double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels *
sizeof(*matrix_dbl));
if (!matrix_dbl)
return AVERROR(ENOMEM);
ret = avresample_build_matrix(avr->in_channel_layout,
avr->out_channel_layout,
avr->center_mix_level,
avr->surround_mix_level,
avr->lfe_mix_level, 1, matrix_dbl,
avr->in_channels);
if (ret < 0) {
av_free(matrix_dbl);
return ret;
}
av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name),
avr->in_channels, avr->in_channel_layout);
av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name),
avr->out_channels, avr->out_channel_layout);
av_log(avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n",
in_layout_name, out_layout_name);
for (i = 0; i < avr->out_channels; i++) {
for (j = 0; j < avr->in_channels; j++) {
av_log(avr, AV_LOG_DEBUG, " %0.3f ",
matrix_dbl[i * avr->in_channels + j]);
}
av_log(avr, AV_LOG_DEBUG, "\n");
}
ret = avresample_set_matrix(avr, matrix_dbl, avr->in_channels);
if (ret < 0) {
av_free(matrix_dbl);
return ret;
}
av_free(matrix_dbl);
}
avr->am->fmt = avr->internal_sample_fmt;
avr->am->coeff_type = avr->mix_coeff_type;
avr->am->in_layout = avr->in_channel_layout;
avr->am->out_layout = avr->out_channel_layout;
avr->am->in_channels = avr->in_channels;
avr->am->out_channels = avr->out_channels;
ret = mix_function_init(avr->am);
if (ret < 0)
return ret;
return 0;
}
void ff_audio_mix_close(AudioMix *am)
{
if (!am)
return;
if (am->matrix) {
av_free(am->matrix[0]);
am->matrix = NULL;
}
memset(am->matrix_q6, 0, sizeof(am->matrix_q6 ));
memset(am->matrix_q15, 0, sizeof(am->matrix_q15));
memset(am->matrix_flt, 0, sizeof(am->matrix_flt));
}
int ff_audio_mix(AudioMix *am, AudioData *src)
{
int use_generic = 1;
int len = src->nb_samples;
/* determine whether to use the optimized function based on pointer and
samples alignment in both the input and output */
if (am->has_optimized_func) {
int aligned_len = FFALIGN(len, am->samples_align);
if (!(src->ptr_align % am->ptr_align) &&
src->samples_align >= aligned_len) {
len = aligned_len;
use_generic = 0;
}
}
av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n",
src->nb_samples, am->in_channels, am->out_channels,
use_generic ? am->func_descr_generic : am->func_descr);
if (use_generic)
am->mix_generic(src->data, am->matrix, len, am->out_channels,
am->in_channels);
else
am->mix(src->data, am->matrix, len, am->out_channels, am->in_channels);
ff_audio_data_set_channels(src, am->out_channels);
return 0;
}
libavresample/audio_mix.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_AUDIO_MIX_H
#define AVRESAMPLE_AUDIO_MIX_H
#include <stdint.h>
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "audio_data.h"
typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch,
int in_ch);
typedef struct AudioMix {
AVAudioResampleContext *avr;
enum AVSampleFormat fmt;
enum AVMixCoeffType coeff_type;
uint64_t in_layout;
uint64_t out_layout;
int in_channels;
int out_channels;
int ptr_align;
int samples_align;
int has_optimized_func;
const char *func_descr;
const char *func_descr_generic;
mix_func *mix;
mix_func *mix_generic;
int16_t *matrix_q6[AVRESAMPLE_MAX_CHANNELS];
int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS];
float *matrix_flt[AVRESAMPLE_MAX_CHANNELS];
void **matrix;
} AudioMix;
/**
* Set mixing function if the parameters match.
*
* This compares the parameters of the mixing function to the parameters in the
* AudioMix context. If the parameters do not match, no changes are made to the
* active functions. If the parameters do match and the alignment is not
* constrained, the function is set as the generic mixing function. If the
* parameters match and the alignment is constrained, the function is set as
* the optimized mixing function.
*
* @param am AudioMix context
* @param fmt input/output sample format
* @param coeff_type mixing coefficient type
* @param in_channels number of input channels, or 0 for any number of channels
* @param out_channels number of output channels, or 0 for any number of channels
* @param ptr_align buffer pointer alignment, in bytes
* @param sample_align buffer size alignment, in samples
* @param descr function type description (e.g. "C" or "SSE")
* @param mix_func mixing function pointer
*/
void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt,
enum AVMixCoeffType coeff_type, int in_channels,
int out_channels, int ptr_align, int samples_align,
const char *descr, void *mix_func);
/**
* Initialize the AudioMix context in the AVAudioResampleContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* AudioMix context and set the mixing matrix.
*
* @param avr AVAudioResampleContext
* @return 0 on success, negative AVERROR code on failure
*/
int ff_audio_mix_init(AVAudioResampleContext *avr);
/**
* Close an AudioMix context.
*
* This clears and frees the mixing matrix arrays.
*/
void ff_audio_mix_close(AudioMix *am);
/**
* Apply channel mixing to audio data using the current mixing matrix.
*/
int ff_audio_mix(AudioMix *am, AudioData *src);
/* arch-specific initialization functions */
void ff_audio_mix_init_x86(AudioMix *am);
#endif /* AVRESAMPLE_AUDIO_MIX_H */
libavresample/audio_mix_matrix.c 0 → 100644
View file @ c8af852b
/*
* Copyright (C) 2011 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 "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"
/* channel positions */
#define FRONT_LEFT 0
#define FRONT_RIGHT 1
#define FRONT_CENTER 2
#define LOW_FREQUENCY 3
#define BACK_LEFT 4
#define BACK_RIGHT 5
#define FRONT_LEFT_OF_CENTER 6
#define FRONT_RIGHT_OF_CENTER 7
#define BACK_CENTER 8
#define SIDE_LEFT 9
#define SIDE_RIGHT 10
#define TOP_CENTER 11
#define TOP_FRONT_LEFT 12
#define TOP_FRONT_CENTER 13
#define TOP_FRONT_RIGHT 14
#define TOP_BACK_LEFT 15
#define TOP_BACK_CENTER 16
#define TOP_BACK_RIGHT 17
#define STEREO_LEFT 29
#define STEREO_RIGHT 30
#define WIDE_LEFT 31
#define WIDE_RIGHT 32
#define SURROUND_DIRECT_LEFT 33
#define SURROUND_DIRECT_RIGHT 34
static av_always_inline int even(uint64_t layout)
{
return (!layout || (layout & (layout - 1)));
}
static int sane_layout(uint64_t layout)
{
/* check that there is at least 1 front speaker */
if (!(layout & AV_CH_LAYOUT_SURROUND))
return 0;
/* check for left/right symmetry */
if (!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT)) ||
!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT)) ||
!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)) ||
!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)) ||
!even(layout & (AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT)) ||
!even(layout & (AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT)) ||
!even(layout & (AV_CH_STEREO_LEFT | AV_CH_STEREO_RIGHT)) ||
!even(layout & (AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT)) ||
!even(layout & (AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT)))
return 0;
return 1;
}
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize,
double *matrix_out, int stride)
{
int i, j, out_i, out_j;
double matrix[64][64] = {{0}};
int64_t unaccounted = in_layout & ~out_layout;
double maxcoef = 0;
int in_channels, out_channels;
in_channels = av_get_channel_layout_nb_channels( in_layout);
out_channels = av_get_channel_layout_nb_channels(out_layout);
memset(matrix_out, 0, out_channels * stride * sizeof(*matrix_out));
/* check if layouts are supported */
if (!in_layout || in_channels > AVRESAMPLE_MAX_CHANNELS)
return AVERROR(EINVAL);
if (!out_layout || out_channels > AVRESAMPLE_MAX_CHANNELS)
return AVERROR(EINVAL);
/* check if layouts are unbalanced or abnormal */
if (!sane_layout(in_layout) || !sane_layout(out_layout))
return AVERROR_PATCHWELCOME;
/* route matching input/output channels */
for (i = 0; i < 64; i++) {
if (in_layout & out_layout & (1ULL << i))
matrix[i][i] = 1.0;
}
/* mix front center to front left/right */
if (unaccounted & AV_CH_FRONT_CENTER) {
if ((out_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) {
matrix[FRONT_LEFT ][FRONT_CENTER] += M_SQRT1_2;
matrix[FRONT_RIGHT][FRONT_CENTER] += M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix front left/right to center */
if (unaccounted & AV_CH_LAYOUT_STEREO) {
if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][FRONT_LEFT ] += M_SQRT1_2;
matrix[FRONT_CENTER][FRONT_RIGHT] += M_SQRT1_2;
/* mix left/right/center to center */
if (in_layout & AV_CH_FRONT_CENTER)
matrix[FRONT_CENTER][FRONT_CENTER] = center_mix_level * M_SQRT2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix back center to back, side, or front */
if (unaccounted & AV_CH_BACK_CENTER) {
if (out_layout & AV_CH_BACK_LEFT) {
matrix[BACK_LEFT ][BACK_CENTER] += M_SQRT1_2;
matrix[BACK_RIGHT][BACK_CENTER] += M_SQRT1_2;
} else if (out_layout & AV_CH_SIDE_LEFT) {
matrix[SIDE_LEFT ][BACK_CENTER] += M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2;
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][BACK_CENTER] += surround_mix_level * M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix back left/right to back center, side, or front */
if (unaccounted & AV_CH_BACK_LEFT) {
if (out_layout & AV_CH_BACK_CENTER) {
matrix[BACK_CENTER][BACK_LEFT ] += M_SQRT1_2;
matrix[BACK_CENTER][BACK_RIGHT] += M_SQRT1_2;
} else if (out_layout & AV_CH_SIDE_LEFT) {
/* if side channels do not exist in the input, just copy back
channels to side channels, otherwise mix back into side */
if (in_layout & AV_CH_SIDE_LEFT) {
matrix[SIDE_LEFT ][BACK_LEFT ] += M_SQRT1_2;
matrix[SIDE_RIGHT][BACK_RIGHT] += M_SQRT1_2;
} else {
matrix[SIDE_LEFT ][BACK_LEFT ] += 1.0;
matrix[SIDE_RIGHT][BACK_RIGHT] += 1.0;
}
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][BACK_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level;
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][BACK_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_CENTER][BACK_RIGHT] += surround_mix_level * M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix side left/right into back or front */
if (unaccounted & AV_CH_SIDE_LEFT) {
if (out_layout & AV_CH_BACK_LEFT) {
/* if back channels do not exist in the input, just copy side
channels to back channels, otherwise mix side into back */
if (in_layout & AV_CH_BACK_LEFT) {
matrix[BACK_LEFT ][SIDE_LEFT ] += M_SQRT1_2;
matrix[BACK_RIGHT][SIDE_RIGHT] += M_SQRT1_2;
} else {
matrix[BACK_LEFT ][SIDE_LEFT ] += 1.0;
matrix[BACK_RIGHT][SIDE_RIGHT] += 1.0;
}
} else if (out_layout & AV_CH_BACK_CENTER) {
matrix[BACK_CENTER][SIDE_LEFT ] += M_SQRT1_2;
matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2;
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][SIDE_LEFT ] += surround_mix_level;
matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level;
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2;
matrix[FRONT_CENTER][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix left-of-center/right-of-center into front left/right or center */
if (unaccounted & AV_CH_FRONT_LEFT_OF_CENTER) {
if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][FRONT_LEFT_OF_CENTER ] += 1.0;
matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER] += 1.0;
} else if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][FRONT_LEFT_OF_CENTER ] += M_SQRT1_2;
matrix[FRONT_CENTER][FRONT_RIGHT_OF_CENTER] += M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* mix LFE into front left/right or center */
if (unaccounted & AV_CH_LOW_FREQUENCY) {
if (out_layout & AV_CH_FRONT_CENTER) {
matrix[FRONT_CENTER][LOW_FREQUENCY] += lfe_mix_level;
} else if (out_layout & AV_CH_FRONT_LEFT) {
matrix[FRONT_LEFT ][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
matrix[FRONT_RIGHT][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2;
} else
return AVERROR_PATCHWELCOME;
}
/* transfer internal matrix to output matrix and calculate maximum
per-channel coefficient sum */
for (out_i = i = 0; out_i < out_channels && i < 64; i++) {
double sum = 0;
for (out_j = j = 0; out_j < in_channels && j < 64; j++) {
matrix_out[out_i * stride + out_j] = matrix[i][j];
sum += fabs(matrix[i][j]);
if (in_layout & (1ULL << j))
out_j++;
}
maxcoef = FFMAX(maxcoef, sum);
if (out_layout & (1ULL << i))
out_i++;
}
/* normalize */
if (normalize && maxcoef > 1.0) {
for (i = 0; i < out_channels; i++)
for (j = 0; j < in_channels; j++)
matrix_out[i * stride + j] /= maxcoef;
}
return 0;
}
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride)
{
int in_channels, out_channels, i, o;
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if ( in_channels < 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
out_channels < 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
return AVERROR(EINVAL);
}
switch (avr->mix_coeff_type) {
case AV_MIX_COEFF_TYPE_Q6:
if (!avr->am->matrix_q6[0]) {
av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
return AVERROR(EINVAL);
}
for (o = 0; o < out_channels; o++)
for (i = 0; i < in_channels; i++)
matrix[o * stride + i] = avr->am->matrix_q6[o][i] / 64.0;
break;
case AV_MIX_COEFF_TYPE_Q15:
if (!avr->am->matrix_q15[0]) {
av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
return AVERROR(EINVAL);
}
for (o = 0; o < out_channels; o++)
for (i = 0; i < in_channels; i++)
matrix[o * stride + i] = avr->am->matrix_q15[o][i] / 32768.0;
break;
case AV_MIX_COEFF_TYPE_FLT:
if (!avr->am->matrix_flt[0]) {
av_log(avr, AV_LOG_ERROR, "matrix is not set\n");
return AVERROR(EINVAL);
}
for (o = 0; o < out_channels; o++)
for (i = 0; i < in_channels; i++)
matrix[o * stride + i] = avr->am->matrix_flt[o][i];
break;
default:
av_log(avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
return 0;
}
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride)
{
int in_channels, out_channels, i, o;
in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if ( in_channels < 0 || in_channels > AVRESAMPLE_MAX_CHANNELS ||
out_channels < 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n");
return AVERROR(EINVAL);
}
if (avr->am->matrix)
av_freep(avr->am->matrix);
#define CONVERT_MATRIX(type, expr) \
avr->am->matrix_## type[0] = av_mallocz(out_channels * in_channels * \
sizeof(*avr->am->matrix_## type[0])); \
if (!avr->am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0; o < out_channels; o++) { \
if (o > 0) \
avr->am->matrix_## type[o] = avr->am->matrix_## type[o - 1] + \
in_channels; \
for (i = 0; i < in_channels; i++) { \
double v = matrix[o * stride + i]; \
avr->am->matrix_## type[o][i] = expr; \
} \
} \
avr->am->matrix = (void **)avr->am->matrix_## type;
switch (avr->mix_coeff_type) {
case AV_MIX_COEFF_TYPE_Q6:
CONVERT_MATRIX(q6, av_clip_int16(lrint(64.0 * v)))
break;
case AV_MIX_COEFF_TYPE_Q15:
CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v)))
break;
case AV_MIX_COEFF_TYPE_FLT:
CONVERT_MATRIX(flt, v)
break;
default:
av_log(avr, AV_LOG_ERROR, "Invalid mix coeff type\n");
return AVERROR(EINVAL);
}
/* TODO: detect situations where we can just swap around pointers
instead of doing matrix multiplications with 0.0 and 1.0 */
return 0;
}
libavresample/avresample-test.c 0 → 100644
View file @ c8af852b
/*
* Copyright (c) 2002 Fabrice Bellard
* 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 <stdio.h>
#include "libavutil/avstring.h"
#include "libavutil/lfg.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
static double dbl_rand(AVLFG *lfg)
{
return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
}
#define PUT_FUNC(name, fmt, type, expr) \
static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
int channels, int sample, int ch, \
double v_dbl) \
{ \
type v = expr; \
type **out = (type **)data; \
if (av_sample_fmt_is_planar(sample_fmt)) \
out[ch][sample] = v; \
else \
out[0][sample * channels + ch] = v; \
}
PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
static void put_sample(void **data, enum AVSampleFormat sample_fmt,
int channels, int sample, int ch, double v_dbl)
{
switch (av_get_packed_sample_fmt(sample_fmt)) {
case AV_SAMPLE_FMT_U8:
put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S16:
put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_S32:
put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_FLT:
put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
break;
case AV_SAMPLE_FMT_DBL:
put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
break;
}
}
static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
int channels, int sample_rate, int nb_samples)
{
int i, ch, k;
double v, f, a, ampa;
double tabf1[AVRESAMPLE_MAX_CHANNELS];
double tabf2[AVRESAMPLE_MAX_CHANNELS];
double taba[AVRESAMPLE_MAX_CHANNELS];
#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
k = 0;
/* 1 second of single freq sinus at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
a += M_PI * 1000.0 * 2.0 / sample_rate;
}
/* 1 second of varing frequency between 100 and 10000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
a += M_PI * f * 2.0 / sample_rate;
}
/* 0.5 second of low amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd);
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 1 second of unrelated ramps for each channel */
for (ch = 0; ch < channels; ch++) {
taba[ch] = 0;
tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
}
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
v = sin(taba[ch]) * 0.30;
PUT_SAMPLE
f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
taba[ch] += M_PI * f * 2.0 / sample_rate;
}
}
/* 2 seconds of 500 Hz with varying volume */
a = 0;
ampa = 0;
for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
double amp = (1.0 + sin(ampa)) * 0.15;
if (ch & 1)
amp = 0.30 - amp;
v = sin(a) * amp;
PUT_SAMPLE
a += M_PI * 500.0 * 2.0 / sample_rate;
ampa += M_PI * 2.0 / sample_rate;
}
}
}
/* formats, rates, and layouts are ordered for priority in testing.
e.g. 'avresample-test 4 2 2' will test all input/output combinations of
S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
static const enum AVSampleFormat formats[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_DBL,
};
static const int rates[] = {
48000,
44100,
16000
};
static const uint64_t layouts[] = {
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_5POINT1,
AV_CH_LAYOUT_7POINT1,
};
int main(int argc, char **argv)
{
AVAudioResampleContext *s;
AVLFG rnd;
int ret = 0;
uint8_t *in_buf = NULL;
uint8_t *out_buf = NULL;
unsigned int in_buf_size;
unsigned int out_buf_size;
uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
int in_linesize;
int out_linesize;
uint64_t in_ch_layout;
int in_channels;
enum AVSampleFormat in_fmt;
int in_rate;
uint64_t out_ch_layout;
int out_channels;
enum AVSampleFormat out_fmt;
int out_rate;
int num_formats, num_rates, num_layouts;
int i, j, k, l, m, n;
num_formats = 2;
num_rates = 2;
num_layouts = 2;
if (argc > 1) {
if (!av_strncasecmp(argv[1], "-h", 3)) {
av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
"[<num sample rates> [<num channel layouts>]]]\n"
"Default is 2 2 2\n");
return 0;
}
num_formats = strtol(argv[1], NULL, 0);
num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
}
if (argc > 2) {
num_rates = strtol(argv[2], NULL, 0);
num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
}
if (argc > 3) {
num_layouts = strtol(argv[3], NULL, 0);
num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
}
av_log_set_level(AV_LOG_DEBUG);
av_lfg_init(&rnd, 0xC0FFEE);
in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
AV_SAMPLE_FMT_DBLP, 0);
out_buf_size = in_buf_size;
in_buf = av_malloc(in_buf_size);
if (!in_buf)
goto end;
out_buf = av_malloc(out_buf_size);
if (!out_buf)
goto end;
s = avresample_alloc_context();
if (!s) {
av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
ret = 1;
goto end;
}
for (i = 0; i < num_formats; i++) {
in_fmt = formats[i];
for (k = 0; k < num_layouts; k++) {
in_ch_layout = layouts[k];
in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
for (m = 0; m < num_rates; m++) {
in_rate = rates[m];
ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
in_channels, in_rate * 6,
in_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
goto end;
}
audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
for (j = 0; j < num_formats; j++) {
out_fmt = formats[j];
for (l = 0; l < num_layouts; l++) {
out_ch_layout = layouts[l];
out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
for (n = 0; n < num_rates; n++) {
out_rate = rates[n];
av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
in_channels, out_channels, in_rate, out_rate);
ret = av_samples_fill_arrays(out_data, &out_linesize,
out_buf, out_channels,
out_rate * 6, out_fmt, 0);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
goto end;
}
av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
av_opt_set_int(s, "in_sample_rate", in_rate, 0);
av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
av_opt_set_int(s, "out_sample_rate", out_rate, 0);
av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
ret = avresample_open(s);
if (ret < 0) {
av_log(s, AV_LOG_ERROR, "Error opening context\n");
goto end;
}
ret = avresample_convert(s, (void **)out_data, out_linesize, out_rate * 6,
(void **) in_data, in_linesize, in_rate * 6);
if (ret < 0) {
char errbuf[256];
av_strerror(ret, errbuf, sizeof(errbuf));
av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
goto end;
}
av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
in_rate * 6, ret);
if (avresample_get_delay(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
avresample_get_delay(s));
if (avresample_available(s) > 0)
av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
avresample_available(s));
av_log(NULL, AV_LOG_INFO, "\n");
avresample_close(s);
}
}
}
}
}
}
ret = 0;
end:
av_freep(&in_buf);
av_freep(&out_buf);
avresample_free(&s);
return ret;
}
libavresample/avresample.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_AVRESAMPLE_H
#define AVRESAMPLE_AVRESAMPLE_H
/**
* @file
* external API header
*/
#include "libavutil/audioconvert.h"
#include "libavutil/avutil.h"
#include "libavutil/dict.h"
#include "libavutil/log.h"
#include "libavresample/version.h"
#define AVRESAMPLE_MAX_CHANNELS 32
typedef struct AVAudioResampleContext AVAudioResampleContext;
/** Mixing Coefficient Types */
enum AVMixCoeffType {
AV_MIX_COEFF_TYPE_Q6, /** 16-bit 10.6 fixed-point */
AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */
AV_MIX_COEFF_TYPE_FLT, /** floating-point */
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/
unsigned avresample_version(void);
/**
* Return the libavresample build-time configuration.
* @return configure string
*/
const char *avresample_configuration(void);
/**
* Return the libavresample license.
*/
const char *avresample_license(void);
/**
* Get the AVClass for AVAudioResampleContext.
*
* Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
* without allocating a context.
*
* @see av_opt_find().
*
* @return AVClass for AVAudioResampleContext
*/
const AVClass *avresample_get_class(void);
/**
* Allocate AVAudioResampleContext and set options.
*
* @return allocated audio resample context, or NULL on failure
*/
AVAudioResampleContext *avresample_alloc_context(void);
/**
* Initialize AVAudioResampleContext.
*
* @param avr audio resample context
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_open(AVAudioResampleContext *avr);
/**
* Close AVAudioResampleContext.
*
* This closes the context, but it does not change the parameters. The context
* can be reopened with avresample_open(). It does, however, clear the output
* FIFO and any remaining leftover samples in the resampling delay buffer. If
* there was a custom matrix being used, that is also cleared.
*
* @see avresample_convert()
* @see avresample_set_matrix()
*
* @param avr audio resample context
*/
void avresample_close(AVAudioResampleContext *avr);
/**
* Free AVAudioResampleContext and associated AVOption values.
*
* This also calls avresample_close() before freeing.
*
* @param avr audio resample context
*/
void avresample_free(AVAudioResampleContext **avr);
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libavresample for building the
* default mixing matrix. It is made public just as a utility function for
* building custom matrices.
*
* @param in_layout input channel layout
* @param out_layout output channel layout
* @param center_mix_level mix level for the center channel
* @param surround_mix_level mix level for the surround channel(s)
* @param lfe_mix_level mix level for the low-frequency effects channel
* @param normalize if 1, coefficients will be normalized to prevent
* overflow. if 0, coefficients will not be
* normalized.
* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, int normalize, double *matrix,
int stride);
/**
* Get the current channel mixing matrix.
*
* @param avr audio resample context
* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
* input channel i in output channel o.
* @param stride distance between adjacent input channels in the matrix array
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
int stride);
/**
* Set channel mixing matrix.
*
* Allows for setting a custom mixing matrix, overriding the default matrix
* generated internally during avresample_open(). This function can be called
* anytime on an allocated context, either before or after calling
* avresample_open(). avresample_convert() always uses the current matrix.
* Calling avresample_close() on the context will clear the current matrix.
*
* @see avresample_close()
*
* @param avr audio resample context
* @param matrix mixing coefficients; matrix[i + stride * o] is the weight of
* input channel i in output channel o.
* @param stride distance between adjacent input channels in the matrix array
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
int stride);
/**
* Set compensation for resampling.
*
* This can be called anytime after avresample_open(). If resampling was not
* being done previously, the AVAudioResampleContext is closed and reopened
* with resampling enabled. In this case, any samples remaining in the output
* FIFO and the current channel mixing matrix will be restored after reopening
* the context.
*
* @param avr audio resample context
* @param sample_delta compensation delta, in samples
* @param compensation_distance compensation distance, in samples
* @return 0 on success, negative AVERROR code on failure
*/
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
int compensation_distance);
/**
* Convert input samples and write them to the output FIFO.
*
* The output data can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* or to avresample_read().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. avresample_get_delay() tells the number of remaining
* samples. To get this data as output, call avresample_convert() with NULL
* input.
*
* At the end of the conversion process, there may be data remaining in the
* internal FIFO buffer. avresample_available() tells the number of remaining
* samples. To get this data as output, either call avresample_convert() with
* NULL input or call avresample_read().
*
* @see avresample_available()
* @see avresample_read()
* @see avresample_get_delay()
*
* @param avr audio resample context
* @param output output data pointers
* @param out_plane_size output plane size, in bytes.
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* output, which could slow down some conversions.
* @param out_samples maximum number of samples that the output buffer can hold
* @param input input data pointers
* @param in_plane_size input plane size, in bytes
* This can be 0 if unknown, but that will lead to
* optimized functions not being used directly on the
* input, which could slow down some conversions.
* @param in_samples number of input samples to convert
* @return number of samples written to the output buffer,
* not including converted samples added to the internal
* output FIFO
*/
int avresample_convert(AVAudioResampleContext *avr, void **output,
int out_plane_size, int out_samples, void **input,
int in_plane_size, int in_samples);
/**
* Return the number of samples currently in the resampling delay buffer.
*
* When resampling, there may be a delay between the input and output. Any
* unconverted samples in each call are stored internally in a delay buffer.
* This function allows the user to determine the current number of samples in
* the delay buffer, which can be useful for synchronization.
*
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples currently in the resampling delay buffer
*/
int avresample_get_delay(AVAudioResampleContext *avr);
/**
* Return the number of available samples in the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. The samples in the FIFO can be read with avresample_read() or
* avresample_convert().
*
* @see avresample_read()
* @see avresample_convert()
*
* @param avr audio resample context
* @return number of samples available for reading
*/
int avresample_available(AVAudioResampleContext *avr);
/**
* Read samples from the output FIFO.
*
* During conversion, if the user does not specify an output buffer or
* specifies an output buffer that is smaller than what is needed, remaining
* samples that are not written to the output are stored to an internal FIFO
* buffer. This function can be used to read samples from that internal FIFO.
*
* @see avresample_available()
* @see avresample_convert()
*
* @param avr audio resample context
* @param output output data pointers
* @param nb_samples number of samples to read from the FIFO
* @return the number of samples written to output
*/
int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples);
#endif /* AVRESAMPLE_AVRESAMPLE_H */
libavresample/internal.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_INTERNAL_H
#define AVRESAMPLE_INTERNAL_H
#include "libavutil/audio_fifo.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "audio_convert.h"
#include "audio_data.h"
#include "audio_mix.h"
#include "resample.h"
struct AVAudioResampleContext {
const AVClass *av_class; /**< AVClass for logging and AVOptions */
uint64_t in_channel_layout; /**< input channel layout */
enum AVSampleFormat in_sample_fmt; /**< input sample format */
int in_sample_rate; /**< input sample rate */
uint64_t out_channel_layout; /**< output channel layout */
enum AVSampleFormat out_sample_fmt; /**< output sample format */
int out_sample_rate; /**< output sample rate */
enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
double center_mix_level; /**< center mix level */
double surround_mix_level; /**< surround mix level */
double lfe_mix_level; /**< lfe mix level */
int force_resampling; /**< force resampling */
int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
int in_channels; /**< number of input channels */
int out_channels; /**< number of output channels */
int resample_channels; /**< number of channels used for resampling */
int downmix_needed; /**< downmixing is needed */
int upmix_needed; /**< upmixing is needed */
int mixing_needed; /**< either upmixing or downmixing is needed */
int resample_needed; /**< resampling is needed */
int in_convert_needed; /**< input sample format conversion is needed */
int out_convert_needed; /**< output sample format conversion is needed */
AudioData *in_buffer; /**< buffer for converted input */
AudioData *resample_out_buffer; /**< buffer for output from resampler */
AudioData *out_buffer; /**< buffer for converted output */
AVAudioFifo *out_fifo; /**< FIFO for output samples */
AudioConvert *ac_in; /**< input sample format conversion context */
AudioConvert *ac_out; /**< output sample format conversion context */
ResampleContext *resample; /**< resampling context */
AudioMix *am; /**< channel mixing context */
};
#endif /* AVRESAMPLE_INTERNAL_H */
libavresample/libavresample.v 0 → 100644
View file @ c8af852b
LIBAVRESAMPLE_$MAJOR {
global: av*;
local: *;
};
libavresample/options.c 0 → 100644
View file @ c8af852b
/*
* 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 "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_mix.h"
/**
* @file
* Options definition for AVAudioResampleContext.
*/
#define OFFSET(x) offsetof(AVAudioResampleContext, x)
#define PARAM AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[] = {
{ "in_channel_layout", "Input Channel Layout", OFFSET(in_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "in_sample_fmt", "Input Sample Format", OFFSET(in_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "in_sample_rate", "Input Sample Rate", OFFSET(in_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM },
{ "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM },
{ "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_FLTP }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM },
{ "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q6, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
{ "q6", "16-bit 10.6 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q6 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "flt", "Floating-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "center_mix_level", "Center Mix Level", OFFSET(center_mix_level), AV_OPT_TYPE_DOUBLE, { M_SQRT1_2 }, -32.0, 32.0, PARAM },
{ "surround_mix_level", "Surround Mix Level", OFFSET(surround_mix_level), AV_OPT_TYPE_DOUBLE, { M_SQRT1_2 }, -32.0, 32.0, PARAM },
{ "lfe_mix_level", "LFE Mix Level", OFFSET(lfe_mix_level), AV_OPT_TYPE_DOUBLE, { 0.0 }, -32.0, 32.0, PARAM },
{ "force_resampling", "Force Resampling", OFFSET(force_resampling), AV_OPT_TYPE_INT, { 0 }, 0, 1, PARAM },
{ "filter_size", "Resampling Filter Size", OFFSET(filter_size), AV_OPT_TYPE_INT, { 16 }, 0, 32, /* ??? */ PARAM },
{ "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { 10 }, 0, 30, /* ??? */ PARAM },
{ "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { 0 }, 0, 1, PARAM },
{ "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { 0.8 }, 0.0, 1.0, PARAM },
{ NULL },
};
static const AVClass av_resample_context_class = {
.class_name = "AVAudioResampleContext",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVAudioResampleContext *avresample_alloc_context(void)
{
AVAudioResampleContext *avr;
avr = av_mallocz(sizeof(*avr));
if (!avr)
return NULL;
avr->av_class = &av_resample_context_class;
av_opt_set_defaults(avr);
avr->am = av_mallocz(sizeof(*avr->am));
if (!avr->am) {
av_free(avr);
return NULL;
}
avr->am->avr = avr;
return avr;
}
const AVClass *avresample_get_class(void)
{
return &av_resample_context_class;
}
libavresample/resample.c 0 → 100644
View file @ c8af852b
/*
* Copyright (c) 2004 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 "libavutil/libm.h"
#include "libavutil/log.h"
#include "internal.h"
#include "audio_data.h"
#ifdef CONFIG_RESAMPLE_FLT
/* float template */
#define FILTER_SHIFT 0
#define FELEM float
#define FELEM2 float
#define FELEML float
#define WINDOW_TYPE 24
#elifdef CONFIG_RESAMPLE_S32
/* s32 template */
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
/* s16 template */
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#endif
struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
double factor;
};
/**
* 0th order modified bessel function of the first kind.
*/
static double bessel(double x)
{
double v = 1;
double lastv = 0;
double t = 1;
int i;
x = x * x / 4;
for (i = 1; v != lastv; i++) {
lastv = v;
t *= x / (i * i);
v += t;
}
return v;
}
/**
* Build a polyphase filterbank.
*
* @param[out] filter filter coefficients
* @param factor resampling factor
* @param tap_count tap count
* @param phase_count phase count
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic
* 1->blackman nuttall windowed sinc
* 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative AVERROR code on failure
*/
static int build_filter(FELEM *filter, double factor, int tap_count,
int phase_count, int scale, int type)
{
int ph, i;
double x, y, w;
double *tab;
const int center = (tap_count - 1) / 2;
tab = av_malloc(tap_count * sizeof(*tab));
if (!tab)
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
for (i = 0; i < tap_count; i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch (type) {
case 0: {
const float d = -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
break;
}
case 1:
w = 2.0 * x / (factor * tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos( w) +
0.1365995 * cos(2 * w) -
0.0106411 * cos(3 * w);
break;
default:
w = 2.0 * x / (factor * tap_count * M_PI);
y *= bessel(type * sqrt(FFMAX(1 - w * w, 0)));
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for (i = 0; i < tap_count; i++) {
#ifdef CONFIG_RESAMPLE_FLT
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
FELEM_MIN, FELEM_MAX);
#endif
}
}
av_free(tab);
return 0;
}
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
{
ResampleContext *c;
int out_rate = avr->out_sample_rate;
int in_rate = avr->in_sample_rate;
double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
int phase_count = 1 << avr->phase_shift;
/* TODO: add support for s32 and float internal formats */
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"resampling: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
return NULL;
}
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
c->avr = avr;
c->phase_shift = avr->phase_shift;
c->phase_mask = phase_count - 1;
c->linear = avr->linear_interp;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
goto error;
memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
c->compensation_distance = 0;
if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
in_rate * (int64_t)phase_count, INT32_MAX / 2))
goto error;
c->ideal_dst_incr = c->dst_incr;
c->index = -phase_count * ((c->filter_length - 1) / 2);
c->frac = 0;
/* allocate internal buffer */
c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
avr->internal_sample_fmt,
"resample buffer");
if (!c->buffer)
goto error;
av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
av_get_sample_fmt_name(avr->internal_sample_fmt),
avr->in_sample_rate, avr->out_sample_rate);
return c;
error:
ff_audio_data_free(&c->buffer);
av_free(c->filter_bank);
av_free(c);
return NULL;
}
void ff_audio_resample_free(ResampleContext **c)
{
if (!*c)
return;
ff_audio_data_free(&(*c)->buffer);
av_free((*c)->filter_bank);
av_freep(c);
}
int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
int compensation_distance)
{
ResampleContext *c;
AudioData *fifo_buf = NULL;
int ret = 0;
if (compensation_distance < 0)
return AVERROR(EINVAL);
if (!compensation_distance && sample_delta)
return AVERROR(EINVAL);
/* if resampling was not enabled previously, re-initialize the
AVAudioResampleContext and force resampling */
if (!avr->resample_needed) {
int fifo_samples;
double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
/* buffer any remaining samples in the output FIFO before closing */
fifo_samples = av_audio_fifo_size(avr->out_fifo);
if (fifo_samples > 0) {
fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
avr->out_sample_fmt, NULL);
if (!fifo_buf)
return AVERROR(EINVAL);
ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
fifo_samples);
if (ret < 0)
goto reinit_fail;
}
/* save the channel mixing matrix */
ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
if (ret < 0)
goto reinit_fail;
/* close the AVAudioResampleContext */
avresample_close(avr);
avr->force_resampling = 1;
/* restore the channel mixing matrix */
ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
if (ret < 0)
goto reinit_fail;
/* re-open the AVAudioResampleContext */
ret = avresample_open(avr);
if (ret < 0)
goto reinit_fail;
/* restore buffered samples to the output FIFO */
if (fifo_samples > 0) {
ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
fifo_samples);
if (ret < 0)
goto reinit_fail;
ff_audio_data_free(&fifo_buf);
}
}
c = avr->resample;
c->compensation_distance = compensation_distance;
if (compensation_distance) {
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
(int64_t)sample_delta / compensation_distance;
} else {
c->dst_incr = c->ideal_dst_incr;
}
return 0;
reinit_fail:
ff_audio_data_free(&fifo_buf);
return ret;
}
static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
int *consumed, int src_size, int dst_size, int update_ctx)
{
int dst_index, i;
int index = c->index;
int frac = c->frac;
int dst_incr_frac = c->dst_incr % c->src_incr;
int dst_incr = c->dst_incr / c->src_incr;
int compensation_distance = c->compensation_distance;
if (!dst != !src)
return AVERROR(EINVAL);
if (compensation_distance == 0 && c->filter_length == 1 &&
c->phase_shift == 0) {
int64_t index2 = ((int64_t)index) << 32;
int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
dst_size = FFMIN(dst_size,
(src_size-1-index) * (int64_t)c->src_incr /
c->dst_incr);
if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
dst[dst_index] = src[index2 >> 32];
index2 += incr;
}
} else {
dst_index = dst_size;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
} else {
for (dst_index = 0; dst_index < dst_size; dst_index++) {
FELEM *filter = c->filter_bank +
c->filter_length * (index & c->phase_mask);
int sample_index = index >> c->phase_shift;
if (!dst && (sample_index + c->filter_length > src_size ||
-sample_index >= src_size))
break;
if (dst) {
FELEM2 val = 0;
if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (sample_index + c->filter_length > src_size) {
break;
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}
#ifdef CONFIG_RESAMPLE_FLT
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = av_clip_int16(val);
#endif
}
frac += dst_incr_frac;
index += dst_incr;
if (frac >= c->src_incr) {
frac -= c->src_incr;
index++;
}
if (dst_index + 1 == compensation_distance) {
compensation_distance = 0;
dst_incr_frac = c->ideal_dst_incr % c->src_incr;
dst_incr = c->ideal_dst_incr / c->src_incr;
}
}
}
if (consumed)
*consumed = FFMAX(index, 0) >> c->phase_shift;
if (update_ctx) {
if (index >= 0)
index &= c->phase_mask;
if (compensation_distance) {
compensation_distance -= dst_index;
if (compensation_distance <= 0)
return AVERROR_BUG;
}
c->frac = frac;
c->index = index;
c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance = compensation_distance;
}
return dst_index;
}
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
int *consumed)
{
int ch, in_samples, in_leftover, out_samples = 0;
int ret = AVERROR(EINVAL);
in_samples = src ? src->nb_samples : 0;
in_leftover = c->buffer->nb_samples;
/* add input samples to the internal buffer */
if (src) {
ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
if (ret < 0)
return ret;
} else if (!in_leftover) {
/* no remaining samples to flush */
return 0;
} else {
/* TODO: pad buffer to flush completely */
}
/* calculate output size and reallocate output buffer if needed */
/* TODO: try to calculate this without the dummy resample() run */
if (!dst->read_only && dst->allow_realloc) {
out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
INT_MAX, 0);
ret = ff_audio_data_realloc(dst, out_samples);
if (ret < 0) {
av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
return ret;
}
}
/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
out_samples = resample(c, (int16_t *)dst->data[ch],
(const int16_t *)c->buffer->data[ch], consumed,
c->buffer->nb_samples, dst->allocated_samples,
ch + 1 == c->buffer->channels);
}
if (out_samples < 0) {
av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
return out_samples;
}
/* drain consumed samples from the internal buffer */
ff_audio_data_drain(c->buffer, *consumed);
av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
in_samples, in_leftover, out_samples, c->buffer->nb_samples);
dst->nb_samples = out_samples;
return 0;
}
int avresample_get_delay(AVAudioResampleContext *avr)
{
if (!avr->resample_needed || !avr->resample)
return 0;
return avr->resample->buffer->nb_samples;
}
libavresample/resample.h 0 → 100644
View file @ c8af852b
/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* 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
*/
#ifndef AVRESAMPLE_RESAMPLE_H
#define AVRESAMPLE_RESAMPLE_H
#include "avresample.h"
#include "audio_data.h"
typedef struct ResampleContext ResampleContext;
/**
* Allocate and initialize a ResampleContext.
*
* The parameters in the AVAudioResampleContext are used to initialize the
* ResampleContext.
*
* @param avr AVAudioResampleContext
* @return newly-allocated ResampleContext
*/
ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr);
/**
* Free a ResampleContext.
*
* @param c ResampleContext
*/
void ff_audio_resample_free(ResampleContext **c);
/**
* Resample audio data.
*
* Changes the sample rate.
*
* @par
* All samples in the source data may not be consumed depending on the
* resampling parameters and the size of the output buffer. The unconsumed
* samples are automatically added to the start of the source in the next call.
* If the destination data can be reallocated, that may be done in this function
* in order to fit all available output. If it cannot be reallocated, fewer
* input samples will be consumed in order to have the output fit in the
* destination data buffers.
*
* @param c ResampleContext
* @param dst destination audio data
* @param src source audio data
* @param consumed number of samples consumed from the source
* @return number of samples written to the destination
*/
int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
int *consumed);
#endif /* AVRESAMPLE_RESAMPLE_H */
libavresample/utils.c 0 → 100644
View file @ c8af852b
/*
* 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 "libavutil/dict.h"
#include "libavutil/error.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avresample.h"
#include "audio_data.h"
#include "internal.h"
int avresample_open(AVAudioResampleContext *avr)
{
int ret;
/* set channel mixing parameters */
avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
avr->in_channel_layout);
return AVERROR(EINVAL);
}
avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
avr->out_channel_layout);
return AVERROR(EINVAL);
}
avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
avr->downmix_needed = avr->in_channels > avr->out_channels;
avr->upmix_needed = avr->out_channels > avr->in_channels ||
avr->am->matrix ||
(avr->out_channels == avr->in_channels &&
avr->in_channel_layout != avr->out_channel_layout);
avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
/* set resampling parameters */
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
/* set sample format conversion parameters */
/* override user-requested internal format to avoid unexpected failures
TODO: support more internal formats */
if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)
avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
avr->in_sample_fmt != avr->internal_sample_fmt;
if (avr->resample_needed || avr->mixing_needed)
avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
else
avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
/* allocate buffers */
if (avr->mixing_needed || avr->in_convert_needed) {
avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
0, avr->internal_sample_fmt,
"in_buffer");
if (!avr->in_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->resample_needed) {
avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
0, avr->internal_sample_fmt,
"resample_out_buffer");
if (!avr->resample_out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
if (avr->out_convert_needed) {
avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
avr->out_sample_fmt, "out_buffer");
if (!avr->out_buffer) {
ret = AVERROR(EINVAL);
goto error;
}
}
avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
1024);
if (!avr->out_fifo) {
ret = AVERROR(ENOMEM);
goto error;
}
/* setup contexts */
if (avr->in_convert_needed) {
avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
avr->in_sample_fmt, avr->in_channels);
if (!avr->ac_in) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->out_convert_needed) {
enum AVSampleFormat src_fmt;
if (avr->in_convert_needed)
src_fmt = avr->internal_sample_fmt;
else
src_fmt = avr->in_sample_fmt;
avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
avr->out_channels);
if (!avr->ac_out) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->resample_needed) {
avr->resample = ff_audio_resample_init(avr);
if (!avr->resample) {
ret = AVERROR(ENOMEM);
goto error;
}
}
if (avr->mixing_needed) {
ret = ff_audio_mix_init(avr);
if (ret < 0)
goto error;
}
return 0;
error:
avresample_close(avr);
return ret;
}
void avresample_close(AVAudioResampleContext *avr)
{
ff_audio_data_free(&avr->in_buffer);
ff_audio_data_free(&avr->resample_out_buffer);
ff_audio_data_free(&avr->out_buffer);
av_audio_fifo_free(avr->out_fifo);
avr->out_fifo = NULL;
av_freep(&avr->ac_in);
av_freep(&avr->ac_out);
ff_audio_resample_free(&avr->resample);
ff_audio_mix_close(avr->am);
return;
}
void avresample_free(AVAudioResampleContext **avr)
{
if (!*avr)
return;
avresample_close(*avr);
av_freep(&(*avr)->am);
av_opt_free(*avr);
av_freep(avr);
}
static int handle_buffered_output(AVAudioResampleContext *avr,
AudioData *output, AudioData *converted)
{
int ret;
if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
(converted && output->allocated_samples < converted->nb_samples)) {
if (converted) {
/* if there are any samples in the output FIFO or if the
user-supplied output buffer is not large enough for all samples,
we add to the output FIFO */
av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
converted->nb_samples);
if (ret < 0)
return ret;
}
/* if the user specified an output buffer, read samples from the output
FIFO to the user output */
if (output && output->allocated_samples > 0) {
av_dlog(avr, "[FIFO] read from out_fifo to output\n");
av_dlog(avr, "[end conversion]\n");
return ff_audio_data_read_from_fifo(avr->out_fifo, output,
output->allocated_samples);
}
} else if (converted) {
/* copy directly to output if it is large enough or there is not any
data in the output FIFO */
av_dlog(avr, "[copy] %s to output\n", converted->name);
output->nb_samples = 0;
ret = ff_audio_data_copy(output, converted);
if (ret < 0)
return ret;
av_dlog(avr, "[end conversion]\n");
return output->nb_samples;
}
av_dlog(avr, "[end conversion]\n");
return 0;
}
int avresample_convert(AVAudioResampleContext *avr, void **output,
int out_plane_size, int out_samples, void **input,
int in_plane_size, int in_samples)
{
AudioData input_buffer;
AudioData output_buffer;
AudioData *current_buffer;
int ret;
/* reset internal buffers */
if (avr->in_buffer) {
avr->in_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->in_buffer,
avr->in_buffer->allocated_channels);
}
if (avr->resample_out_buffer) {
avr->resample_out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->resample_out_buffer,
avr->resample_out_buffer->allocated_channels);
}
if (avr->out_buffer) {
avr->out_buffer->nb_samples = 0;
ff_audio_data_set_channels(avr->out_buffer,
avr->out_buffer->allocated_channels);
}
av_dlog(avr, "[start conversion]\n");
/* initialize output_buffer with output data */
if (output) {
ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
avr->out_channels, out_samples,
avr->out_sample_fmt, 0, "output");
if (ret < 0)
return ret;
output_buffer.nb_samples = 0;
}
if (input) {
/* initialize input_buffer with input data */
ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
avr->in_channels, in_samples,
avr->in_sample_fmt, 1, "input");
if (ret < 0)
return ret;
current_buffer = &input_buffer;
if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
!avr->out_convert_needed && output && out_samples >= in_samples) {
/* in some rare cases we can copy input to output and upmix
directly in the output buffer */
av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
ret = ff_audio_data_copy(&output_buffer, current_buffer);
if (ret < 0)
return ret;
current_buffer = &output_buffer;
} else if (avr->mixing_needed || avr->in_convert_needed) {
/* if needed, copy or convert input to in_buffer, and downmix if
applicable */
if (avr->in_convert_needed) {
ret = ff_audio_data_realloc(avr->in_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
} else {
av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
if (ret < 0)
return ret;
}
ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
if (avr->downmix_needed) {
av_dlog(avr, "[downmix] in_buffer\n");
ret = ff_audio_mix(avr->am, avr->in_buffer);
if (ret < 0)
return ret;
}
current_buffer = avr->in_buffer;
}
} else {
/* flush resampling buffer and/or output FIFO if input is NULL */
if (!avr->resample_needed)
return handle_buffered_output(avr, output ? &output_buffer : NULL,
NULL);
current_buffer = NULL;
}
if (avr->resample_needed) {
AudioData *resample_out;
int consumed = 0;
if (!avr->out_convert_needed && output && out_samples > 0)
resample_out = &output_buffer;
else
resample_out = avr->resample_out_buffer;
av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
resample_out->name);
ret = ff_audio_resample(avr->resample, resample_out,
current_buffer, &consumed);
if (ret < 0)
return ret;
/* if resampling did not produce any samples, just return 0 */
if (resample_out->nb_samples == 0) {
av_dlog(avr, "[end conversion]\n");
return 0;
}
current_buffer = resample_out;
}
if (avr->upmix_needed) {
av_dlog(avr, "[upmix] %s\n", current_buffer->name);
ret = ff_audio_mix(avr->am, current_buffer);
if (ret < 0)
return ret;
}
/* if we resampled or upmixed directly to output, return here */
if (current_buffer == &output_buffer) {
av_dlog(avr, "[end conversion]\n");
return current_buffer->nb_samples;
}
if (avr->out_convert_needed) {
if (output && out_samples >= current_buffer->nb_samples) {
/* convert directly to output */
av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[end conversion]\n");
return output_buffer.nb_samples;
} else {
ret = ff_audio_data_realloc(avr->out_buffer,
current_buffer->nb_samples);
if (ret < 0)
return ret;
av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
current_buffer, current_buffer->nb_samples);
if (ret < 0)
return ret;
current_buffer = avr->out_buffer;
}
}
return handle_buffered_output(avr, &output_buffer, current_buffer);
}
int avresample_available(AVAudioResampleContext *avr)
{
return av_audio_fifo_size(avr->out_fifo);
}
int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
{
return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
}
unsigned avresample_version(void)
{
return LIBAVRESAMPLE_VERSION_INT;
}
const char *avresample_license(void)
{
#define LICENSE_PREFIX "libavresample license: "
return LICENSE_PREFIX LIBAV_LICENSE + sizeof(LICENSE_PREFIX) - 1;
}
const char *avresample_configuration(void)
{
return LIBAV_CONFIGURATION;
}
libavresample/version.h 0 → 100644
View file @ c8af852b
/*
* 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
*/
#ifndef AVRESAMPLE_VERSION_H
#define AVRESAMPLE_VERSION_H
#define LIBAVRESAMPLE_VERSION_MAJOR 0
#define LIBAVRESAMPLE_VERSION_MINOR 0
#define LIBAVRESAMPLE_VERSION_MICRO 0
#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \
LIBAVRESAMPLE_VERSION_MINOR, \
LIBAVRESAMPLE_VERSION_MICRO)
#define LIBAVRESAMPLE_VERSION AV_VERSION(LIBAVRESAMPLE_VERSION_MAJOR, \
LIBAVRESAMPLE_VERSION_MINOR, \
LIBAVRESAMPLE_VERSION_MICRO)
#define LIBAVRESAMPLE_BUILD LIBAVRESAMPLE_VERSION_INT
#define LIBAVRESAMPLE_IDENT "Lavr" AV_STRINGIFY(LIBAVRESAMPLE_VERSION)
/**
* These FF_API_* defines are not part of public API.
* They may change, break or disappear at any time.
*/
#endif /* AVRESAMPLE_VERSION_H */
libavresample/x86/Makefile 0 → 100644
View file @ c8af852b
OBJS += x86/audio_convert_init.o \
x86/audio_mix_init.o
YASM-OBJS += x86/audio_convert.o \
x86/audio_mix.o
libavresample/x86/audio_convert.asm 0 → 100644
View file @ c8af852b
;******************************************************************************
;* x86 optimized Format Conversion Utils
;* Copyright (c) 2008 Loren Merritt
;*
;* 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 "x86inc.asm"
%include "x86util.asm"
SECTION_TEXT
;-----------------------------------------------------------------------------
; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len,
; int channels);
;-----------------------------------------------------------------------------
%macro CONV_FLTP_TO_FLT_6CH 0
cglobal conv_fltp_to_flt_6ch, 2,8,7, dst, src, src1, src2, src3, src4, src5, len
%if ARCH_X86_64
mov lend, r2d
%else
%define lend dword r2m
%endif
mov src1q, [srcq+1*gprsize]
mov src2q, [srcq+2*gprsize]
mov src3q, [srcq+3*gprsize]
mov src4q, [srcq+4*gprsize]
mov src5q, [srcq+5*gprsize]
mov srcq, [srcq]
sub src1q, srcq
sub src2q, srcq
sub src3q, srcq
sub src4q, srcq
sub src5q, srcq
.loop:
mova m0, [srcq ]
mova m1, [srcq+src1q]
mova m2, [srcq+src2q]
mova m3, [srcq+src3q]
mova m4, [srcq+src4q]
mova m5, [srcq+src5q]
%if cpuflag(sse)
SBUTTERFLYPS 0, 1, 6
SBUTTERFLYPS 2, 3, 6
SBUTTERFLYPS 4, 5, 6
movaps m6, m4
shufps m4, m0, q3210
movlhps m0, m2
movhlps m6, m2
movaps [dstq ], m0
movaps [dstq+16], m4
movaps [dstq+32], m6
movaps m6, m5
shufps m5, m1, q3210
movlhps m1, m3
movhlps m6, m3
movaps [dstq+48], m1
movaps [dstq+64], m5
movaps [dstq+80], m6
%else ; mmx
SBUTTERFLY dq, 0, 1, 6
SBUTTERFLY dq, 2, 3, 6
SBUTTERFLY dq, 4, 5, 6
movq [dstq ], m0
movq [dstq+ 8], m2
movq [dstq+16], m4
movq [dstq+24], m1
movq [dstq+32], m3
movq [dstq+40], m5
%endif
add srcq, mmsize
add dstq, mmsize*6
sub lend, mmsize/4
jg .loop
%if mmsize == 8
emms
RET
%else
REP_RET
%endif
%endmacro
INIT_MMX mmx
CONV_FLTP_TO_FLT_6CH
INIT_XMM sse
CONV_FLTP_TO_FLT_6CH
libavresample/x86/audio_convert_init.c 0 → 100644
View file @ c8af852b
/*
* 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 "config.h"
#include "libavutil/cpu.h"
#include "libavresample/audio_convert.h"
extern void ff_conv_fltp_to_flt_6ch_mmx(float *dst, float *const *src, int len);
extern void ff_conv_fltp_to_flt_6ch_sse(float *dst, float *const *src, int len);
av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
{
#if HAVE_YASM
int mm_flags = av_get_cpu_flags();
if (mm_flags & AV_CPU_FLAG_MMX && HAVE_MMX) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx);
}
if (mm_flags & AV_CPU_FLAG_SSE && HAVE_SSE) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
6, 16, 4, "SSE", ff_conv_fltp_to_flt_6ch_sse);
}
#endif
}
libavresample/x86/audio_mix.asm 0 → 100644
View file @ c8af852b
;******************************************************************************
;* x86 optimized channel mixing
;* 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 "x86inc.asm"
%include "x86util.asm"
SECTION_TEXT
;-----------------------------------------------------------------------------
; void ff_mix_2_to_1_fltp_flt(float **src, float **matrix, int len,
; int out_ch, int in_ch);
;-----------------------------------------------------------------------------
%macro MIX_2_TO_1_FLTP_FLT 0
cglobal mix_2_to_1_fltp_flt, 3,4,6, src, matrix, len, src1
mov src1q, [srcq+gprsize]
mov srcq, [srcq ]
sub src1q, srcq
mov matrixq, [matrixq ]
VBROADCASTSS m4, [matrixq ]
VBROADCASTSS m5, [matrixq+4]
ALIGN 16
.loop:
mulps m0, m4, [srcq ]
mulps m1, m5, [srcq+src1q ]
mulps m2, m4, [srcq+ mmsize]
mulps m3, m5, [srcq+src1q+mmsize]
addps m0, m0, m1
addps m2, m2, m3
mova [srcq ], m0
mova [srcq+mmsize], m2
add srcq, mmsize*2
sub lend, mmsize*2/4
jg .loop
%if mmsize == 32
vzeroupper
RET
%else
REP_RET
%endif
%endmacro
INIT_XMM sse
MIX_2_TO_1_FLTP_FLT
INIT_YMM avx
MIX_2_TO_1_FLTP_FLT
libavresample/x86/audio_mix_init.c 0 → 100644
View file @ c8af852b
/*
* 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 "config.h"
#include "libavutil/cpu.h"
#include "libavresample/audio_mix.h"
extern void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len,
int out_ch, int in_ch);
extern void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len,
int out_ch, int in_ch);
av_cold void ff_audio_mix_init_x86(AudioMix *am)
{
#if HAVE_YASM
int mm_flags = av_get_cpu_flags();
if (mm_flags & AV_CPU_FLAG_SSE && HAVE_SSE) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse);
}
if (mm_flags & AV_CPU_FLAG_AVX && HAVE_AVX) {
ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,
2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx);
}
#endif
}
libavutil/x86/x86util.asm
View file @ c8af852b
......@@ -585,3 +585,12 @@
pminsd %1, %3
pmaxsd %1, %2
%endmacro
%macro VBROADCASTSS 2 ; dst xmm/ymm, src m32
%if cpuflag(avx)
vbroadcastss %1, %2
%else ; sse
movss %1, %2
shufps %1, %1, 0
%endif
%endmacro
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