FFmpeg/libavfilter/af_aconvert.c
Michael Niedermayer 8e576d5830 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  libavutil: add utility functions to simplify allocation of audio buffers.
  libavutil: add planar sample formats and av_sample_fmt_is_planar()
  avconv: fix segfault at EOF with delayed pictures
  pcmdec: remove unneeded resetting of samples pointer
  avconv: remove a now unused parameter from output_packet().
  avconv: formatting fixes in output_packet()
  avconv: declare some variables in blocks where they are used
  avconv: use the same behavior when decoding audio/video/subs
  bethsoftvideo: return proper consumed size for palette packets.
  cdg: skip packets that don't contain a cdg command.
  crcenc: add flags
  avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
  tiffenc: add a private option for selecting compression algorithm
  md5enc: add flags
  ARM: remove needless .text/.align directives

Conflicts:
	doc/APIchanges
	libavcodec/tiffenc.c
	libavutil/avutil.h
	libavutil/samplefmt.c
	libavutil/samplefmt.h
	tests/ref/fate/bethsoft-vid
	tests/ref/fate/cdgraphics
	tests/ref/fate/film-cvid-pcm-stereo-8bit
	tests/ref/fate/mpeg2-field-enc
	tests/ref/fate/nuv
	tests/ref/fate/tiertex-seq
	tests/ref/fate/tscc-32bit
	tests/ref/fate/vmnc-32bit

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-24 03:32:24 +01:00

419 lines
17 KiB
C

/*
* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@ucsd.edu>
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2011 Mina Nagy Zaki
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* sample format and channel layout conversion audio filter
* based on code in libavcodec/resample.c by Fabrice Bellard and
* libavcodec/audioconvert.c by Michael Niedermayer
*/
#include "libavutil/audioconvert.h"
#include "libavutil/avstring.h"
#include "libavcodec/audioconvert.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
enum AVSampleFormat out_sample_fmt, in_sample_fmt; ///< in/out sample formats
int64_t out_chlayout, in_chlayout; ///< in/out channel layout
int out_nb_channels, in_nb_channels; ///< number of in/output channels
enum AVFilterPacking out_packing_fmt, in_packing_fmt; ///< output packing format
int max_nb_samples; ///< maximum number of buffered samples
AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer
AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions
uint8_t *in_mix[8], *out_mix[8]; ///< input/output for rematrixing functions
uint8_t *packed_data[8]; ///< pointers for packing conversion
int out_strides[8], in_strides[8]; ///< input/output strides for av_audio_convert
uint8_t **in_conv, **out_conv; ///< input/output for av_audio_convert
AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format
void (*convert_chlayout)(); ///< function to do the requested rematrixing
} AConvertContext;
#define REMATRIX_FUNC_SIG(NAME) static void REMATRIX_FUNC_NAME(NAME) \
(FMT_TYPE *outp[], FMT_TYPE *inp[], int nb_samples, AConvertContext *aconvert)
#define FMT_TYPE uint8_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _u8
#include "af_aconvert_rematrix.c"
#define FMT_TYPE int16_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _s16
#include "af_aconvert_rematrix.c"
#define FMT_TYPE int32_t
#define REMATRIX_FUNC_NAME(NAME) NAME ## _s32
#include "af_aconvert_rematrix.c"
#define FLOATING
#define FMT_TYPE float
#define REMATRIX_FUNC_NAME(NAME) NAME ## _flt
#include "af_aconvert_rematrix.c"
#define FMT_TYPE double
#define REMATRIX_FUNC_NAME(NAME) NAME ## _dbl
#include "af_aconvert_rematrix.c"
#define FMT_TYPE uint8_t
#define REMATRIX_FUNC_NAME(NAME) NAME
REMATRIX_FUNC_SIG(stereo_remix_planar)
{
int size = av_get_bytes_per_sample(aconvert->in_sample_fmt) * nb_samples;
memcpy(outp[0], inp[0], size);
memcpy(outp[1], inp[aconvert->in_nb_channels == 1 ? 0 : 1], size);
}
#define REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC, PACKING) \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_U8, FUNC##_u8}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S16, FUNC##_s16}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S32, FUNC##_s32}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_FLT, FUNC##_flt}, \
{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_DBL, FUNC##_dbl},
#define REGISTER_FUNC(INCHLAYOUT, OUTCHLAYOUT, FUNC) \
REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_packed, AVFILTER_PACKED) \
REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_planar, AVFILTER_PLANAR)
static const struct RematrixFunctionInfo {
int64_t in_chlayout, out_chlayout;
int planar, sfmt;
void (*func)();
} rematrix_funcs[] = {
REGISTER_FUNC (AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, stereo_to_surround_5p1)
REGISTER_FUNC (AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_5p1_to_stereo)
REGISTER_FUNC_PACKING(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, stereo_to_mono_packed, AVFILTER_PACKED)
REGISTER_FUNC_PACKING(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, mono_to_stereo_packed, AVFILTER_PACKED)
REGISTER_FUNC (0, AV_CH_LAYOUT_MONO, mono_downmix)
REGISTER_FUNC_PACKING(0, AV_CH_LAYOUT_STEREO, stereo_downmix_packed, AVFILTER_PACKED)
// This function works for all sample formats
{0, AV_CH_LAYOUT_STEREO, AVFILTER_PLANAR, -1, stereo_remix_planar}
};
static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
{
AConvertContext *aconvert = ctx->priv;
char *arg, *ptr = NULL;
int ret = 0;
char *args = av_strdup(args0);
aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE;
aconvert->out_chlayout = 0;
aconvert->out_packing_fmt = -1;
if ((arg = av_strtok(args, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0)
goto end;
}
if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0)
goto end;
}
if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
if ((ret = ff_parse_packing_format((int *)&aconvert->out_packing_fmt, arg, ctx)) < 0)
goto end;
}
end:
av_freep(&args);
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AConvertContext *aconvert = ctx->priv;
avfilter_unref_buffer(aconvert->mix_samplesref);
avfilter_unref_buffer(aconvert->out_samplesref);
if (aconvert->audioconvert_ctx)
av_audio_convert_free(aconvert->audioconvert_ctx);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AConvertContext *aconvert = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&inlink->out_formats);
if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_sample_fmt);
avfilter_formats_ref(formats, &outlink->in_formats);
} else
avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
&outlink->in_formats);
avfilter_formats_ref(avfilter_make_all_channel_layouts(),
&inlink->out_chlayouts);
if (aconvert->out_chlayout != 0) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_chlayout);
avfilter_formats_ref(formats, &outlink->in_chlayouts);
} else
avfilter_formats_ref(avfilter_make_all_channel_layouts(),
&outlink->in_chlayouts);
avfilter_formats_ref(avfilter_make_all_packing_formats(),
&inlink->out_packing);
if (aconvert->out_packing_fmt != -1) {
formats = NULL;
avfilter_add_format(&formats, aconvert->out_packing_fmt);
avfilter_formats_ref(formats, &outlink->in_packing);
} else
avfilter_formats_ref(avfilter_make_all_packing_formats(),
&outlink->in_packing);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterLink *inlink = outlink->src->inputs[0];
AConvertContext *aconvert = outlink->src->priv;
char buf1[64], buf2[64];
aconvert->in_sample_fmt = inlink->format;
aconvert->in_packing_fmt = inlink->planar;
if (aconvert->out_packing_fmt == -1)
aconvert->out_packing_fmt = outlink->planar;
aconvert->in_chlayout = inlink->channel_layout;
aconvert->in_nb_channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
/* if not specified in args, use the format and layout of the output */
if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE)
aconvert->out_sample_fmt = outlink->format;
if (aconvert->out_chlayout == 0)
aconvert->out_chlayout = outlink->channel_layout;
aconvert->out_nb_channels =
av_get_channel_layout_nb_channels(outlink->channel_layout);
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(outlink->src, AV_LOG_INFO,
"fmt:%s cl:%s planar:%i -> fmt:%s cl:%s planar:%i\n",
av_get_sample_fmt_name(inlink ->format), buf1, inlink ->planar,
av_get_sample_fmt_name(outlink->format), buf2, outlink->planar);
/* compute which channel layout conversion to use */
if (inlink->channel_layout != outlink->channel_layout) {
int i;
for (i = 0; i < sizeof(rematrix_funcs); i++) {
const struct RematrixFunctionInfo *f = &rematrix_funcs[i];
if ((f->in_chlayout == 0 || f->in_chlayout == inlink ->channel_layout) &&
(f->out_chlayout == 0 || f->out_chlayout == outlink->channel_layout) &&
(f->planar == -1 || f->planar == inlink->planar) &&
(f->sfmt == -1 || f->sfmt == inlink->format)
) {
aconvert->convert_chlayout = f->func;
break;
}
}
if (!aconvert->convert_chlayout) {
av_log(outlink->src, AV_LOG_ERROR,
"Unsupported channel layout conversion '%s -> %s' requested!\n",
buf1, buf2);
return AVERROR(EINVAL);
}
}
return 0;
}
static int init_buffers(AVFilterLink *inlink, int nb_samples)
{
AConvertContext *aconvert = inlink->dst->priv;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int i, packed_stride = 0;
const unsigned
packing_conv = inlink->planar != outlink->planar &&
aconvert->out_nb_channels != 1,
format_conv = inlink->format != outlink->format;
int nb_channels = aconvert->out_nb_channels;
uninit(inlink->dst);
aconvert->max_nb_samples = nb_samples;
if (aconvert->convert_chlayout) {
/* allocate buffer for storing intermediary mixing samplesref */
uint8_t *data[8];
int linesize[8];
int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
if (av_samples_alloc(data, linesize, nb_channels, nb_samples,
inlink->format, 16) < 0)
goto fail_no_mem;
aconvert->mix_samplesref =
avfilter_get_audio_buffer_ref_from_arrays(data, linesize, AV_PERM_WRITE,
nb_samples, inlink->format,
outlink->channel_layout,
inlink->planar);
if (!aconvert->mix_samplesref)
goto fail_no_mem;
}
// if there's a format/packing conversion we need an audio_convert context
if (format_conv || packing_conv) {
aconvert->out_samplesref =
avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!aconvert->out_samplesref)
goto fail_no_mem;
aconvert->in_strides [0] = av_get_bytes_per_sample(inlink ->format);
aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format);
aconvert->out_conv = aconvert->out_samplesref->data;
if (aconvert->mix_samplesref)
aconvert->in_conv = aconvert->mix_samplesref->data;
if (packing_conv) {
// packed -> planar
if (outlink->planar == AVFILTER_PLANAR) {
if (aconvert->mix_samplesref)
aconvert->packed_data[0] = aconvert->mix_samplesref->data[0];
aconvert->in_conv = aconvert->packed_data;
packed_stride = aconvert->in_strides[0];
aconvert->in_strides[0] *= nb_channels;
// planar -> packed
} else {
aconvert->packed_data[0] = aconvert->out_samplesref->data[0];
aconvert->out_conv = aconvert->packed_data;
packed_stride = aconvert->out_strides[0];
aconvert->out_strides[0] *= nb_channels;
}
} else if (outlink->planar == AVFILTER_PACKED) {
/* If there's no packing conversion, and the stream is packed
* then we treat the entire stream as one big channel
*/
nb_channels = 1;
}
for (i = 1; i < nb_channels; i++) {
aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
aconvert->in_strides[i] = aconvert->in_strides[0];
aconvert->out_strides[i] = aconvert->out_strides[0];
}
aconvert->audioconvert_ctx =
av_audio_convert_alloc(outlink->format, nb_channels,
inlink->format, nb_channels, NULL, 0);
if (!aconvert->audioconvert_ctx)
goto fail_no_mem;
}
return 0;
fail_no_mem:
av_log(inlink->dst, AV_LOG_ERROR, "Could not allocate memory.\n");
return AVERROR(ENOMEM);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
AVFilterBufferRef *curbuf = insamplesref;
AVFilterLink * const outlink = inlink->dst->outputs[0];
int chan_mult;
/* in/reinint the internal buffers if this is the first buffer
* provided or it is needed to use a bigger one */
if (!aconvert->max_nb_samples ||
(curbuf->audio->nb_samples > aconvert->max_nb_samples))
if (init_buffers(inlink, curbuf->audio->nb_samples) < 0) {
av_log(inlink->dst, AV_LOG_ERROR, "Could not initialize buffers.\n");
return;
}
/* if channel mixing is required */
if (aconvert->mix_samplesref) {
memcpy(aconvert->in_mix, curbuf->data, sizeof(aconvert->in_mix));
memcpy(aconvert->out_mix, aconvert->mix_samplesref->data, sizeof(aconvert->out_mix));
aconvert->convert_chlayout(aconvert->out_mix,
aconvert->in_mix,
curbuf->audio->nb_samples,
aconvert);
curbuf = aconvert->mix_samplesref;
}
if (aconvert->audioconvert_ctx) {
if (!aconvert->mix_samplesref) {
if (aconvert->in_conv == aconvert->packed_data) {
int i, packed_stride = av_get_bytes_per_sample(inlink->format);
aconvert->packed_data[0] = curbuf->data[0];
for (i = 1; i < aconvert->out_nb_channels; i++)
aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
} else {
aconvert->in_conv = curbuf->data;
}
}
chan_mult = inlink->planar == outlink->planar && inlink->planar == 0 ?
aconvert->out_nb_channels : 1;
av_audio_convert(aconvert->audioconvert_ctx,
(void * const *) aconvert->out_conv,
aconvert->out_strides,
(const void * const *) aconvert->in_conv,
aconvert->in_strides,
curbuf->audio->nb_samples * chan_mult);
curbuf = aconvert->out_samplesref;
}
avfilter_copy_buffer_ref_props(curbuf, insamplesref);
curbuf->audio->channel_layout = outlink->channel_layout;
curbuf->audio->planar = outlink->planar;
avfilter_filter_samples(inlink->dst->outputs[0],
avfilter_ref_buffer(curbuf, ~0));
avfilter_unref_buffer(insamplesref);
}
AVFilter avfilter_af_aconvert = {
.name = "aconvert",
.description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout:packed_fmt."),
.priv_size = sizeof(AConvertContext),
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output, },
{ .name = NULL}},
};