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Add libswresample.

Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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1 parent 53e3784 commit b5875b91113a0f3de6ad61e9ff8b74b81de94760 @michaelni michaelni committed Sep 19, 2011
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@@ -33,6 +33,7 @@ FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
+FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil
View
@@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
-ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
View
@@ -89,6 +89,7 @@ Configuration options:
--disable-avdevice disable libavdevice build
--disable-avcodec disable libavcodec build
--disable-avformat disable libavformat build
+ --disable-swresample disable libswresample build
--disable-swscale disable libswscale build
--disable-postproc disable libpostproc build
--disable-avfilter disable video filter support [no]
@@ -1037,6 +1038,7 @@ CONFIG_LIST="
small
sram
static
+ swresample
swscale
swscale_alpha
thumb
@@ -1603,7 +1605,7 @@ avformat_deps="avcodec"
postproc_deps="gpl"
# programs
-ffmpeg_deps="avcodec avformat swscale"
+ffmpeg_deps="avcodec avformat swscale swresample"
ffmpeg_select="buffer_filter buffersink_filter"
avconv_deps="avcodec avformat swscale"
avconv_select="buffer_filter"
@@ -1766,6 +1768,7 @@ enable postproc
enable protocols
enable static
enable stripping
+enable swresample
enable swscale
enable swscale_alpha
@@ -3143,7 +3146,7 @@ enabled extra_warnings && check_cflags -Winline
# add some linker flags
check_ldflags -Wl,--warn-common
-check_ldflags -Wl,-rpath-link=libpostproc:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil
+check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil
test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic
echo "X{};" > $TMPV
View
@@ -45,6 +45,7 @@
#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "libavformat/os_support.h"
+#include "libswresample/swresample.h"
#include "libavformat/ffm.h" // not public API
@@ -229,15 +230,14 @@ typedef struct OutputStream {
/* audio only */
int audio_resample;
- ReSampleContext *resample; /* for audio resampling */
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
- int reformat_pair;
- AVAudioConvert *reformat_ctx;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
FILE *logfile;
+ struct SwrContext *swr;
+
#if CONFIG_AVFILTER
AVFilterContext *output_video_filter;
AVFilterContext *input_video_filter;
@@ -843,14 +843,15 @@ static void do_audio_out(AVFormatContext *s,
exit_program(1);
}
- if (enc->channels != dec->channels)
+ if (enc->channels != dec->channels
+ || enc->sample_fmt != dec->sample_fmt)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
- if ((ost->audio_resample && !ost->resample) || resample_changed) {
+ if ((ost->audio_resample && !ost->swr) || resample_changed) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d.%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
ist->file_index, ist->st->index,
@@ -859,24 +860,29 @@ static void do_audio_out(AVFormatContext *s,
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_sample_rate = dec->sample_rate;
- if (ost->resample)
- audio_resample_close(ost->resample);
+ swr_free(&ost->swr);
}
/* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
if (audio_sync_method <= 1 &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
- ost->resample = NULL;
+ //ost->swr = NULL;
ost->audio_resample = 0;
} else {
- if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
- fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
- ost->resample = av_audio_resample_init(enc->channels, dec->channels,
- enc->sample_rate, dec->sample_rate,
- enc->sample_fmt, dec->sample_fmt,
- 16, 10, 0, 0.8);
- if (!ost->resample) {
+ ost->swr = swr_alloc2(ost->swr,
+ enc->channel_layout, enc->sample_fmt, enc->sample_rate,
+ dec->channel_layout, dec->sample_fmt, dec->sample_rate,
+ 0, NULL);
+ av_set_int(ost->swr, "ich", dec->channels);
+ av_set_int(ost->swr, "och", enc->channels);
+ if(audio_sync_method>1) av_set_int(ost->swr, "flags", SWR_FLAG_RESAMPLE);
+ if(ost->swr && swr_init(ost->swr) < 0){
+ fprintf(stderr, "swr_init() failed\n");
+ swr_free(&ost->swr);
+ }
+
+ if (!ost->swr) {
fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
enc->channels, enc->sample_rate);
@@ -885,21 +891,7 @@ static void do_audio_out(AVFormatContext *s,
}
}
-#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
- if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
- MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
- if (ost->reformat_ctx)
- av_audio_convert_free(ost->reformat_ctx);
- ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1,
- dec->sample_fmt, 1, NULL, 0);
- if (!ost->reformat_ctx) {
- fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
- av_get_sample_fmt_name(dec->sample_fmt),
- av_get_sample_fmt_name(enc->sample_fmt));
- exit_program(1);
- }
- ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
- }
+ av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
@@ -941,7 +933,7 @@ static void do_audio_out(AVFormatContext *s,
if(verbose > 2)
fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
// fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2));
- av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
+ swr_compensate(ost->swr, comp, enc->sample_rate);
}
}
}else
@@ -950,30 +942,15 @@ static void do_audio_out(AVFormatContext *s,
if (ost->audio_resample) {
buftmp = audio_buf;
- size_out = audio_resample(ost->resample,
- (short *)buftmp, (short *)buf,
- size / (dec->channels * isize));
+ size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, audio_buf_size / (enc->channels * osize),
+ (const uint8_t*[]){buf }, size / (dec->channels * isize));
size_out = size_out * enc->channels * osize;
} else {
buftmp = buf;
size_out = size;
}
- if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
- const void *ibuf[6]= {buftmp};
- void *obuf[6]= {audio_buf};
- int istride[6]= {isize};
- int ostride[6]= {osize};
- int len= size_out/istride[0];
- if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
- printf("av_audio_convert() failed\n");
- if (exit_on_error)
- exit_program(1);
- return;
- }
- buftmp = audio_buf;
- size_out = len*osize;
- }
+ av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
/* now encode as many frames as possible */
if (enc->frame_size > 1) {
@@ -2133,7 +2110,6 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (!ost->fifo) {
return AVERROR(ENOMEM);
}
- ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
if (!codec->sample_rate) {
codec->sample_rate = icodec->sample_rate;
}
@@ -2149,6 +2125,8 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels)
codec->channel_layout = 0;
ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
+ ost->audio_resample |= codec->sample_fmt != icodec->sample_fmt
+ || codec->channel_layout != icodec->channel_layout;
icodec->request_channels = codec->channels;
ist->decoding_needed = 1;
ost->encoding_needed = 1;
@@ -2679,10 +2657,7 @@ static int transcode(OutputFile *output_files, int nb_output_files,
av_free(ost->forced_kf_pts);
if (ost->video_resample)
sws_freeContext(ost->img_resample_ctx);
- if (ost->resample)
- audio_resample_close(ost->resample);
- if (ost->reformat_ctx)
- av_audio_convert_free(ost->reformat_ctx);
+ swr_free(&ost->swr);
av_dict_free(&ost->opts);
}
}
View
@@ -0,0 +1,12 @@
+include $(SUBDIR)../config.mak
+
+NAME = swresample
+FFLIBS = avutil
+
+HEADERS = swresample.h
+
+OBJS = swresample.o audioconvert.o resample2.o rematrix.o
+
+TESTPROGS = swresample_test
+
+include $(SUBDIR)../subdir.mak
@@ -0,0 +1,113 @@
+/*
+ * audio conversion
+ * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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
+ * audio conversion
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/avassert.h"
+#include "libavutil/libm.h"
+#include "libavutil/samplefmt.h"
+#include "audioconvert.h"
+
+
+struct AVAudioConvert {
+ int channels;
+ int fmt_pair;
+};
+
+AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
+ enum AVSampleFormat in_fmt,
+ int channels, int flags)
+{
+ AVAudioConvert *ctx;
+ ctx = av_malloc(sizeof(AVAudioConvert));
+ if (!ctx)
+ return NULL;
+ ctx->channels = channels;
+ ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
+ return ctx;
+}
+
+void swr_audio_convert_free(AVAudioConvert **ctx)
+{
+ av_freep(ctx);
+}
+
+int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData*in, int len)
+{
+ int ch;
+
+ av_assert0(ctx->channels == out->ch_count);
+
+ //FIXME optimize common cases
+
+ for(ch=0; ch<ctx->channels; ch++){
+ const int is= (in ->planar ? 1 : in->ch_count) * in->bps;
+ const int os= (out->planar ? 1 :out->ch_count) *out->bps;
+ const uint8_t *pi= in ->ch[ch];
+ uint8_t *po= out->ch[ch];
+ uint8_t *end= po + os*len;
+ if(!po)
+ continue;
+
+#define CONV(ofmt, otype, ifmt, expr)\
+if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
+ do{\
+ *(otype*)po = expr; pi += is; po += os;\
+ }while(po < end);\
+}
+
+//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
+//FIXME rounding ?
+
+ CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
+ else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
+ else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
+ else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
+ else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
+ else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
+ else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
+ else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
+ else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
+ else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
+ else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
+ else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
+ else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
+ else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
+ else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
+ else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
+ else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
+ else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
+ else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
+ else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
+ else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
+ else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
+ else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
+ else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
+ else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
+ else return -1;
+ }
+ return 0;
+}
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