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ffmpeg_ntsc.cpp
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ffmpeg_ntsc.cpp
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// NTS: This is not like modern "posterize" filters where the pixels are quantizied to N levels then scaled out to 0..255
// That requires a multiply/divide per pixel. Think old-school hardware where such operations were too expensive.
// The "posterize" we emulate here is more the type where you run the video through an ADC, truncate the least significant
// bits, then run back through a DAC on the other side (well within the realm of 1980s/1990s hardware)
#define __STDC_CONSTANT_MACROS
#define __STDC_LIMIT_MACROS
#include <sys/types.h>
#include <signal.h>
#include <stdint.h>
#include <assert.h>
#include <unistd.h>
#include <stdlib.h>
#include <stdio.h>
#include <fcntl.h>
#include <math.h>
extern "C" {
#include <libavutil/opt.h>
#include <libavutil/avutil.h>
#include <libavutil/pixfmt.h>
#include <libavutil/pixdesc.h>
#include <libavutil/samplefmt.h>
#include <libavutil/pixelutils.h>
#include <libavcodec/avcodec.h>
#include <libavcodec/version.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavformat/version.h>
#include <libswscale/swscale.h>
#include <libswscale/version.h>
#include <libswresample/swresample.h>
#include <libswresample/version.h>
}
using namespace std;
#include <map>
#include <string>
#include <vector>
#include <stdexcept>
/* return a floating point value specifying what to scale the sample
* value by to reduce it from full volume to dB decibels */
double dBFS(double dB)
{
/* 10 ^ (dB / 20),
based on reversing the formula for converting samples to decibels:
dB = 20.0 * log10(sample);
where "sample" is -1.0 <= x <= 1.0 */
return pow(10.0,dB / 20.0);
}
/* attenuate a sample value by this many dBFS */
/* so if you want to reduce it by 20dBFS you pass -20 as dB */
double attenuate_dBFS(double sample,double dB)
{
return sample * dBFS(dB);
}
/* opposite: convert sample to decibels */
double dBFS_measure(double sample) {
return 20.0 * log10(sample);
}
// lowpass filter
// you can make it a highpass filter by applying a lowpass then subtracting from source.
class LowpassFilter {
public:
LowpassFilter() : timeInterval(0), cutoff(0), alpha(0), prev(0), tau(0) {
}
void setFilter(const double rate/*sample rate of audio*/,const double hz/*cutoff*/) {
#ifndef M_PI
#error your math.h does not include M_PI constant
#endif
timeInterval = 1.0 / rate;
tau = 1 / (hz * 2 * M_PI);
cutoff = hz;
alpha = timeInterval / (tau + timeInterval);
}
void resetFilter(const double val=0) {
prev = val;
}
double lowpass(const double sample) {
const double stage1 = sample * alpha;
const double stage2 = prev - (prev * alpha); /* NTS: Instead of prev * (1.0 - alpha) */
return (prev = (stage1 + stage2)); /* prev = stage1+stage2 then return prev */
}
double highpass(const double sample) {
const double stage1 = sample * alpha;
const double stage2 = prev - (prev * alpha); /* NTS: Instead of prev * (1.0 - alpha) */
return sample - (prev = (stage1 + stage2)); /* prev = stage1+stage2 then return (sample - prev) */
}
public:
double timeInterval;
double cutoff;
double alpha; /* timeInterval / (tau + timeInterval) */
double prev;
double tau;
};
class HiLoPair {
public:
LowpassFilter hi,lo; // highpass, lowpass
public:
void setFilter(const double rate/*sample rate of audio*/,const double low_hz,const double high_hz) {
lo.setFilter(rate,low_hz);
hi.setFilter(rate,high_hz);
}
double filter(const double sample) {
return hi.highpass(lo.lowpass(sample)); /* first lowpass, then highpass */
}
};
class HiLoPass : public vector<HiLoPair> { // all passes, one sample of one channel
public:
HiLoPass() : vector() { }
public:
void setFilter(const double rate/*sample rate of audio*/,const double low_hz,const double high_hz) {
for (size_t i=0;i < size();i++) (*this)[i].setFilter(rate,low_hz,high_hz);
}
double filter(double sample) {
for (size_t i=0;i < size();i++) sample = (*this)[i].lo.lowpass(sample);
for (size_t i=0;i < size();i++) sample = (*this)[i].hi.highpass(sample);
return sample;
}
void init(const unsigned int passes) {
clear();
resize(passes);
assert(size() >= passes);
}
};
class HiLoSample : public vector<HiLoPass> { // all passes, all channels of one sample period
public:
HiLoSample() : vector() { }
public:
void init(const unsigned int channels,const unsigned int passes) {
clear();
resize(channels);
assert(size() >= channels);
for (size_t i=0;i < size();i++) (*this)[i].init(passes);
}
void setFilter(const double rate/*sample rate of audio*/,const double low_hz,const double high_hz) {
for (size_t i=0;i < size();i++) (*this)[i].setFilter(rate,low_hz,high_hz);
}
};
class HiLoComboPass {
public:
HiLoComboPass() : passes(0), channels(0), rate(0), low_cutoff(0), high_cutoff(0) {
}
~HiLoComboPass() {
clear();
}
void setChannels(const size_t _channels) {
if (channels != _channels) {
clear();
channels = _channels;
}
}
void setCutoff(const double _low_cutoff,const double _high_cutoff) {
if (low_cutoff != _low_cutoff || high_cutoff != _high_cutoff) {
clear();
low_cutoff = _low_cutoff;
high_cutoff = _high_cutoff;
}
}
void setRate(const double _rate) {
if (rate != _rate) {
clear();
rate = _rate;
}
}
void setPasses(const size_t _passes) {
if (passes != _passes) {
clear();
passes = _passes;
}
}
void clear() {
audiostate.clear();
}
void init() {
clear();
if (channels == 0 || passes == 0 || rate == 0 || low_cutoff == 0 || high_cutoff == 0) return;
audiostate.init(channels,passes);
audiostate.setFilter(rate,low_cutoff,high_cutoff);
}
public:
double rate;
size_t passes;
size_t channels;
double low_cutoff;
double high_cutoff;
HiLoSample audiostate;
};
bool use_422_colorspace = false; // I would default this to true but Adobe Premiere Pro apparently can't handle 4:2:2 H.264 >:(
AVRational output_field_rate = { 60000, 1001 }; // NTSC 60Hz default
int output_width = 720;
int output_height = 480;
bool output_ntsc = true; // NTSC color subcarrier emulation
bool output_pal = false; // PAL color subcarrier emulation
int output_audio_channels = 2; // VHS stereo (set to 1 for mono)
int output_audio_rate = 44100; // VHS Hi-Fi goes up to 20KHz
int video_scanline_phase_shift = 180;
int video_scanline_phase_shift_offset = 0;
#define RGBTRIPLET(r,g,b) (((uint32_t)(r) << (uint32_t)16) + ((uint32_t)(g) << (uint32_t)8) + ((uint32_t)(b) << (uint32_t)0))
AVFormatContext* output_avfmt = NULL;
AVStream* output_avstream_audio = NULL; // do not free
AVCodecContext* output_avstream_audio_codec_context = NULL; // do not free
AVStream* output_avstream_video = NULL; // do not free
AVCodecContext* output_avstream_video_codec_context = NULL; // do not free
std::vector<AVFrame*> output_avstream_video_frame; // ARGB
AVFrame* output_avstream_video_encode_frame = NULL; // 4:2:2 or 4:2:0
size_t output_avstream_video_frame_delay = 1;
size_t output_avstream_video_frame_index = 0;
struct SwsContext* output_avstream_video_resampler = NULL;
class InputFile {
public:
InputFile() {
input_avfmt = NULL;
audio_dst_data = NULL;
input_avstream_audio = NULL;
input_avstream_audio_frame = NULL;
input_avstream_audio_resampler = NULL;
input_avstream_audio_codec_context = NULL;
input_avstream_video = NULL;
input_avstream_video_frame = NULL;
input_avstream_video_frame_rgb = NULL;
input_avstream_video_resampler = NULL;
input_avstream_video_codec_context = NULL;
next_pts = next_dts = -1LL;
avpkt_valid = false;
eof_stream = false;
eof = false;
}
~InputFile() {
close_input();
}
public:
void reset_on_dup(void) {
path.clear();
}
bool open_input(void) {
if (input_avfmt == NULL) {
if (avformat_open_input(&input_avfmt,path.c_str(),NULL,NULL) < 0) {
fprintf(stderr,"Failed to open input file\n");
close_input();
return false;
}
if (avformat_find_stream_info(input_avfmt,NULL) < 0)
fprintf(stderr,"WARNING: Did not find stream info on input\n");
/* scan streams for one video, one audio */
{
size_t i;
AVStream *is;
int ac=0,vc=0;
AVCodecContext *isctx;
fprintf(stderr,"Input format: %u streams found\n",input_avfmt->nb_streams);
for (i=0;i < (size_t)input_avfmt->nb_streams;i++) {
is = input_avfmt->streams[i];
if (is == NULL) continue;
isctx = is->codec;
if (isctx == NULL) continue;
if (isctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (input_avstream_audio == NULL && ac == 0) {
if (avcodec_open2(isctx,avcodec_find_decoder(isctx->codec_id),NULL) >= 0) {
input_avstream_audio = is;
input_avstream_audio_codec_context = isctx;
fprintf(stderr,"Found audio stream idx=%zu %u-channel %uHz\n",
i,
input_avstream_audio_codec_context->channels,
input_avstream_audio_codec_context->sample_rate);
}
else {
fprintf(stderr,"Found audio stream but not able to decode\n");
}
}
ac++;
}
else if (isctx->codec_type == AVMEDIA_TYPE_VIDEO) {
if (input_avstream_video == NULL && vc == 0) {
if (avcodec_open2(isctx,avcodec_find_decoder(isctx->codec_id),NULL) >= 0) {
input_avstream_video = is;
input_avstream_video_codec_context = isctx;
fprintf(stderr,"Found video stream idx=%zu\n",i);
}
else {
fprintf(stderr,"Found video stream but not able to decode\n");
}
}
vc++;
}
}
if (input_avstream_video == NULL && input_avstream_audio == NULL) {
fprintf(stderr,"Neither video nor audio found\n");
close_input();
return 1;
}
}
}
/* prepare audio decoding */
if (input_avstream_audio != NULL) {
input_avstream_audio_frame = av_frame_alloc();
if (input_avstream_audio_frame == NULL) {
fprintf(stderr,"Failed to alloc audio frame\n");
close_input();
return 1;
}
}
/* prepare video decoding */
if (input_avstream_video != NULL) {
input_avstream_video_frame = av_frame_alloc();
if (input_avstream_video_frame == NULL) {
fprintf(stderr,"Failed to alloc video frame\n");
close_input();
return 1;
}
/* prepare video encoding */
input_avstream_video_frame_rgb = av_frame_alloc();
if (input_avstream_video_frame_rgb == NULL) {
fprintf(stderr,"Failed to alloc video frame\n");
close_input();
return 1;
}
input_avstream_video_frame_rgb->format = AV_PIX_FMT_BGRA;
input_avstream_video_frame_rgb->height = output_height;
input_avstream_video_frame_rgb->width = output_width;
if (av_frame_get_buffer(input_avstream_video_frame_rgb,64) < 0) {
fprintf(stderr,"Failed to alloc render frame\n");
close_input();
return 1;
}
memset(input_avstream_video_frame_rgb->data[0],0,input_avstream_video_frame_rgb->linesize[0]*input_avstream_video_frame_rgb->height);
}
input_avstream_video_resampler_format = AV_PIX_FMT_NONE;
input_avstream_video_resampler_height = -1;
input_avstream_video_resampler_width = -1;
last_written_sample = 0;
audio_dst_data_out_audio_sample = 0;
audio_sample = 0;
audio_dst_data = NULL;
audio_dst_data_alloc_samples = 0;
audio_dst_data_linesize = 0;
audio_dst_data_samples = 0;
audio_dst_data_out_samples = 0;
input_avstream_audio_resampler_channels = -1;
input_avstream_audio_resampler_rate = -1;
eof_stream = false;
got_audio = false;
got_video = false;
adj_time = 0;
t = pt = -1;
eof = false;
avpkt_init();
next_pts = next_dts = -1LL;
return (input_avfmt != NULL);
}
bool next_packet(void) {
if (eof) return false;
if (input_avfmt == NULL) return false;
do {
if (eof_stream) break;
avpkt_release();
avpkt_init();
if (av_read_frame(input_avfmt,&avpkt) < 0) {
eof_stream = true;
return false;
}
if (avpkt.stream_index >= input_avfmt->nb_streams)
continue;
// ugh... this can happen if the source is an AVI file
if (avpkt.pts == AV_NOPTS_VALUE) avpkt.pts = avpkt.dts;
/* track time and keep things monotonic for our code */
if (avpkt.pts != AV_NOPTS_VALUE) {
t = avpkt.pts * av_q2d(input_avfmt->streams[avpkt.stream_index]->time_base);
if (pt < 0)
adj_time = -t;
else if ((t+1.5) < pt) { // time code jumps backwards (1.5 is safe for DVD timecode resets)
adj_time += pt - t;
fprintf(stderr,"Time code jump backwards %.6f->%.6f. adj_time=%.6f\n",pt,t,adj_time);
}
else if (t > (pt+5)) { // time code jumps forwards
adj_time += pt - t;
fprintf(stderr,"Time code jump forwards %.6f->%.6f. adj_time=%.6f\n",pt,t,adj_time);
}
pt = t;
}
if (pt < 0)
continue;
if (avpkt.pts != AV_NOPTS_VALUE) {
avpkt.pts += (adj_time * input_avfmt->streams[avpkt.stream_index]->time_base.den) /
input_avfmt->streams[avpkt.stream_index]->time_base.num;
}
if (avpkt.dts != AV_NOPTS_VALUE) {
avpkt.dts += (adj_time * input_avfmt->streams[avpkt.stream_index]->time_base.den) /
input_avfmt->streams[avpkt.stream_index]->time_base.num;
}
got_audio = false;
got_video = false;
if (input_avstream_audio != NULL && avpkt.stream_index == input_avstream_audio->index) {
if (got_audio) fprintf(stderr,"Audio content lost\n");
av_packet_rescale_ts(&avpkt,input_avstream_audio->time_base,output_avstream_audio->time_base);
handle_audio(/*&*/avpkt);
got_audio = true;
break;
}
else if (input_avstream_video != NULL && avpkt.stream_index == input_avstream_video->index) {
if (got_video) fprintf(stderr,"Video content lost\n");
AVRational m = (AVRational){output_field_rate.den, output_field_rate.num};
av_packet_rescale_ts(&avpkt,input_avstream_video->time_base,m); // convert to FIELD number
handle_frame(/*&*/avpkt); // will set got_video
break;
}
avpkt_release();
} while (1);
if (eof_stream) {
avpkt_release();
avpkt.size = 0;
avpkt.data = NULL;
handle_frame(/*&*/avpkt); // will set got_video
if (!got_video) eof = true;
else fprintf(stderr,"Got latent frame\n");
}
return true;
}
void handle_audio(AVPacket &pkt) {
int got_frame = 0;
if (avcodec_decode_audio4(input_avstream_audio_codec_context,input_avstream_audio_frame,&got_frame,&pkt) >= 0) {
if (got_frame != 0 && input_avstream_audio_frame->nb_samples != 0) {
if (input_avstream_audio_frame->pts == AV_NOPTS_VALUE)
input_avstream_audio_frame->pts = pkt.pts;
if (input_avstream_audio_resampler != NULL) {
if (input_avstream_audio_resampler_rate != input_avstream_audio_codec_context->sample_rate ||
input_avstream_audio_resampler_channels != input_avstream_audio_codec_context->channels) {
fprintf(stderr,"Audio format changed\n");
swr_free(&input_avstream_audio_resampler);
}
}
if (input_avstream_audio_resampler == NULL) {
input_avstream_audio_resampler = swr_alloc();
av_opt_set_int(input_avstream_audio_resampler, "in_channel_count", input_avstream_audio_codec_context->channels, 0); // FIXME: FFMPEG should document this!!
av_opt_set_int(input_avstream_audio_resampler, "out_channel_count", output_avstream_audio_codec_context->channels, 0); // FIXME: FFMPEG should document this!!
av_opt_set_int(input_avstream_audio_resampler, "in_channel_layout", input_avstream_audio_codec_context->channel_layout, 0);
av_opt_set_int(input_avstream_audio_resampler, "out_channel_layout", output_avstream_audio_codec_context->channel_layout, 0);
av_opt_set_int(input_avstream_audio_resampler, "in_sample_rate", input_avstream_audio_codec_context->sample_rate, 0);
av_opt_set_int(input_avstream_audio_resampler, "out_sample_rate", output_avstream_audio_codec_context->sample_rate, 0);
av_opt_set_sample_fmt(input_avstream_audio_resampler, "in_sample_fmt", input_avstream_audio_codec_context->sample_fmt, 0);
av_opt_set_sample_fmt(input_avstream_audio_resampler, "out_sample_fmt", output_avstream_audio_codec_context->sample_fmt, 0);
if (swr_init(input_avstream_audio_resampler) < 0) {
fprintf(stderr,"Failed to init audio resampler\n");
swr_free(&input_avstream_audio_resampler);
return;
}
input_avstream_audio_resampler_rate = input_avstream_audio_codec_context->sample_rate;
input_avstream_audio_resampler_channels = input_avstream_audio_codec_context->channels;
if (audio_dst_data != NULL) {
av_freep(&audio_dst_data[0]); // NTS: Why??
av_freep(&audio_dst_data);
}
audio_dst_data_alloc_samples = 0;
fprintf(stderr,"Audio resampler init %uHz -> %uHz\n",
input_avstream_audio_codec_context->sample_rate,
output_avstream_audio_codec_context->sample_rate);
}
audio_dst_data_samples = av_rescale_rnd(
swr_get_delay(input_avstream_audio_resampler, input_avstream_audio_frame->sample_rate) + input_avstream_audio_frame->nb_samples,
output_avstream_audio_codec_context->sample_rate, input_avstream_audio_frame->sample_rate, AV_ROUND_UP);
if (audio_dst_data == NULL || audio_dst_data_samples > audio_dst_data_alloc_samples) {
if (audio_dst_data != NULL) {
av_freep(&audio_dst_data[0]); // NTS: Why??
av_freep(&audio_dst_data);
}
audio_dst_data_alloc_samples = 0;
fprintf(stderr,"Allocating audio buffer %u samples\n",(unsigned int)audio_dst_data_samples);
if (av_samples_alloc_array_and_samples(&audio_dst_data,&audio_dst_data_linesize,
output_avstream_audio_codec_context->channels,audio_dst_data_samples,
output_avstream_audio_codec_context->sample_fmt, 0) >= 0) {
audio_dst_data_alloc_samples = audio_dst_data_samples;
}
else {
fprintf(stderr,"Failure to allocate audio buffer\n");
audio_dst_data_alloc_samples = 0;
}
}
if (audio_dst_data != NULL) {
if ((audio_dst_data_out_samples=swr_convert(input_avstream_audio_resampler,audio_dst_data,audio_dst_data_samples,
(const uint8_t**)input_avstream_audio_frame->data,input_avstream_audio_frame->nb_samples)) > 0) {
}
else if (audio_dst_data_out_samples < 0) {
fprintf(stderr,"Failed to resample audio\n");
}
audio_dst_data_out_audio_sample = audio_sample;
audio_sample += audio_dst_data_out_samples;
}
}
}
}
void frame_copy_scale(void) {
if (input_avstream_video_frame_rgb == NULL) {
fprintf(stderr,"New input frame\n");
input_avstream_video_frame_rgb = av_frame_alloc();
if (input_avstream_video_frame_rgb == NULL) {
fprintf(stderr,"Failed to alloc video frame\n");
return;
}
input_avstream_video_frame_rgb->format = AV_PIX_FMT_BGRA;
input_avstream_video_frame_rgb->height = output_height;
input_avstream_video_frame_rgb->width = output_width;
if (av_frame_get_buffer(input_avstream_video_frame_rgb,64) < 0) {
fprintf(stderr,"Failed to alloc render frame\n");
return;
}
memset(input_avstream_video_frame_rgb->data[0],0,input_avstream_video_frame_rgb->linesize[0]*input_avstream_video_frame_rgb->height);
}
if (input_avstream_video_resampler != NULL) { // pixel format change or width/height change = free resampler and reinit
if (input_avstream_video_resampler_format != input_avstream_video_frame->format ||
input_avstream_video_resampler_width != input_avstream_video_frame->width ||
input_avstream_video_resampler_height != input_avstream_video_frame->height) {
sws_freeContext(input_avstream_video_resampler);
input_avstream_video_resampler = NULL;
}
}
if (input_avstream_video_resampler == NULL) {
input_avstream_video_resampler = sws_getContext(
// source
input_avstream_video_frame->width,
input_avstream_video_frame->height,
(AVPixelFormat)input_avstream_video_frame->format,
// dest
input_avstream_video_frame_rgb->width,
input_avstream_video_frame_rgb->height,
(AVPixelFormat)input_avstream_video_frame_rgb->format,
// opt
SWS_BILINEAR, NULL, NULL, NULL);
if (input_avstream_video_resampler != NULL) {
fprintf(stderr,"sws_getContext new context\n");
input_avstream_video_resampler_format = (AVPixelFormat)input_avstream_video_frame->format;
input_avstream_video_resampler_width = input_avstream_video_frame->width;
input_avstream_video_resampler_height = input_avstream_video_frame->height;
}
else {
fprintf(stderr,"sws_getContext fail\n");
}
}
if (input_avstream_video_resampler != NULL) {
input_avstream_video_frame_rgb->pts = input_avstream_video_frame->pts;
input_avstream_video_frame_rgb->pkt_pts = input_avstream_video_frame->pkt_pts;
input_avstream_video_frame_rgb->pkt_dts = input_avstream_video_frame->pkt_dts;
input_avstream_video_frame_rgb->top_field_first = input_avstream_video_frame->top_field_first;
input_avstream_video_frame_rgb->interlaced_frame = input_avstream_video_frame->interlaced_frame;
if (sws_scale(input_avstream_video_resampler,
// source
input_avstream_video_frame->data,
input_avstream_video_frame->linesize,
0,input_avstream_video_frame->height,
// dest
input_avstream_video_frame_rgb->data,
input_avstream_video_frame_rgb->linesize) <= 0)
fprintf(stderr,"WARNING: sws_scale failed\n");
}
}
void handle_frame(AVPacket &pkt) {
int got_frame = 0;
if (avcodec_decode_video2(input_avstream_video_codec_context,input_avstream_video_frame,&got_frame,&pkt) >= 0) {
if (got_frame != 0 && input_avstream_video_frame->width > 0 && input_avstream_video_frame->height > 0) {
got_video = true;
}
}
else {
fprintf(stderr,"No video decoded\n");
}
}
void avpkt_init(void) {
if (!avpkt_valid) {
avpkt_valid = true;
av_init_packet(&avpkt);
}
}
void avpkt_release(void) {
if (avpkt_valid) {
avpkt_valid = false;
av_packet_unref(&avpkt);
}
got_audio = false;
got_video = false;
}
void close_input(void) {
eof = true;
avpkt_release();
if (input_avstream_audio_codec_context != NULL) {
avcodec_close(input_avstream_audio_codec_context);
input_avstream_audio_codec_context = NULL;
input_avstream_audio = NULL;
}
if (input_avstream_video_codec_context != NULL) {
avcodec_close(input_avstream_video_codec_context);
input_avstream_video_codec_context = NULL;
input_avstream_video = NULL;
}
if (input_avstream_audio_frame != NULL)
av_frame_free(&input_avstream_audio_frame);
if (input_avstream_video_frame != NULL)
av_frame_free(&input_avstream_video_frame);
if (input_avstream_video_frame_rgb != NULL)
av_frame_free(&input_avstream_video_frame_rgb);
if (input_avstream_audio_resampler != NULL)
swr_free(&input_avstream_audio_resampler);
if (input_avstream_video_resampler != NULL) {
sws_freeContext(input_avstream_video_resampler);
input_avstream_video_resampler = NULL;
}
if (audio_dst_data != NULL) {
av_freep(&audio_dst_data[0]); // NTS: Why??
av_freep(&audio_dst_data);
}
input_avstream_audio_resampler_channels = -1;
input_avstream_audio_resampler_rate = -1;
avformat_close_input(&input_avfmt);
}
public:
std::string path;
uint32_t color;
bool eof;
bool eof_stream;
bool got_audio;
bool got_video;
public:
unsigned long long last_written_sample;
unsigned long long audio_sample;
uint8_t** audio_dst_data;
int audio_dst_data_alloc_samples;
int audio_dst_data_linesize;
int audio_dst_data_samples;
int audio_dst_data_out_samples;
unsigned long long audio_dst_data_out_audio_sample;
int input_avstream_audio_resampler_rate;
int input_avstream_audio_resampler_channels;
AVFormatContext* input_avfmt;
AVStream* input_avstream_audio; // do not free
AVCodecContext* input_avstream_audio_codec_context; // do not free
AVFrame* input_avstream_audio_frame;
AVStream* input_avstream_video; // do not free
AVCodecContext* input_avstream_video_codec_context; // do not free
AVFrame* input_avstream_video_frame;
AVFrame* input_avstream_video_frame_rgb;
struct SwrContext* input_avstream_audio_resampler;
struct SwsContext* input_avstream_video_resampler;
AVPixelFormat input_avstream_video_resampler_format;
int input_avstream_video_resampler_height;
int input_avstream_video_resampler_width;
signed long long next_pts;
signed long long next_dts;
AVPacket avpkt;
bool avpkt_valid;
double adj_time;
double t,pt;
};
std::vector<InputFile> input_files;
std::string output_file;
InputFile ¤t_input_file(void) {
if (input_files.empty()) {
std::string what = "input files empty";
throw std::out_of_range(/*&*/what);
}
return *(input_files.rbegin()); /* last one */
}
InputFile &new_input_file(void) {
if (!input_files.empty()) {
/* copy the last one, except for some fields */
{
InputFile &last = current_input_file();
input_files.push_back(last);
}
{
InputFile &last = current_input_file();
last.reset_on_dup();
}
}
else {
/* make a new one with defaults */
input_files.push_back(InputFile());
}
return current_input_file();
}
volatile int DIE = 0;
HiLoComboPass audio_hilopass;
// preemphsis emuluation
LowpassFilter audio_linear_preemphasis_pre[2];
LowpassFilter audio_linear_preemphasis_post[2];
double composite_preemphasis = 0; // analog artifacts related to anything that affects the raw composite signal i.e. CATV modulation
double composite_preemphasis_cut = 1000000;
double vhs_out_sharpen = 1.5;
bool vhs_head_switching = false;
double vhs_head_switching_point = 1.0 - ((4.5+0.01/*slight error, like most VHS tapes*/) / 262.5); // 4 scanlines NTSC up from vsync
double vhs_head_switching_phase = ((1.0-0.01/*slight error, like most VHS tapes*/) / 262.5); // 4 scanlines NTSC up from vsync
double vhs_head_switching_phase_noise = (((1.0 / 500)/*slight error, like most VHS tapes*/) / 262.5); // 1/500th of a scanline
bool composite_in_chroma_lowpass = true; // apply chroma lowpass before composite encode
bool composite_out_chroma_lowpass = true;
bool composite_out_chroma_lowpass_lite = true;
int video_yc_recombine = 0; // additional Y/C combine/sep phases (testing)
int video_color_fields = 4; // NTSC color framing
int video_chroma_noise = 0;
int video_chroma_phase_noise = 0;
int video_chroma_loss = 0;
int video_noise = 2;
int subcarrier_amplitude = 50;
int subcarrier_amplitude_back = 50;
double output_audio_hiss_db = -72;
double output_audio_linear_buzz = -42; // how loud the "buzz" is audible in dBFS (S/N). Ever notice on old VHS tapes (prior to Hi-Fi) you can almost hear the video signal sync pulses in the audio?
double output_audio_highpass = 20; // highpass to filter out below 20Hz
double output_audio_lowpass = 20000; // lowpass to filter out above 20KHz
double vhs_linear_high_boost = 0.25;
// NTS:
// VHS Hi-Fi: 20Hz - 20KHz (70dBFS S/N)
// VHS SP: 100Hz - 10KHz (42dBFS S/N)
// VHS LP: 100Hz - 7KHz (right??) (42dBFS S/N)
// VHS EP: 100Hz - 4KHz (42dBFS S/N)
bool output_vhs_hifi = true;
bool output_vhs_linear_stereo = false; // not common
bool output_vhs_linear_audio = false; // if true (non Hi-Fi) then we emulate hiss and noise of linear VHS tracks including the video sync pulses audible in the audio.
bool emulating_vhs = false;
bool emulating_preemphasis = true; // emulate preemphasis
bool emulating_deemphasis = true; // emulate deemphasis
bool nocolor_subcarrier = false; // if set, emulate subcarrier but do not decode back to color (debug)
bool nocolor_subcarrier_after_yc_sep = false;// if set, separate luma-chroma but do not decode back to color (debug)
bool vhs_chroma_vert_blend = true; // if set, and VHS, blend vertically the chroma scanlines (as the VHS format does)
bool vhs_svideo_out = false; // if not set, and VHS, video is recombined as if composite out on VCR
bool enable_composite_emulation = true; // if not set, video goes straight back out to the encoder.
bool enable_audio_emulation = true;
int output_audio_hiss_level = 0; // out of 10000
enum {
VHS_SP=0,
VHS_LP,
VHS_EP
};
int output_vhs_tape_speed = VHS_SP;
void sigma(int x) {
if (++DIE >= 20) abort();
}
void preset_PAL() {
output_field_rate.num = 50;
output_field_rate.den = 1;
output_height = 576;
output_width = 720;
output_pal = true;
output_ntsc = false;
}
void preset_NTSC() {
output_field_rate.num = 60000;
output_field_rate.den = 1001;
output_height = 480;
output_width = 720;
output_pal = false;
output_ntsc = true;
}
static void help(const char *arg0) {
fprintf(stderr,"%s [options]\n",arg0);
fprintf(stderr," -i <input file> you can specify more than one input file, in order of layering\n");
fprintf(stderr," -o <output file>\n");
fprintf(stderr," -d <n> Video delay buffer (n frames)\n");
fprintf(stderr," -tvstd <pal|ntsc>\n");
fprintf(stderr," -vhs Emulation of VHS artifacts\n");
fprintf(stderr," -vhs-hifi <0|1> (default on)\n");
fprintf(stderr," -vhs-speed <ep|lp|sp> (default sp)\n");
fprintf(stderr," -preemphasis <0|1> Enable preemphasis emulation\n");
fprintf(stderr," -deemphasis <0|1> Enable deepmhasis emulation\n");
fprintf(stderr," -nocolor-subcarrier Emulate color subcarrier but do not decode back (debug)\n");
fprintf(stderr," -nocolor-subcarrier-after-yc-sep Emulate Y/C subcarrier separation but do not decode back (debug)\n");
fprintf(stderr," -subcarrier-amp <0...100> Subcarrier amplitude (0 to 100 percent of luma)\n");
fprintf(stderr," -noise <0..100> Noise amplitude\n");
fprintf(stderr," -chroma-noise <0..100> Chroma noise amplitude\n");
fprintf(stderr," -audio-hiss <-120..0> Audio hiss in decibels (0=100%)\n");
fprintf(stderr," -vhs-linear-video-crosstalk <x> Emulate video crosstalk in audio. Loudness in dBFS (0=100%)\n");
fprintf(stderr," -chroma-phase-noise <x> Chroma phase noise (0...100)\n");
fprintf(stderr," -vhs-chroma-vblend <0|1> Vertically blend chroma scanlines (as VHS format does)\n");
fprintf(stderr," -vhs-svideo <0|1> Render VHS as if S-Video (luma and chroma separate out of VHS)\n");
fprintf(stderr," -yc-recomb <n> Recombine Y/C n-times\n");
fprintf(stderr," -a <n> Pick the n'th audio stream\n");
fprintf(stderr," -an Don't render any audio stream\n");
fprintf(stderr," -v <n> Pick the n'th video stream\n");
fprintf(stderr," -vn Don't render any video stream\n");
fprintf(stderr," -comp-pre <s> Composite preemphasis scale\n");
fprintf(stderr," -comp-cut <f> Composite preemphasis freq\n");
fprintf(stderr," -comp-catv Composite preemphasis preset, as if CATV #1\n");
fprintf(stderr," -comp-catv2 Composite preemphasis preset, as if CATV #2\n");
fprintf(stderr," -comp-catv3 Composite preemphasis preset, as if CATV #3\n");
fprintf(stderr," -comp-catv4 Composite preemphasis preset, as if CATV #4\n");
fprintf(stderr," -vi Render video at frame rate, interlaced\n");
fprintf(stderr," -vp Render video at field rate, progressive (with bob filter)\n");
fprintf(stderr," -chroma-dropout <x> Chroma scanline dropouts (0...10000)\n");
fprintf(stderr," -vhs-linear-high-boost <x> Boost high frequencies in VHS audio (linear tracks)\n");
fprintf(stderr," -vhs-head-switching <0|1> Enable/disable VHS head switching emulation\n");
fprintf(stderr," -vhs-head-switching-point <x> Head switching point (0....1)\n");
fprintf(stderr," -vhs-head-switching-phase <x> Head switching displacement (-1....1)\n");
fprintf(stderr," -vhs-head-switching-noise-level <x> Head switching noise (variation)\n");
fprintf(stderr," -422 Render in 4:2:2 colorspace\n");
fprintf(stderr," -420 Render in 4:2:0 colorspace (default)\n"); // dammit Premiere >:(
fprintf(stderr," -nocomp Don't apply emulation, just transcode\n");
fprintf(stderr," -ss <t> Start transcoding from t seconds\n");
fprintf(stderr," -se <t> Stop transcoding at t seconds\n");
fprintf(stderr," -t <t> Transcode only t seconds\n");
fprintf(stderr," -in-composite-lowpass <n> Enable/disable chroma lowpass on composite in\n");
fprintf(stderr," -out-composite-lowpass <n> Enable/disable chroma lowpass on composite out\n");
fprintf(stderr," -out-composite-lowpass-lite <n> Enable/disable chroma lowpass on composite out (lite)\n");
fprintf(stderr," -bkey-feedback <n> Black key feedback (black level <= N)\n");
fprintf(stderr," -comp-phase <n> NTSC subcarrier phase per scanline (0, 90, 180, or 270)\n");
fprintf(stderr,"\n");
fprintf(stderr," Output file will be up/down converted to 720x480 (NTSC 29.97fps) or 720x576 (PAL 25fps).\n");
fprintf(stderr," Output will be rendered as interlaced video.\n");
}
static unsigned long long audio_proc_count = 0;
static LowpassFilter audio_post_vhs_boost[2];
static inline int clips16(const int x) {
if (x < -32768)
return -32768;
else if (x > 32767)
return 32767;
return x;
}
void composite_audio_process(int16_t *audio,unsigned int samples) { // number of channels = output_audio_channels, sample rate = output_audio_rate. audio is interleaved.
assert(audio_hilopass.audiostate.size() >= output_audio_channels);
double linear_buzz = dBFS(output_audio_linear_buzz);
double hsync_hz = output_ntsc ? /*NTSC*/15734 : /*PAL*/15625;
int vsync_lines = output_ntsc ? /*NTSC*/525 : /*PAL*/625;
int vpulse_end = output_ntsc ? /*NTSC*/10 : /*PAL*/12;
double hpulse_end = output_ntsc ? /*NTSC*/(hsync_hz * (4.7/*us*/ / 1000000)) : /*PAL*/(hsync_hz * (4.0/*us*/ / 1000000));
for (unsigned int s=0;s < samples;s++,audio += output_audio_channels) {
for (unsigned int c=0;c < output_audio_channels;c++) {
double s;
s = (double)audio[c] / 32768;
/* lowpass filter */
s = audio_hilopass.audiostate[c].filter(s);
/* preemphasis */
if (emulating_preemphasis) {
for (unsigned int i=0;i < output_audio_channels;i++) {
s = s + audio_linear_preemphasis_pre[i].highpass(s);
}
}
/* that faint "buzzing" noise on linear tracks because of audio/video crosstalk */
if (!output_vhs_hifi && linear_buzz > 0.000000001) {
const unsigned int oversample = 16;
for (unsigned int oi=0;oi < oversample;oi++) {
double t = ((((double)audio_proc_count * oversample) + oi) * hsync_hz) / output_audio_rate / oversample;
double hpos = fmod(t,1.0);
int vline = (int)fmod(floor(t + 0.0001/*fudge*/ - hpos),(double)vsync_lines / 2);
bool pulse = false;
if (hpos < hpulse_end)
pulse = true; // HSYNC
if (vline < vpulse_end)
pulse = true; // VSYNC
if (pulse)
s -= linear_buzz / oversample / 2;
}
}
/* analog limiting (when the signal is too loud) */
if (s > 1.0)
s = 1.0;
else if (s < -1.0)
s = -1.0;
/* hiss */
if (output_audio_hiss_level != 0)
s += ((double)(((int)((unsigned int)rand() % ((output_audio_hiss_level * 2) + 1))) - output_audio_hiss_level)) / 20000;
/* some VCRs (at least mine) will boost higher frequencies if playing linear tracks */
if (!output_vhs_hifi && vhs_linear_high_boost > 0)
s += audio_post_vhs_boost[c].highpass(s) * vhs_linear_high_boost;
/* deemphasis */
if (emulating_deemphasis) {
for (unsigned int i=0;i < output_audio_channels;i++) {
s = audio_linear_preemphasis_post[i].lowpass(s);
}
}
audio[c] = clips16(s * 32768);
}
audio_proc_count++;
}
}
static int parse_argv(int argc,char **argv) {
const char *a;
int i;
for (i=1;i < argc;) {
a = argv[i++];
if (*a == '-') {
do { a++; } while (*a == '-');
if (!strcmp(a,"h") || !strcmp(a,"help")) {
help(argv[0]);
return 1;
}
else if (!strcmp(a,"comp-phase-offset")) {
video_scanline_phase_shift_offset = atoi(argv[i++]);
}
else if (!strcmp(a,"comp-phase")) {
video_scanline_phase_shift = atoi(argv[i++]);
if (!(video_scanline_phase_shift == 0 || video_scanline_phase_shift == 90 ||
video_scanline_phase_shift == 180 || video_scanline_phase_shift == 270)) {
fprintf(stderr,"Invalid phase\n");
return 1;
}
}
else if (!strcmp(a,"width")) {
a = argv[i++];
if (a == NULL) return 1;
output_width = (int)strtoul(a,NULL,0);