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ffmpeg.wasm-core
Commit 369cb092 authored by Anton Khirnov's avatar Anton Khirnov
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avconv: add support for audio filters. 

The FATE changes are all off-by-one due to different rounding being used
(lrintf vs av_rescale_q).
parent 9f26421b
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Showing with 359 additions and 504 deletions
avconv.c
View file @ 369cb092
......@@ -140,11 +140,6 @@ static float dts_delta_threshold = 10;
static int print_stats = 1;
static uint8_t *audio_buf;
static unsigned int allocated_audio_buf_size;
static uint8_t *async_buf;
static unsigned int allocated_async_buf_size;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
typedef struct InputFilter {
......@@ -211,6 +206,11 @@ typedef struct InputStream {
int resample_width;
int resample_pix_fmt;
int resample_sample_fmt;
int resample_sample_rate;
int resample_channels;
uint64_t resample_channel_layout;
/* a pool of free buffers for decoded data */
FrameBuffer *buffer_pool;
......@@ -249,7 +249,6 @@ typedef struct OutputStream {
AVBitStreamFilterContext *bitstream_filters;
AVCodec *enc;
int64_t max_frames;
AVFrame *output_frame;
AVFrame *filtered_frame;
/* video only */
......@@ -265,14 +264,6 @@ typedef struct OutputStream {
int forced_kf_count;
int forced_kf_index;
/* audio only */
int audio_resample;
AVAudioResampleContext *avr;
int resample_sample_fmt;
int resample_channels;
uint64_t resample_channel_layout;
int resample_sample_rate;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
FILE *logfile;
OutputFilter *filter;
......@@ -583,42 +574,150 @@ static void filter_release_buffer(AVFilterBuffer *fb)
unref_buffer(buf->ist, buf);
}
static char *choose_pixel_fmts(OutputStream *ost)
/**
* Define a function for building a string containing a list of
* allowed formats,
*/
#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator) \
static char *choose_ ## var ## s(OutputStream *ost) \
{ \
if (ost->st->codec->var != none) { \
get_name(ost->st->codec->var); \
return av_strdup(name); \
} else if (ost->enc->supported_list) { \
const type *p; \
AVIOContext *s = NULL; \
uint8_t *ret; \
int len; \
\
if (avio_open_dyn_buf(&s) < 0) \
exit_program(1); \
\
for (p = ost->enc->supported_list; *p != none; p++) { \
get_name(*p); \
avio_printf(s, "%s" separator, name); \
} \
len = avio_close_dyn_buf(s, &ret); \
ret[len - 1] = 0; \
return ret; \
} else \
return NULL; \
}
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
DEF_CHOOSE_FORMAT(enum PixelFormat, pix_fmt, pix_fmts, PIX_FMT_NONE,
GET_PIX_FMT_NAME, ":")
#define GET_SAMPLE_FMT_NAME(sample_fmt)\
const char *name = av_get_sample_fmt_name(sample_fmt)
DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")
#define GET_SAMPLE_RATE_NAME(rate)\
char name[16];\
snprintf(name, sizeof(name), "%d", rate);
DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
GET_SAMPLE_RATE_NAME, ",")
#define GET_CH_LAYOUT_NAME(ch_layout)\
char name[16];\
snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
GET_CH_LAYOUT_NAME, ",")
static int configure_audio_filters(FilterGraph *fg, AVFilterContext **in_filter,
AVFilterContext **out_filter)
{
if (ost->st->codec->pix_fmt != PIX_FMT_NONE) {
return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt));
} else if (ost->enc->pix_fmts) {
const enum PixelFormat *p;
AVIOContext *s = NULL;
uint8_t *ret;
int len;
InputStream *ist = fg->inputs[0]->ist;
OutputStream *ost = fg->outputs[0]->ost;
AVCodecContext *codec = ost->st->codec;
AVCodecContext *icodec = ist->st->codec;
char *sample_fmts, *sample_rates, *channel_layouts;
char args[256];
int ret;
if (avio_open_dyn_buf(&s) < 0)
exit_program(1);
avfilter_graph_free(&fg->graph);
if (!(fg->graph = avfilter_graph_alloc()))
return AVERROR(ENOMEM);
for (p = ost->enc->pix_fmts; *p != PIX_FMT_NONE; p++)
avio_printf(s, "%s:", av_get_pix_fmt_name(*p));
len = avio_close_dyn_buf(s, &ret);
ret[len - 1] = 0;
snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:"
"channel_layout=0x%"PRIx64, ist->st->time_base.num,
ist->st->time_base.den, icodec->sample_rate,
av_get_sample_fmt_name(icodec->sample_fmt), icodec->channel_layout);
ret = avfilter_graph_create_filter(&fg->inputs[0]->filter,
avfilter_get_by_name("abuffer"),
"src", args, NULL, fg->graph);
if (ret < 0)
return ret;
} else
return NULL;
ret = avfilter_graph_create_filter(&fg->outputs[0]->filter,
avfilter_get_by_name("abuffersink"),
"out", NULL, NULL, fg->graph);
if (ret < 0)
return ret;
*in_filter = fg->inputs[0]->filter;
*out_filter = fg->outputs[0]->filter;
if (codec->channels && !codec->channel_layout)
codec->channel_layout = av_get_default_channel_layout(codec->channels);
sample_fmts = choose_sample_fmts(ost);
sample_rates = choose_sample_rates(ost);
channel_layouts = choose_channel_layouts(ost);
if (sample_fmts || sample_rates || channel_layouts) {
AVFilterContext *format;
char args[256];
int len = 0;
if (sample_fmts)
len += snprintf(args + len, sizeof(args) - len, "sample_fmts=%s:",
sample_fmts);
if (sample_rates)
len += snprintf(args + len, sizeof(args) - len, "sample_rates=%s:",
sample_rates);
if (channel_layouts)
len += snprintf(args + len, sizeof(args) - len, "channel_layouts=%s:",
channel_layouts);
args[len - 1] = 0;
av_freep(&sample_fmts);
av_freep(&sample_rates);
av_freep(&channel_layouts);
ret = avfilter_graph_create_filter(&format,
avfilter_get_by_name("aformat"),
"aformat", args, NULL, fg->graph);
if (ret < 0)
return ret;
ret = avfilter_link(format, 0, fg->outputs[0]->filter, 0);
if (ret < 0)
return ret;
*out_filter = format;
}
return 0;
}
static int configure_video_filters(FilterGraph *fg)
static int configure_video_filters(FilterGraph *fg, AVFilterContext **in_filter,
AVFilterContext **out_filter)
{
InputStream *ist = fg->inputs[0]->ist;
OutputStream *ost = fg->outputs[0]->ost;
AVFilterContext *in_filter, *out_filter, *filter;
AVFilterContext *filter;
AVCodecContext *codec = ost->st->codec;
char *pix_fmts;
AVRational sample_aspect_ratio;
char args[255];
int ret;
avfilter_graph_free(&fg->graph);
fg->graph = avfilter_graph_alloc();
if (ist->st->sample_aspect_ratio.num) {
sample_aspect_ratio = ist->st->sample_aspect_ratio;
} else
......@@ -638,8 +737,8 @@ static int configure_video_filters(FilterGraph *fg)
"out", NULL, NULL, fg->graph);
if (ret < 0)
return ret;
in_filter = fg->inputs[0]->filter;
out_filter = fg->outputs[0]->filter;
*in_filter = fg->inputs[0]->filter;
*out_filter = fg->outputs[0]->filter;
if (codec->width || codec->height) {
snprintf(args, 255, "%d:%d:flags=0x%X",
......@@ -649,27 +748,51 @@ static int configure_video_filters(FilterGraph *fg)
if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
NULL, args, NULL, fg->graph)) < 0)
return ret;
if ((ret = avfilter_link(in_filter, 0, filter, 0)) < 0)
if ((ret = avfilter_link(*in_filter, 0, filter, 0)) < 0)
return ret;
in_filter = filter;
*in_filter = filter;
}
if ((pix_fmts = choose_pixel_fmts(ost))) {
if ((pix_fmts = choose_pix_fmts(ost))) {
if ((ret = avfilter_graph_create_filter(&filter,
avfilter_get_by_name("format"),
"format", pix_fmts, NULL,
fg->graph)) < 0)
return ret;
if ((ret = avfilter_link(filter, 0, out_filter, 0)) < 0)
if ((ret = avfilter_link(filter, 0, *out_filter, 0)) < 0)
return ret;
out_filter = filter;
*out_filter = filter;
av_freep(&pix_fmts);
}
snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
fg->graph->scale_sws_opts = av_strdup(args);
return 0;
}
static int configure_simple_filtergraph(FilterGraph *fg)
{
OutputStream *ost = fg->outputs[0]->ost;
AVFilterContext *in_filter, *out_filter;
int ret;
avfilter_graph_free(&fg->graph);
fg->graph = avfilter_graph_alloc();
switch (ost->st->codec->codec_type) {
case AVMEDIA_TYPE_VIDEO:
ret = configure_video_filters(fg, &in_filter, &out_filter);
break;
case AVMEDIA_TYPE_AUDIO:
ret = configure_audio_filters(fg, &in_filter, &out_filter);
break;
default: av_assert0(0);
}
if (ret < 0)
return ret;
if (ost->avfilter) {
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
......@@ -834,7 +957,7 @@ static int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFil
pad_idx = 0;
}
if ((pix_fmts = choose_pixel_fmts(ofilter->ost))) {
if ((pix_fmts = choose_pix_fmts(ofilter->ost))) {
AVFilterContext *filter;
if ((ret = avfilter_graph_create_filter(&filter,
avfilter_get_by_name("format"),
......@@ -931,7 +1054,8 @@ static int configure_complex_filters(void)
static int configure_filtergraph(FilterGraph *fg)
{
return fg->graph_desc ? configure_complex_filter(fg) : configure_video_filters(fg);
return fg->graph_desc ? configure_complex_filter(fg) :
configure_simple_filtergraph(fg);
}
static int ist_in_filtergraph(FilterGraph *fg, InputStream *ist)
......@@ -1009,13 +1133,6 @@ void exit_program(int ret)
}
output_streams[i]->bitstream_filters = NULL;
if (output_streams[i]->output_frame) {
AVFrame *frame = output_streams[i]->output_frame;
if (frame->extended_data != frame->data)
av_freep(&frame->extended_data);
av_freep(&frame);
}
av_freep(&output_streams[i]->avfilter);
av_freep(&output_streams[i]->filtered_frame);
av_freep(&output_streams[i]);
......@@ -1042,10 +1159,6 @@ void exit_program(int ret)
av_freep(&output_files);
uninit_opts();
av_free(audio_buf);
allocated_audio_buf_size = 0;
av_free(async_buf);
allocated_async_buf_size = 0;
avfilter_uninit();
avformat_network_deinit();
......@@ -1085,25 +1198,6 @@ static void assert_codec_experimental(AVCodecContext *c, int encoder)
}
}
static void choose_sample_fmt(AVStream *st, AVCodec *codec)
{
if (codec && codec->sample_fmts) {
const enum AVSampleFormat *p = codec->sample_fmts;
for (; *p != -1; p++) {
if (*p == st->codec->sample_fmt)
break;
}
if (*p == -1) {
av_log(NULL, AV_LOG_WARNING,
"Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
av_get_sample_fmt_name(st->codec->sample_fmt),
codec->name,
av_get_sample_fmt_name(codec->sample_fmts[0]));
st->codec->sample_fmt = codec->sample_fmts[0];
}
}
}
/**
* Update the requested input sample format based on the output sample format.
* This is currently only used to request float output from decoders which
......@@ -1144,26 +1238,6 @@ static void update_sample_fmt(AVCodecContext *dec, AVCodec *dec_codec,
}
}
static void choose_sample_rate(AVStream *st, AVCodec *codec)
{
if (codec && codec->supported_samplerates) {
const int *p = codec->supported_samplerates;
int best = 0;
int best_dist = INT_MAX;
for (; *p; p++) {
int dist = abs(st->codec->sample_rate - *p);
if (dist < best_dist) {
best_dist = dist;
best = *p;
}
}
if (best_dist) {
av_log(st->codec, AV_LOG_WARNING, "Requested sampling rate unsupported using closest supported (%d)\n", best);
}
st->codec->sample_rate = best;
}
}
static double
get_sync_ipts(const OutputStream *ost, int64_t pts)
{
......@@ -1235,107 +1309,24 @@ static int check_recording_time(OutputStream *ost)
return 1;
}
static void get_default_channel_layouts(OutputStream *ost, InputStream *ist)
{
char layout_name[256];
AVCodecContext *enc = ost->st->codec;
AVCodecContext *dec = ist->st->codec;
if (dec->channel_layout &&
av_get_channel_layout_nb_channels(dec->channel_layout) != dec->channels) {
av_get_channel_layout_string(layout_name, sizeof(layout_name),
dec->channels, dec->channel_layout);
av_log(NULL, AV_LOG_ERROR, "New channel layout (%s) is invalid\n",
layout_name);
dec->channel_layout = 0;
}
if (!dec->channel_layout) {
if (enc->channel_layout && dec->channels == enc->channels) {
dec->channel_layout = enc->channel_layout;
} else {
dec->channel_layout = av_get_default_channel_layout(dec->channels);
if (!dec->channel_layout) {
av_log(NULL, AV_LOG_FATAL, "Unable to find default channel "
"layout for Input Stream #%d.%d\n", ist->file_index,
ist->st->index);
exit_program(1);
}
}
av_get_channel_layout_string(layout_name, sizeof(layout_name),
dec->channels, dec->channel_layout);
av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream "
"#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name);
}
if (!enc->channel_layout) {
if (dec->channels == enc->channels) {
enc->channel_layout = dec->channel_layout;
return;
} else {
enc->channel_layout = av_get_default_channel_layout(enc->channels);
}
if (!enc->channel_layout) {
av_log(NULL, AV_LOG_FATAL, "Unable to find default channel layout "
"for Output Stream #%d.%d\n", ost->file_index,
ost->st->index);
exit_program(1);
}
av_get_channel_layout_string(layout_name, sizeof(layout_name),
enc->channels, enc->channel_layout);
av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Output Stream "
"#%d.%d : %s\n", ost->file_index, ost->st->index, layout_name);
}
}
static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
const uint8_t *buf, int buf_size)
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
AVFrame *frame)
{
AVCodecContext *enc = ost->st->codec;
AVFrame *frame = NULL;
AVPacket pkt;
int ret, got_packet;
int got_packet = 0;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (buf) {
if (!ost->output_frame) {
ost->output_frame = avcodec_alloc_frame();
if (!ost->output_frame) {
av_log(NULL, AV_LOG_FATAL, "out-of-memory in encode_audio_frame()\n");
exit_program(1);
}
}
frame = ost->output_frame;
if (frame->extended_data != frame->data)
av_freep(&frame->extended_data);
avcodec_get_frame_defaults(frame);
frame->nb_samples = buf_size /
(enc->channels * av_get_bytes_per_sample(enc->sample_fmt));
if ((ret = avcodec_fill_audio_frame(frame, enc->channels, enc->sample_fmt,
buf, buf_size, 1)) < 0) {
av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
exit_program(1);
}
if (!check_recording_time(ost))
return 0;
return;
if (frame->pts == AV_NOPTS_VALUE || audio_sync_method < 0)
frame->pts = ost->sync_opts;
ost->sync_opts += frame->nb_samples;
}
ost->sync_opts = frame->pts + frame->nb_samples;
got_packet = 0;
if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
exit_program(1);
......@@ -1353,207 +1344,6 @@ static int encode_audio_frame(AVFormatContext *s, OutputStream *ost,
audio_size += pkt.size;
}
return pkt.size;
}
static int alloc_audio_output_buf(AVCodecContext *dec, AVCodecContext *enc,
int nb_samples, int *buf_linesize)
{
int64_t audio_buf_samples;
int audio_buf_size;
/* calculate required number of samples to allocate */
audio_buf_samples = ((int64_t)nb_samples * enc->sample_rate + dec->sample_rate) /
dec->sample_rate;
audio_buf_samples = 4 * audio_buf_samples + 16; // safety factors for resampling
audio_buf_samples = FFMAX(audio_buf_samples, enc->frame_size);
if (audio_buf_samples > INT_MAX)
return AVERROR(EINVAL);
audio_buf_size = av_samples_get_buffer_size(buf_linesize, enc->channels,
audio_buf_samples,
enc->sample_fmt, 0);
if (audio_buf_size < 0)
return audio_buf_size;
av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
if (!audio_buf)
return AVERROR(ENOMEM);
return 0;
}
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
InputStream *ist, AVFrame *decoded_frame)
{
uint8_t *buftmp;
int size_out, frame_bytes, resample_changed, ret;
AVCodecContext *enc = ost->st->codec;
AVCodecContext *dec = ist->st->codec;
int osize = av_get_bytes_per_sample(enc->sample_fmt);
int isize = av_get_bytes_per_sample(dec->sample_fmt);
uint8_t *buf = decoded_frame->data[0];
int size = decoded_frame->nb_samples * dec->channels * isize;
int out_linesize = 0;
int buf_linesize = decoded_frame->linesize[0];
get_default_channel_layouts(ost, ist);
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples, &out_linesize) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
if (audio_sync_method > 1 ||
enc->channels != dec->channels ||
enc->channel_layout != dec->channel_layout ||
enc->sample_rate != dec->sample_rate ||
dec->sample_fmt != enc->sample_fmt)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_channel_layout != dec->channel_layout ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->avr) || resample_changed) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:0x%"PRIx64" to rate:%d fmt:%s ch:%d chl:0x%"PRIx64"\n",
ist->file_index, ist->st->index,
ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt),
ost->resample_channels, ost->resample_channel_layout,
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt),
dec->channels, dec->channel_layout);
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_channel_layout = dec->channel_layout;
ost->resample_sample_rate = dec->sample_rate;
if (ost->avr)
avresample_close(ost->avr);
}
/* 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_channel_layout == enc->channel_layout &&
ost->resample_sample_rate == enc->sample_rate) {
ost->audio_resample = 0;
} else if (ost->audio_resample) {
if (!ost->avr) {
ost->avr = avresample_alloc_context();
if (!ost->avr) {
av_log(NULL, AV_LOG_FATAL, "Error allocating context for libavresample\n");
exit_program(1);
}
}
av_opt_set_int(ost->avr, "in_channel_layout", dec->channel_layout, 0);
av_opt_set_int(ost->avr, "in_sample_fmt", dec->sample_fmt, 0);
av_opt_set_int(ost->avr, "in_sample_rate", dec->sample_rate, 0);
av_opt_set_int(ost->avr, "out_channel_layout", enc->channel_layout, 0);
av_opt_set_int(ost->avr, "out_sample_fmt", enc->sample_fmt, 0);
av_opt_set_int(ost->avr, "out_sample_rate", enc->sample_rate, 0);
if (audio_sync_method > 1)
av_opt_set_int(ost->avr, "force_resampling", 1, 0);
/* if both the input and output formats are s16 or u8, use s16 as
the internal sample format */
if (av_get_bytes_per_sample(dec->sample_fmt) <= 2 &&
av_get_bytes_per_sample(enc->sample_fmt) <= 2) {
av_opt_set_int(ost->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
}
ret = avresample_open(ost->avr);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "Error opening libavresample\n");
exit_program(1);
}
}
}
if (audio_sync_method > 0) {
double delta = get_sync_ipts(ost, ist->last_dts) * enc->sample_rate - ost->sync_opts -
av_fifo_size(ost->fifo) / (enc->channels * osize);
int idelta = delta * dec->sample_rate / enc->sample_rate;
int byte_delta = idelta * isize * dec->channels;
// FIXME resample delay
if (fabs(delta) > 50) {
if (ist->is_start || fabs(delta) > audio_drift_threshold*enc->sample_rate) {
if (byte_delta < 0) {
byte_delta = FFMAX(byte_delta, -size);
size += byte_delta;
buf -= byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n",
-byte_delta / (isize * dec->channels));
if (!size)
return;
ist->is_start = 0;
} else {
av_fast_malloc(&async_buf, &allocated_async_buf_size,
byte_delta + size);
if (!async_buf) {
av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
exit_program(1);
}
if (alloc_audio_output_buf(dec, enc, decoded_frame->nb_samples + idelta, &out_linesize) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error allocating audio buffer\n");
exit_program(1);
}
ist->is_start = 0;
generate_silence(async_buf, dec->sample_fmt, byte_delta);
memcpy(async_buf + byte_delta, buf, size);
buf = async_buf;
size += byte_delta;
buf_linesize = allocated_async_buf_size;
av_log(NULL, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", idelta);
}
} else if (audio_sync_method > 1) {
int comp = av_clip(delta, -audio_sync_method, audio_sync_method);
av_log(NULL, AV_LOG_VERBOSE, "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));
avresample_set_compensation(ost->avr, comp, enc->sample_rate);
}
}
} else if (audio_sync_method == 0)
ost->sync_opts = lrintf(get_sync_ipts(ost, ist->last_dts) * enc->sample_rate) -
av_fifo_size(ost->fifo) / (enc->channels * osize); // FIXME wrong
if (ost->audio_resample) {
buftmp = audio_buf;
size_out = avresample_convert(ost->avr, (void **)&buftmp,
allocated_audio_buf_size, out_linesize,
(void **)&buf, buf_linesize,
size / (dec->channels * isize));
size_out = size_out * enc->channels * osize;
} else {
buftmp = buf;
size_out = size;
}
/* now encode as many frames as possible */
if (!(enc->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
/* output resampled raw samples */
if (av_fifo_realloc2(ost->fifo, av_fifo_size(ost->fifo) + size_out) < 0) {
av_log(NULL, AV_LOG_FATAL, "av_fifo_realloc2() failed\n");
exit_program(1);
}
av_fifo_generic_write(ost->fifo, buftmp, size_out, NULL);
frame_bytes = enc->frame_size * osize * enc->channels;
while (av_fifo_size(ost->fifo) >= frame_bytes) {
av_fifo_generic_read(ost->fifo, audio_buf, frame_bytes, NULL);
encode_audio_frame(s, ost, audio_buf, frame_bytes);
}
} else {
encode_audio_frame(s, ost, buftmp, size_out);
}
}
static void pre_process_video_frame(InputStream *ist, AVPicture *picture, void **bufp)
......@@ -1852,6 +1642,7 @@ static int poll_filters(void)
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
OutputFile *of = output_files[ost->file_index];
int ret = 0;
if (!ost->filter || ost->is_past_recording_time)
continue;
......@@ -1862,11 +1653,29 @@ static int poll_filters(void)
avcodec_get_frame_defaults(ost->filtered_frame);
filtered_frame = ost->filtered_frame;
while (av_buffersink_read(ost->filter->filter, &picref) >= 0) {
while (ret >= 0) {
if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
ret = av_buffersink_read_samples(ost->filter->filter, &picref,
ost->st->codec->frame_size);
else
ret = av_buffersink_read(ost->filter->filter, &picref);
if (ret < 0)
break;
avfilter_copy_buf_props(filtered_frame, picref);
if (ost->enc->type == AVMEDIA_TYPE_VIDEO)
filtered_frame->pts = av_rescale_q(picref->pts,
ost->filter->filter->inputs[0]->time_base,
AV_TIME_BASE_Q);
else if (picref->pts != AV_NOPTS_VALUE)
filtered_frame->pts = av_rescale_q(picref->pts,
ost->filter->filter->inputs[0]->time_base,
ost->st->codec->time_base) -
av_rescale_q(of->start_time,
AV_TIME_BASE_Q,
ost->st->codec->time_base);
if (of->start_time && filtered_frame->pts < of->start_time)
return 0;
......@@ -1882,8 +1691,11 @@ static int poll_filters(void)
if (vstats_filename && frame_size)
do_video_stats(of->ctx, ost, frame_size);
break;
case AVMEDIA_TYPE_AUDIO:
do_audio_out(of->ctx, ost, filtered_frame);
break;
default:
// TODO support audio/subtitle filters
// TODO support subtitle filters
av_assert0(0);
}
......@@ -2033,38 +1845,38 @@ static void flush_encoders(void)
continue;
for (;;) {
AVPacket pkt;
int fifo_bytes, got_packet;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
int (*encode)(AVCodecContext*, AVPacket*, const AVFrame*, int*) = NULL;
const char *desc;
int64_t *size;
switch (ost->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
fifo_bytes = av_fifo_size(ost->fifo);
if (fifo_bytes > 0) {
/* encode any samples remaining in fifo */
int frame_bytes = fifo_bytes;
av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
encode_audio_frame(os, ost, audio_buf, frame_bytes);
} else {
/* flush encoder with NULL frames until it is done
returning packets */
if (encode_audio_frame(os, ost, NULL, 0) == 0) {
stop_encoding = 1;
break;
}
}
encode = avcodec_encode_audio2;
desc = "Audio";
size = &audio_size;
break;
case AVMEDIA_TYPE_VIDEO:
ret = avcodec_encode_video2(enc, &pkt, NULL, &got_packet);
encode = avcodec_encode_video2;
desc = "Video";
size = &video_size;
break;
default:
stop_encoding = 1;
}
if (encode) {
AVPacket pkt;
int got_packet;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
ret = encode(enc, &pkt, NULL, &got_packet);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n");
av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
exit_program(1);
}
video_size += ret;
*size += ret;
if (ost->logfile && enc->stats_out) {
fprintf(ost->logfile, "%s", enc->stats_out);
}
......@@ -2077,10 +1889,8 @@ static void flush_encoders(void)
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
write_frame(os, &pkt, ost);
break;
default:
stop_encoding = 1;
}
if (stop_encoding)
break;
}
......@@ -2172,12 +1982,30 @@ static void rate_emu_sleep(InputStream *ist)
}
}
static int guess_input_channel_layout(InputStream *ist)
{
AVCodecContext *dec = ist->st->codec;
if (!dec->channel_layout) {
char layout_name[256];
dec->channel_layout = av_get_default_channel_layout(dec->channels);
if (!dec->channel_layout)
return 0;
av_get_channel_layout_string(layout_name, sizeof(layout_name),
dec->channels, dec->channel_layout);
av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream "
"#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name);
}
return 1;
}
static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
{
AVFrame *decoded_frame;
AVCodecContext *avctx = ist->st->codec;
int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
int i, ret;
int i, ret, resample_changed;
if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
......@@ -2192,6 +2020,9 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
if (!*got_output) {
/* no audio frame */
if (!pkt->size)
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_buffer(ist->filters[i]->filter, NULL);
return ret;
}
......@@ -2199,11 +2030,10 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
the decoder could be delaying output by a packet or more. */
if (decoded_frame->pts != AV_NOPTS_VALUE)
ist->next_dts = decoded_frame->pts;
/* increment next_dts to use for the case where the input stream does not
have timestamps or there are multiple frames in the packet */
ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /
avctx->sample_rate;
else if (pkt->pts != AV_NOPTS_VALUE) {
decoded_frame->pts = pkt->pts;
pkt->pts = AV_NOPTS_VALUE;
}
// preprocess audio (volume)
if (audio_volume != 256) {
......@@ -2265,14 +2095,50 @@ static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
rate_emu_sleep(ist);
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
resample_changed = ist->resample_sample_fmt != decoded_frame->format ||
ist->resample_channels != avctx->channels ||
ist->resample_channel_layout != decoded_frame->channel_layout ||
ist->resample_sample_rate != decoded_frame->sample_rate;
if (resample_changed) {
char layout1[64], layout2[64];
if (!check_output_constraints(ist, ost) || !ost->encoding_needed)
continue;
do_audio_out(output_files[ost->file_index]->ctx, ost, ist, decoded_frame);
if (!guess_input_channel_layout(ist)) {
av_log(NULL, AV_LOG_FATAL, "Unable to find default channel "
"layout for Input Stream #%d.%d\n", ist->file_index,
ist->st->index);
exit_program(1);
}
decoded_frame->channel_layout = avctx->channel_layout;
av_get_channel_layout_string(layout1, sizeof(layout1), ist->resample_channels,
ist->resample_channel_layout);
av_get_channel_layout_string(layout2, sizeof(layout2), avctx->channels,
decoded_frame->channel_layout);
av_log(NULL, AV_LOG_INFO,
"Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:%s to rate:%d fmt:%s ch:%d chl:%s\n",
ist->file_index, ist->st->index,
ist->resample_sample_rate, av_get_sample_fmt_name(ist->resample_sample_fmt),
ist->resample_channels, layout1,
decoded_frame->sample_rate, av_get_sample_fmt_name(decoded_frame->format),
avctx->channels, layout2);
ist->resample_sample_fmt = decoded_frame->format;
ist->resample_sample_rate = decoded_frame->sample_rate;
ist->resample_channel_layout = decoded_frame->channel_layout;
ist->resample_channels = avctx->channels;
for (i = 0; i < nb_filtergraphs; i++)
if (ist_in_filtergraph(filtergraphs[i], ist) &&
configure_filtergraph(filtergraphs[i]) < 0) {
av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n");
exit_program(1);
}
}
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_write_frame(ist->filters[i]->filter, decoded_frame);
return ret;
}
......@@ -2550,17 +2416,6 @@ static int init_input_stream(int ist_index, char *error, int error_len)
}
assert_codec_experimental(ist->st->codec, 0);
assert_avoptions(ist->opts);
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
if (ost->source_index == ist_index) {
if (!ist->st->codec->channel_layout || !ost->st->codec->channel_layout)
get_default_channel_layouts(ost, ist);
break;
}
}
}
}
ist->last_dts = ist->st->avg_frame_rate.num ? - ist->st->codec->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0;
......@@ -2732,45 +2587,26 @@ static int transcode_init(void)
ist->decoding_needed = 1;
ost->encoding_needed = 1;
switch (codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ost->fifo = av_fifo_alloc(1024);
if (!ost->fifo) {
return AVERROR(ENOMEM);
}
if (!codec->sample_rate)
codec->sample_rate = icodec->sample_rate;
choose_sample_rate(ost->st, ost->enc);
codec->time_base = (AVRational){ 1, codec->sample_rate };
if (codec->sample_fmt == AV_SAMPLE_FMT_NONE)
codec->sample_fmt = icodec->sample_fmt;
choose_sample_fmt(ost->st, ost->enc);
if (!codec->channels)
codec->channels = icodec->channels;
if (!codec->channel_layout)
codec->channel_layout = icodec->channel_layout;
if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels)
codec->channel_layout = 0;
icodec->request_channels = codec-> channels;
ost->resample_sample_fmt = icodec->sample_fmt;
ost->resample_sample_rate = icodec->sample_rate;
ost->resample_channels = icodec->channels;
ost->resample_channel_layout = icodec->channel_layout;
break;
case AVMEDIA_TYPE_VIDEO:
if (!ost->filter) {
if (!ost->filter &&
(codec->codec_type == AVMEDIA_TYPE_VIDEO ||
codec->codec_type == AVMEDIA_TYPE_AUDIO)) {
FilterGraph *fg;
fg = init_simple_filtergraph(ist, ost);
if (configure_video_filters(fg)) {
if (configure_simple_filtergraph(fg)) {
av_log(NULL, AV_LOG_FATAL, "Error opening filters!\n");
exit(1);
}
}
switch (codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
codec->sample_fmt = ost->filter->filter->inputs[0]->format;
codec->sample_rate = ost->filter->filter->inputs[0]->sample_rate;
codec->channel_layout = ost->filter->filter->inputs[0]->channel_layout;
codec->channels = av_get_channel_layout_nb_channels(codec->channel_layout);
codec->time_base = (AVRational){ 1, codec->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
/*
* We want CFR output if and only if one of those is true:
* 1) user specified output framerate with -r
......@@ -3227,12 +3063,8 @@ static int transcode(void)
fclose(ost->logfile);
ost->logfile = NULL;
}
av_fifo_free(ost->fifo); /* works even if fifo is not
initialized but set to zero */
av_freep(&ost->st->codec->subtitle_header);
av_free(ost->forced_kf_pts);
if (ost->avr)
avresample_free(&ost->avr);
av_dict_free(&ost->opts);
}
}
......@@ -3564,6 +3396,14 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
break;
case AVMEDIA_TYPE_AUDIO:
guess_input_channel_layout(ist);
ist->resample_sample_fmt = dec->sample_fmt;
ist->resample_sample_rate = dec->sample_rate;
ist->resample_channels = dec->channels;
ist->resample_channel_layout = dec->channel_layout;
break;
case AVMEDIA_TYPE_DATA:
case AVMEDIA_TYPE_SUBTITLE:
case AVMEDIA_TYPE_ATTACHMENT:
......@@ -4095,7 +3935,7 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc)
audio_enc->codec_type = AVMEDIA_TYPE_AUDIO;
if (!ost->stream_copy) {
char *sample_fmt = NULL;
char *sample_fmt = NULL, *filters = NULL;;
MATCH_PER_STREAM_OPT(audio_channels, i, audio_enc->channels, oc, st);
......@@ -4107,6 +3947,10 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc)
}
MATCH_PER_STREAM_OPT(audio_sample_rate, i, audio_enc->sample_rate, oc, st);
MATCH_PER_STREAM_OPT(filters, str, filters, oc, st);
if (filters)
ost->avfilter = av_strdup(filters);
}
return ost;
......@@ -4850,6 +4694,11 @@ static int opt_video_filters(OptionsContext *o, const char *opt, const char *arg
return parse_option(o, "filter:v", arg, options);
}
static int opt_audio_filters(OptionsContext *o, const char *opt, const char *arg)
{
return parse_option(o, "filter:a", arg, options);
}
static int opt_vsync(const char *opt, const char *arg)
{
if (!av_strcasecmp(arg, "cfr")) video_sync_method = VSYNC_CFR;
......@@ -5022,6 +4871,7 @@ static const OptionDef options[] = {
{ "vol", OPT_INT | HAS_ARG | OPT_AUDIO, {(void*)&audio_volume}, "change audio volume (256=normal)" , "volume" }, //
{ "sample_fmt", HAS_ARG | OPT_EXPERT | OPT_AUDIO | OPT_SPEC | OPT_STRING, {.off = OFFSET(sample_fmts)}, "set sample format", "format" },
{ "channel_layout", HAS_ARG | OPT_EXPERT | OPT_AUDIO | OPT_FUNC2, {(void*)opt_channel_layout}, "set channel layout", "layout" },
{ "af", HAS_ARG | OPT_AUDIO | OPT_FUNC2, {(void*)opt_audio_filters}, "audio filters", "filter list" },
/* subtitle options */
{ "sn", OPT_BOOL | OPT_SUBTITLE | OPT_OFFSET, {.off = OFFSET(subtitle_disable)}, "disable subtitle" },
......
doc/avconv.texi
View file @ 369cb092
......@@ -435,6 +435,11 @@ Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
Set the audio sample format. Use @code{-sample_fmts} to get a list
of supported sample formats.
@item -af @var{filter_graph} (@emph{output})
@var{filter_graph} is a description of the filter graph to apply to
the input audio.
Use the option "-filters" to show all the available filters (including
also sources and sinks). This is an alias for @code{-filter:a}.
@end table
@section Advanced Audio options:
......
tests/ref/fate/adpcm-ima-amv
View file @ 369cb092
......@@ -3,7 +3,7 @@
0, 1378, 1378, 1378, 2756, 0x8462443f
0, 2756, 2756, 1378, 2756, 0x9f493ba6
0, 4134, 4134, 1378, 2756, 0x634e5f06
0, 5512, 5512, 1380, 2760, 0x51f35cd4
0, 5513, 5513, 1380, 2760, 0x51f35cd4
0, 6891, 6891, 1378, 2756, 0x011c51e5
0, 8269, 8269, 1378, 2756, 0x8c2c198c
0, 9647, 9647, 1378, 2756, 0x2b4a3397
......@@ -19,7 +19,7 @@
0, 23428, 23428, 1378, 2756, 0x96346ba6
0, 24806, 24806, 1378, 2756, 0x3d54543b
0, 26184, 26184, 1380, 2760, 0x601786e1
0, 27562, 27562, 1378, 2756, 0xf22a5793
0, 27563, 27563, 1378, 2756, 0xf22a5793
0, 28941, 28941, 1378, 2756, 0x21f54d49
0, 30319, 30319, 1378, 2756, 0x0c6d4399
0, 31697, 31697, 1378, 2756, 0x17282f8e
......@@ -35,7 +35,7 @@
0, 45478, 45478, 1380, 2760, 0x98a76091
0, 46856, 46856, 1378, 2756, 0x5d357141
0, 48234, 48234, 1378, 2756, 0x65ea2657
0, 49612, 49612, 1378, 2756, 0xb5e1334a
0, 49613, 49613, 1378, 2756, 0xb5e1334a
0, 50991, 50991, 1378, 2756, 0x32cd5d91
0, 52369, 52369, 1378, 2756, 0xdc23722b
0, 53747, 53747, 1378, 2756, 0x2ba34684
......@@ -51,7 +51,7 @@
0, 67528, 67528, 1378, 2756, 0x39c2586c
0, 68906, 68906, 1378, 2756, 0x7ffc46e5
0, 70284, 70284, 1378, 2756, 0xa2766664
0, 71662, 71662, 1378, 2756, 0xacb50c6c
0, 71663, 71663, 1378, 2756, 0xacb50c6c
0, 73041, 73041, 1378, 2756, 0x7f659084
0, 74419, 74419, 1378, 2756, 0xc72e6a12
0, 75797, 75797, 1380, 2760, 0xdb6944df
......@@ -67,7 +67,7 @@
0, 89578, 89578, 1378, 2756, 0x5c17abef
0, 90956, 90956, 1378, 2756, 0xb3235184
0, 92334, 92334, 1378, 2756, 0xdabb64a6
0, 93712, 93712, 1378, 2756, 0xa95dc58d
0, 93713, 93713, 1378, 2756, 0xa95dc58d
0, 95091, 95091, 1380, 2760, 0x8e7ac9eb
0, 96469, 96469, 1378, 2756, 0x492b658e
0, 97847, 97847, 1378, 2756, 0x377483ab
......@@ -83,7 +83,7 @@
0, 111628, 111628, 1378, 2756, 0xf0de66ae
0, 113006, 113006, 1378, 2756, 0xeabf3c32
0, 114384, 114384, 1378, 2756, 0xe98e81d1
0, 115762, 115762, 1380, 2760, 0x56aa5889
0, 115763, 115763, 1380, 2760, 0x56aa5889
0, 117141, 117141, 1378, 2756, 0x4fd34c0e
0, 118519, 118519, 1378, 2756, 0x67cf6912
0, 119897, 119897, 1378, 2756, 0xfa944def
......@@ -99,7 +99,7 @@
0, 133678, 133678, 1378, 2756, 0x2e0e3f2e
0, 135056, 135056, 1380, 2760, 0xdf534478
0, 136434, 136434, 1378, 2756, 0xca000a2e
0, 137812, 137812, 1378, 2756, 0x87472df3
0, 137813, 137813, 1378, 2756, 0x87472df3
0, 139191, 139191, 1378, 2756, 0x16733810
0, 140569, 140569, 1378, 2756, 0xfa0734b4
0, 141947, 141947, 1378, 2756, 0x5eff3fc4
......@@ -115,7 +115,7 @@
0, 155728, 155728, 1380, 2760, 0x8eca4bdb
0, 157106, 157106, 1378, 2756, 0x62bd4162
0, 158484, 158484, 1378, 2756, 0x9f744aa4
0, 159862, 159862, 1378, 2756, 0x0f3f6409
0, 159863, 159863, 1378, 2756, 0x0f3f6409
0, 161241, 161241, 1378, 2756, 0x3fee827a
0, 162619, 162619, 1378, 2756, 0x48a0ac19
0, 163997, 163997, 1378, 2756, 0x8e4ce0d0
......@@ -131,7 +131,7 @@
0, 177778, 177778, 1378, 2756, 0xab01fb12
0, 179156, 179156, 1378, 2756, 0x04cffe5c
0, 180534, 180534, 1378, 2756, 0xef661c5e
0, 181912, 181912, 1378, 2756, 0x094c5fc5
0, 181913, 181913, 1378, 2756, 0x094c5fc5
0, 183291, 183291, 1378, 2756, 0xe0c1486a
0, 184669, 184669, 1380, 2760, 0x8c3535b7
0, 186047, 186047, 1378, 2756, 0x594934aa
......@@ -147,7 +147,7 @@
0, 199828, 199828, 1378, 2756, 0x448e681d
0, 201206, 201206, 1378, 2756, 0x0ba9826e
0, 202584, 202584, 1378, 2756, 0x049f36fa
0, 203962, 203962, 1378, 2756, 0x096a2b62
0, 203963, 203963, 1378, 2756, 0x096a2b62
0, 205341, 205341, 1380, 2760, 0x579e2035
0, 206719, 206719, 1378, 2756, 0xd13e30e1
0, 208097, 208097, 1378, 2756, 0x30b6412b
......
tests/ref/fate/adpcm-ima-smjpeg
View file @ 369cb092
......@@ -27,7 +27,7 @@
0, 12789, 12789, 512, 1024, 0x3866b03b
0, 13296, 13296, 512, 1024, 0xbc005403
0, 13803, 13803, 512, 1024, 0xe9dfcc51
0, 14332, 14332, 512, 1024, 0x83c837cb
0, 14333, 14333, 512, 1024, 0x83c837cb
0, 14840, 14840, 512, 1024, 0xfa649580
0, 15347, 15347, 512, 1024, 0x519452ea
0, 15854, 15854, 512, 1024, 0xd4978774
......@@ -108,7 +108,7 @@
0, 54265, 54265, 512, 1024, 0xd6ece2a1
0, 54772, 54772, 512, 1024, 0x33ab9553
0, 55279, 55279, 512, 1024, 0xd50c73a6
0, 55786, 55786, 512, 1024, 0xfe25b63a
0, 55787, 55787, 512, 1024, 0xfe25b63a
0, 56316, 56316, 512, 1024, 0x7e2959e3
0, 56823, 56823, 512, 1024, 0xa4c07b34
0, 57330, 57330, 512, 1024, 0xd6d8f15c
......@@ -139,7 +139,7 @@
0, 70141, 70141, 512, 1024, 0xc434d238
0, 70648, 70648, 512, 1024, 0xe9ad7562
0, 71155, 71155, 512, 1024, 0xb51b6b50
0, 71662, 71662, 512, 1024, 0xe70aecd3
0, 71663, 71663, 512, 1024, 0xe70aecd3
0, 72192, 72192, 512, 1024, 0x03c816b2
0, 72699, 72699, 512, 1024, 0x869fdf25
0, 73206, 73206, 512, 1024, 0xd40a0a62
......@@ -170,7 +170,7 @@
0, 85995, 85995, 512, 1024, 0xaa919ccc
0, 86524, 86524, 512, 1024, 0x15993dbc
0, 87031, 87031, 512, 1024, 0xbe01a7b9
0, 87538, 87538, 512, 1024, 0xefe93c09
0, 87539, 87539, 512, 1024, 0xefe93c09
0, 88046, 88046, 512, 1024, 0x1bb566e5
0, 88575, 88575, 512, 1024, 0x15ce6237
0, 89082, 89082, 512, 1024, 0xa8552e66
......@@ -201,7 +201,7 @@
0, 101871, 101871, 512, 1024, 0x1e01fb02
0, 102378, 102378, 512, 1024, 0x4ed2c1d8
0, 102907, 102907, 512, 1024, 0xf2fdbe63
0, 103414, 103414, 512, 1024, 0x8d6f63a1
0, 103415, 103415, 512, 1024, 0x8d6f63a1
0, 103922, 103922, 512, 1024, 0xded468d9
0, 104429, 104429, 512, 1024, 0xccad839e
0, 104958, 104958, 512, 1024, 0xdde7c082
......@@ -232,7 +232,7 @@
0, 117747, 117747, 512, 1024, 0xb01e2365
0, 118254, 118254, 512, 1024, 0x14e200d3
0, 118783, 118783, 512, 1024, 0xd1184c98
0, 119290, 119290, 512, 1024, 0xef9140e9
0, 119291, 119291, 512, 1024, 0xef9140e9
0, 119798, 119798, 512, 1024, 0x4cbb645e
0, 120305, 120305, 512, 1024, 0xe7fe2f06
0, 120812, 120812, 512, 1024, 0xf8c45028
......@@ -263,7 +263,7 @@
0, 133623, 133623, 512, 1024, 0xa5099687
0, 134130, 134130, 512, 1024, 0xbff10707
0, 134637, 134637, 512, 1024, 0x37c4ffc0
0, 135166, 135166, 512, 1024, 0xf9fb6caa
0, 135167, 135167, 512, 1024, 0xf9fb6caa
0, 135674, 135674, 512, 1024, 0x3b6a3a1f
0, 136181, 136181, 512, 1024, 0x83431edb
0, 136688, 136688, 512, 1024, 0x1eb713cf
......@@ -344,7 +344,7 @@
0, 175099, 175099, 512, 1024, 0xd009a7ca
0, 175606, 175606, 512, 1024, 0xb6d5a938
0, 176113, 176113, 512, 1024, 0xf3d45e47
0, 176620, 176620, 512, 1024, 0xea8e04fc
0, 176621, 176621, 512, 1024, 0xea8e04fc
0, 177150, 177150, 512, 1024, 0x0b928bd8
0, 177657, 177657, 512, 1024, 0x0f02caec
0, 178164, 178164, 512, 1024, 0xe2b137a8
......
tests/ref/fate/smjpeg
View file @ 369cb092
......@@ -34,7 +34,7 @@
1, 12789, 12789, 512, 1024, 0x3866b03b
1, 13296, 13296, 512, 1024, 0xbc005403
1, 13803, 13803, 512, 1024, 0xe9dfcc51
1, 14332, 14332, 512, 1024, 0x83c837cb
1, 14333, 14333, 512, 1024, 0x83c837cb
0, 666, 666, 0, 6808, 0x1f6eb7c3
1, 14840, 14840, 512, 1024, 0xfa649580
1, 15347, 15347, 512, 1024, 0x519452ea
......@@ -132,7 +132,7 @@
1, 54265, 54265, 512, 1024, 0xd6ece2a1
1, 54772, 54772, 512, 1024, 0x33ab9553
1, 55279, 55279, 512, 1024, 0xd50c73a6
1, 55786, 55786, 512, 1024, 0xfe25b63a
1, 55787, 55787, 512, 1024, 0xfe25b63a
1, 56316, 56316, 512, 1024, 0x7e2959e3
0, 2555, 2555, 0, 7213, 0x97b89994
1, 56823, 56823, 512, 1024, 0xa4c07b34
......@@ -170,7 +170,7 @@
1, 70648, 70648, 512, 1024, 0xe9ad7562
0, 3222, 3222, 0, 7065, 0x45035c5c
1, 71155, 71155, 512, 1024, 0xb51b6b50
1, 71662, 71662, 512, 1024, 0xe70aecd3
1, 71663, 71663, 512, 1024, 0xe70aecd3
1, 72192, 72192, 512, 1024, 0x03c816b2
1, 72699, 72699, 512, 1024, 0x869fdf25
1, 73206, 73206, 512, 1024, 0xd40a0a62
......@@ -207,7 +207,7 @@
1, 85995, 85995, 512, 1024, 0xaa919ccc
1, 86524, 86524, 512, 1024, 0x15993dbc
1, 87031, 87031, 512, 1024, 0xbe01a7b9
1, 87538, 87538, 512, 1024, 0xefe93c09
1, 87539, 87539, 512, 1024, 0xefe93c09
1, 88046, 88046, 512, 1024, 0x1bb566e5
0, 4000, 4000, 0, 7246, 0x519a7a3c
1, 88575, 88575, 512, 1024, 0x15ce6237
......@@ -245,7 +245,7 @@
1, 102378, 102378, 512, 1024, 0x4ed2c1d8
0, 4666, 4666, 111, 7104, 0xc0d14f78
1, 102907, 102907, 512, 1024, 0xf2fdbe63
1, 103414, 103414, 512, 1024, 0x8d6f63a1
1, 103415, 103415, 512, 1024, 0x8d6f63a1
1, 103922, 103922, 512, 1024, 0xded468d9
1, 104429, 104429, 512, 1024, 0xccad839e
1, 104958, 104958, 512, 1024, 0xdde7c082
......@@ -282,7 +282,7 @@
1, 117747, 117747, 512, 1024, 0xb01e2365
1, 118254, 118254, 512, 1024, 0x14e200d3
1, 118783, 118783, 512, 1024, 0xd1184c98
1, 119290, 119290, 512, 1024, 0xef9140e9
1, 119291, 119291, 512, 1024, 0xef9140e9
1, 119798, 119798, 512, 1024, 0x4cbb645e
0, 5444, 5444, 111, 7106, 0xd1a83ddc
1, 120305, 120305, 512, 1024, 0xe7fe2f06
......@@ -320,7 +320,7 @@
1, 134130, 134130, 512, 1024, 0xbff10707
1, 134637, 134637, 512, 1024, 0x37c4ffc0
0, 6111, 6111, 111, 7283, 0x19dd7319
1, 135166, 135166, 512, 1024, 0xf9fb6caa
1, 135167, 135167, 512, 1024, 0xf9fb6caa
1, 135674, 135674, 512, 1024, 0x3b6a3a1f
1, 136181, 136181, 512, 1024, 0x83431edb
1, 136688, 136688, 512, 1024, 0x1eb713cf
......@@ -418,7 +418,7 @@
1, 175606, 175606, 512, 1024, 0xb6d5a938
1, 176113, 176113, 512, 1024, 0xf3d45e47
0, 8000, 8000, 111, 2153, 0xf4e3bc17
1, 176620, 176620, 512, 1024, 0xea8e04fc
1, 176621, 176621, 512, 1024, 0xea8e04fc
1, 177150, 177150, 512, 1024, 0x0b928bd8
1, 177657, 177657, 512, 1024, 0x0f02caec
1, 178164, 178164, 512, 1024, 0xe2b137a8
......
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