diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2011-10-22 01:03:27 +0200 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2011-10-22 01:16:41 +0200 |
commit | aedc908601de7396751a9a4504e064782d9f6a0b (patch) | |
tree | 8f04b899142439893bac426ac83d05c4068b099c /libavdevice | |
parent | 1a7090bfafe986d4470ba8059c815939171ddb74 (diff) | |
parent | f4b51d061f0f34e36be876b562b8abe47f4b9c1c (diff) | |
download | ffmpeg-aedc908601de7396751a9a4504e064782d9f6a0b.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (35 commits)
flvdec: Do not call parse_keyframes_index with a NULL stream
libspeexdec: include system headers before local headers
libspeexdec: return meaningful error codes
libspeexdec: cosmetics: reindent
libspeexdec: decode one frame at a time.
swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
Move timefilter code from lavf to lavd.
mov: add support for hdvd and pgapmetadata atoms
mov: rename function _stik, some indentation cosmetics
mov: rename function _int8 to remove ambiguity, some indentation cosmetics
mov: parse the gnre atom
mp3on4: check for allocation failures in decode_init_mp3on4()
mp3on4: create a separate flush function for MP3onMP4.
mp3on4: ensure that the frame channel count does not exceed the codec channel count.
mp3on4: set channel layout
mp3on4: fix the output channel order
mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
mp3on4: copy MPADSPContext from first context to all contexts.
fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
...
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_ps.c
libavcodec/h264dsp_template.c
libavcodec/h264idct_template.c
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavcodec/x86/h264dsp_mmx.c
libavdevice/Makefile
libavdevice/jack_audio.c
libavformat/Makefile
libavformat/flvdec.c
libavformat/flvenc.c
libavutil/pixfmt.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavdevice')
-rw-r--r-- | libavdevice/Makefile | 6 | ||||
-rw-r--r-- | libavdevice/alsa-audio.h | 2 | ||||
-rw-r--r-- | libavdevice/jack_audio.c | 3 | ||||
-rw-r--r-- | libavdevice/timefilter.c | 151 | ||||
-rw-r--r-- | libavdevice/timefilter.h | 97 |
5 files changed, 255 insertions, 4 deletions
diff --git a/libavdevice/Makefile b/libavdevice/Makefile index 879e933994..97dd380776 100644 --- a/libavdevice/Makefile +++ b/libavdevice/Makefile @@ -10,7 +10,7 @@ OBJS = alldevices.o avdevice.o # input/output devices OBJS-$(CONFIG_ALSA_INDEV) += alsa-audio-common.o \ - alsa-audio-dec.o + alsa-audio-dec.o timefilter.o OBJS-$(CONFIG_ALSA_OUTDEV) += alsa-audio-common.o \ alsa-audio-enc.o OBJS-$(CONFIG_BKTR_INDEV) += bktr.o @@ -19,7 +19,7 @@ OBJS-$(CONFIG_DSHOW_INDEV) += dshow.o dshow_enummediatypes.o \ dshow_pin.o dshow_common.o OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o -OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o +OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.o OBJS-$(CONFIG_LAVFI_INDEV) += lavfi.o OBJS-$(CONFIG_OPENAL_INDEV) += openal-dec.o OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o @@ -39,4 +39,6 @@ OBJS-$(CONFIG_LIBDC1394_INDEV) += libdc1394.o SKIPHEADERS-$(HAVE_ALSA_ASOUNDLIB_H) += alsa-audio.h SKIPHEADERS-$(HAVE_SNDIO_H) += sndio_common.h +TESTPROGS = timefilter + include $(SRC_PATH)/subdir.mak diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h index 83af464865..e453a2011b 100644 --- a/libavdevice/alsa-audio.h +++ b/libavdevice/alsa-audio.h @@ -33,7 +33,7 @@ #include <alsa/asoundlib.h> #include "config.h" #include "libavutil/log.h" -#include "libavformat/timefilter.h" +#include "timefilter.h" #include "avdevice.h" /* XXX: we make the assumption that the soundcard accepts this format */ diff --git a/libavdevice/jack_audio.c b/libavdevice/jack_audio.c index 72554f0ebe..42499d363b 100644 --- a/libavdevice/jack_audio.c +++ b/libavdevice/jack_audio.c @@ -28,7 +28,8 @@ #include "libavutil/fifo.h" #include "libavutil/opt.h" #include "libavcodec/avcodec.h" -#include "libavformat/timefilter.h" +#include "libavformat/avformat.h" +#include "timefilter.h" #include "avdevice.h" /** diff --git a/libavdevice/timefilter.c b/libavdevice/timefilter.c new file mode 100644 index 0000000000..3c67a59881 --- /dev/null +++ b/libavdevice/timefilter.c @@ -0,0 +1,151 @@ +/* + * Delay Locked Loop based time filter + * Copyright (c) 2009 Samalyse + * Copyright (c) 2009 Michael Niedermayer + * Author: Olivier Guilyardi <olivier samalyse com> + * 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 + */ + + +#include "config.h" +#include "timefilter.h" +#include "libavutil/mem.h" + +struct TimeFilter { + /// Delay Locked Loop data. These variables refer to mathematical + /// concepts described in: http://www.kokkinizita.net/papers/usingdll.pdf + double cycle_time; + double feedback2_factor; + double feedback3_factor; + double clock_period; + int count; +}; + +TimeFilter * ff_timefilter_new(double clock_period, double feedback2_factor, double feedback3_factor) +{ + TimeFilter *self = av_mallocz(sizeof(TimeFilter)); + self->clock_period = clock_period; + self->feedback2_factor = feedback2_factor; + self->feedback3_factor = feedback3_factor; + return self; +} + +void ff_timefilter_destroy(TimeFilter *self) +{ + av_freep(&self); +} + +void ff_timefilter_reset(TimeFilter *self) +{ + self->count = 0; +} + +double ff_timefilter_update(TimeFilter *self, double system_time, double period) +{ + self->count++; + if (self->count==1) { + /// init loop + self->cycle_time = system_time; + } else { + double loop_error; + self->cycle_time += self->clock_period * period; + /// calculate loop error + loop_error = system_time - self->cycle_time; + + /// update loop + self->cycle_time += FFMAX(self->feedback2_factor, 1.0/(self->count)) * loop_error; + self->clock_period += self->feedback3_factor * loop_error / period; + } + return self->cycle_time; +} + +#ifdef TEST +#include "libavutil/lfg.h" +#define LFG_MAX ((1LL << 32) - 1) + +#undef printf + +int main(void) +{ + AVLFG prng; + double n0,n1; +#define SAMPLES 1000 + double ideal[SAMPLES]; + double samples[SAMPLES]; +#if 1 + for(n0= 0; n0<40; n0=2*n0+1){ + for(n1= 0; n1<10; n1=2*n1+1){ +#else + {{ + n0=7; + n1=1; +#endif + double best_error= 1000000000; + double bestpar0=1; + double bestpar1=0.001; + int better, i; + + av_lfg_init(&prng, 123); + for(i=0; i<SAMPLES; i++){ + ideal[i] = 10 + i + n1*i/(1000); + samples[i] = ideal[i] + n0 * (av_lfg_get(&prng) - LFG_MAX / 2) + / (LFG_MAX * 10LL); + } + + do{ + double par0, par1; + better=0; + for(par0= bestpar0*0.8; par0<=bestpar0*1.21; par0+=bestpar0*0.05){ + for(par1= bestpar1*0.8; par1<=bestpar1*1.21; par1+=bestpar1*0.05){ + double error=0; + TimeFilter *tf= ff_timefilter_new(1, par0, par1); + for(i=0; i<SAMPLES; i++){ + double filtered; + filtered= ff_timefilter_update(tf, samples[i], 1); + error += (filtered - ideal[i]) * (filtered - ideal[i]); + } + ff_timefilter_destroy(tf); + if(error < best_error){ + best_error= error; + bestpar0= par0; + bestpar1= par1; + better=1; + } + } + } + }while(better); +#if 0 + double lastfil=9; + TimeFilter *tf= ff_timefilter_new(1, bestpar0, bestpar1); + for(i=0; i<SAMPLES; i++){ + double filtered; + filtered= ff_timefilter_update(tf, samples[i], 1); + printf("%f %f %f %f\n", i - samples[i] + 10, filtered - samples[i], samples[FFMAX(i, 1)] - samples[FFMAX(i-1, 0)], filtered - lastfil); + lastfil= filtered; + } + ff_timefilter_destroy(tf); +#else + printf(" [%f %f %9f]", bestpar0, bestpar1, best_error); +#endif + } + printf("\n"); + } + return 0; +} +#endif diff --git a/libavdevice/timefilter.h b/libavdevice/timefilter.h new file mode 100644 index 0000000000..4580904092 --- /dev/null +++ b/libavdevice/timefilter.h @@ -0,0 +1,97 @@ +/* + * Delay Locked Loop based time filter prototypes and declarations + * Copyright (c) 2009 Samalyse + * Copyright (c) 2009 Michael Niedermayer + * Author: Olivier Guilyardi <olivier samalyse com> + * 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 + */ + +#ifndef AVDEVICE_TIMEFILTER_H +#define AVDEVICE_TIMEFILTER_H + +/** + * Opaque type representing a time filter state + * + * The purpose of this filter is to provide a way to compute accurate time + * stamps that can be compared to wall clock time, especially when dealing + * with two clocks: the system clock and a hardware device clock, such as + * a soundcard. + */ +typedef struct TimeFilter TimeFilter; + + +/** + * Create a new Delay Locked Loop time filter + * + * feedback2_factor and feedback3_factor are the factors used for the + * multiplications that are respectively performed in the second and third + * feedback paths of the loop. + * + * Unless you know what you are doing, you should set these as follow: + * + * o = 2 * M_PI * bandwidth * period + * feedback2_factor = sqrt(2 * o) + * feedback3_factor = o * o + * + * Where bandwidth is up to you to choose. Smaller values will filter out more + * of the jitter, but also take a longer time for the loop to settle. A good + * starting point is something between 0.3 and 3 Hz. + * + * @param clock_period period of the hardware clock in seconds + * (for example 1.0/44100) + * + * For more details about these parameters and background concepts please see: + * http://www.kokkinizita.net/papers/usingdll.pdf + */ +TimeFilter * ff_timefilter_new(double clock_period, double feedback2_factor, double feedback3_factor); + +/** + * Update the filter + * + * This function must be called in real time, at each process cycle. + * + * @param period the device cycle duration in clock_periods. For example, at + * 44.1kHz and a buffer size of 512 frames, period = 512 when clock_period + * was 1.0/44100, or 512/44100 if clock_period was 1. + * + * system_time, in seconds, should be the value of the system clock time, + * at (or as close as possible to) the moment the device hardware interrupt + * occured (or any other event the device clock raises at the beginning of a + * cycle). + * + * @return the filtered time, in seconds + */ +double ff_timefilter_update(TimeFilter *self, double system_time, double period); + +/** + * Reset the filter + * + * This function should mainly be called in case of XRUN. + * + * Warning: after calling this, the filter is in an undetermined state until + * the next call to ff_timefilter_update() + */ +void ff_timefilter_reset(TimeFilter *); + +/** + * Free all resources associated with the filter + */ +void ff_timefilter_destroy(TimeFilter *); + +#endif /* AVDEVICE_TIMEFILTER_H */ |