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Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  doc/avconv: replace forgotten av by avconv.
  avconv: reset input_ts_offset between files.
  doc/avconv: extend -ss documentation.
  avconv: call flush_encoders() from transcode() directly.
  avconv: fix broken indentation.
  avconv: rescue poor abused limit_filesize global.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-08-20 15:43:53 +02:00
parent 4b66c8b425 f5bae2c6ed
commit f138c7f993
2 changed files with 21 additions and 18 deletions
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26
avconv.c
View File

@ -213,7 +213,7 @@ static int64_t extra_size = 0;
static int nb_frames_dup = 0;
static int nb_frames_drop = 0;
static int input_sync;
static uint64_t limit_filesize = 0;
static uint64_t limit_filesize = UINT64_MAX;
static int force_fps = 0;
static char *forced_key_frames = NULL;
@ -324,6 +324,7 @@ typedef struct OutputFile {
int ost_index; /* index of the first stream in output_streams */
int64_t recording_time; /* desired length of the resulting file in microseconds */
int64_t start_time; /* start time in microseconds */
uint64_t limit_filesize;
} OutputFile;
static InputStream *input_streams = NULL;
@ -1455,7 +1456,7 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
memset(buf, fill_char, size);
}
static void flush_encoders(int ist_index, OutputStream *ost_table, int nb_ostreams)
static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
{
int i, ret;
@ -1464,7 +1465,7 @@ static void flush_encoders(int ist_index, OutputStream *ost_table, int nb_ostrea
AVCodecContext *enc = ost->st->codec;
AVFormatContext *os = output_files[ost->file_index].ctx;
if (ost->source_index != ist_index || !ost->encoding_needed)
if (!ost->encoding_needed)
continue;
if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1)
@ -1865,10 +1866,6 @@ static int output_packet(InputStream *ist, int ist_index,
}
}
discard_packet:
if (pkt == NULL) {
/* EOF handling */
flush_encoders(ist_index, ost_table, nb_ostreams);
}
return 0;
}
@ -2354,14 +2351,17 @@ static int transcode(OutputFile *output_files,
smallest output pts */
file_index = -1;
for (i = 0; i < nb_output_streams; i++) {
OutputFile *of;
int64_t ipts;
double opts;
ost = &output_streams[i];
of = &output_files[ost->file_index];
os = output_files[ost->file_index].ctx;
ist = &input_streams[ost->source_index];
if(ost->is_past_recording_time || no_packet[ist->file_index])
if (ost->is_past_recording_time || no_packet[ist->file_index] ||
(os->pb && avio_tell(os->pb) >= of->limit_filesize))
continue;
opts = ost->st->pts.val * av_q2d(ost->st->time_base);
opts = ost->st->pts.val * av_q2d(ost->st->time_base);
ipts = ist->pts;
if (!input_files[ist->file_index].eof_reached){
if(ipts < ipts_min) {
@ -2389,10 +2389,6 @@ static int transcode(OutputFile *output_files,
break;
}
/* finish if limit size exhausted */
if (limit_filesize != 0 && limit_filesize <= avio_tell(output_files[0].ctx->pb))
break;
/* read a frame from it and output it in the fifo */
is = input_files[file_index].ctx;
ret= av_read_frame(is, &pkt);
@ -2479,6 +2475,7 @@ static int transcode(OutputFile *output_files,
output_packet(ist, i, output_streams, nb_output_streams, NULL);
}
}
flush_encoders(output_streams, nb_output_streams);
term_exit();
@ -3182,6 +3179,7 @@ static int opt_input_file(const char *opt, const char *filename)
audio_channels = 0;
audio_sample_fmt = AV_SAMPLE_FMT_NONE;
av_dict_free(&ts_scale);
input_ts_offset = 0;
for (i = 0; i < orig_nb_streams; i++)
av_dict_free(&opts[i]);
@ -3668,6 +3666,7 @@ static int opt_output_file(const char *opt, const char *filename)
output_files[nb_output_files - 1].ost_index = nb_output_streams - oc->nb_streams;
output_files[nb_output_files - 1].recording_time = recording_time;
output_files[nb_output_files - 1].start_time = start_time;
output_files[nb_output_files - 1].limit_filesize = limit_filesize;
av_dict_copy(&output_files[nb_output_files - 1].opts, format_opts, 0);
/* check filename in case of an image number is expected */
@ -3794,6 +3793,7 @@ static int opt_output_file(const char *opt, const char *filename)
chapters_input_file = INT_MAX;
recording_time = INT64_MAX;
start_time = 0;
limit_filesize = UINT64_MAX;
av_freep(&meta_data_maps);
nb_meta_data_maps = 0;

13
doc/avconv.texi
View File

@ -68,7 +68,7 @@ specified for the inputs.
@chapter Stream selection
@c man begin STREAM SELECTION
By default av tries to pick the "best" stream of each type present in input
By default avconv tries to pick the "best" stream of each type present in input
files and add them to each output file. For video, this means the highest
resolution, for audio the highest channel count. For subtitle it's simply the
first subtitle stream.
@ -134,13 +134,16 @@ to the duration specified in seconds.
Set the file size limit.
@item -ss @var{position}
Seek to given time position in seconds.
@code{hh:mm:ss[.xxx]} syntax is also supported.
When used as an input option (before @code{-i}), seeks in this input file to
@var{position}. When used as an output option (before an output filename),
decodes but discards input until the timestamps reach @var{position}. This is
slower, but more accurate.
@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
@item -itsoffset @var{offset}
Set the input time offset in seconds.
@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
This option affects all the input files that follow it.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by 'offset' seconds.
@ -683,7 +686,7 @@ It disables matching streams from already created mappings.
For example, to map ALL streams from the first input file to output
@example
av -i INPUT -map 0 output
avconv -i INPUT -map 0 output
@end example
For example, if you have two audio streams in the first input file,