mirror of https://git.ffmpeg.org/ffmpeg.git
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>
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commit
f138c7f993
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avconv.c
24
avconv.c
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@ -213,7 +213,7 @@ static int64_t extra_size = 0;
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static int nb_frames_dup = 0;
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static int nb_frames_drop = 0;
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static int input_sync;
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static uint64_t limit_filesize = 0;
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static uint64_t limit_filesize = UINT64_MAX;
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static int force_fps = 0;
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static char *forced_key_frames = NULL;
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@ -324,6 +324,7 @@ typedef struct OutputFile {
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int ost_index; /* index of the first stream in output_streams */
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int64_t recording_time; /* desired length of the resulting file in microseconds */
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int64_t start_time; /* start time in microseconds */
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uint64_t limit_filesize;
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} OutputFile;
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static InputStream *input_streams = NULL;
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@ -1455,7 +1456,7 @@ static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_
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memset(buf, fill_char, size);
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}
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static void flush_encoders(int ist_index, OutputStream *ost_table, int nb_ostreams)
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static void flush_encoders(OutputStream *ost_table, int nb_ostreams)
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{
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int i, ret;
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@ -1464,7 +1465,7 @@ static void flush_encoders(int ist_index, OutputStream *ost_table, int nb_ostrea
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AVCodecContext *enc = ost->st->codec;
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AVFormatContext *os = output_files[ost->file_index].ctx;
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if (ost->source_index != ist_index || !ost->encoding_needed)
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if (!ost->encoding_needed)
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continue;
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if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <=1)
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@ -1865,10 +1866,6 @@ static int output_packet(InputStream *ist, int ist_index,
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}
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}
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discard_packet:
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if (pkt == NULL) {
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/* EOF handling */
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flush_encoders(ist_index, ost_table, nb_ostreams);
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}
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return 0;
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}
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@ -2354,12 +2351,15 @@ static int transcode(OutputFile *output_files,
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smallest output pts */
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file_index = -1;
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for (i = 0; i < nb_output_streams; i++) {
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OutputFile *of;
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int64_t ipts;
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double opts;
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ost = &output_streams[i];
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of = &output_files[ost->file_index];
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os = output_files[ost->file_index].ctx;
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ist = &input_streams[ost->source_index];
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if(ost->is_past_recording_time || no_packet[ist->file_index])
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if (ost->is_past_recording_time || no_packet[ist->file_index] ||
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(os->pb && avio_tell(os->pb) >= of->limit_filesize))
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continue;
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opts = ost->st->pts.val * av_q2d(ost->st->time_base);
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ipts = ist->pts;
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@ -2389,10 +2389,6 @@ static int transcode(OutputFile *output_files,
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break;
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}
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/* finish if limit size exhausted */
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if (limit_filesize != 0 && limit_filesize <= avio_tell(output_files[0].ctx->pb))
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break;
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/* read a frame from it and output it in the fifo */
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is = input_files[file_index].ctx;
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ret= av_read_frame(is, &pkt);
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@ -2479,6 +2475,7 @@ static int transcode(OutputFile *output_files,
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output_packet(ist, i, output_streams, nb_output_streams, NULL);
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}
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}
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flush_encoders(output_streams, nb_output_streams);
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term_exit();
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@ -3182,6 +3179,7 @@ static int opt_input_file(const char *opt, const char *filename)
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audio_channels = 0;
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audio_sample_fmt = AV_SAMPLE_FMT_NONE;
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av_dict_free(&ts_scale);
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input_ts_offset = 0;
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for (i = 0; i < orig_nb_streams; i++)
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av_dict_free(&opts[i]);
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@ -3668,6 +3666,7 @@ static int opt_output_file(const char *opt, const char *filename)
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output_files[nb_output_files - 1].ost_index = nb_output_streams - oc->nb_streams;
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output_files[nb_output_files - 1].recording_time = recording_time;
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output_files[nb_output_files - 1].start_time = start_time;
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output_files[nb_output_files - 1].limit_filesize = limit_filesize;
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av_dict_copy(&output_files[nb_output_files - 1].opts, format_opts, 0);
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/* check filename in case of an image number is expected */
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@ -3794,6 +3793,7 @@ static int opt_output_file(const char *opt, const char *filename)
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chapters_input_file = INT_MAX;
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recording_time = INT64_MAX;
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start_time = 0;
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limit_filesize = UINT64_MAX;
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av_freep(&meta_data_maps);
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nb_meta_data_maps = 0;
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@ -68,7 +68,7 @@ specified for the inputs.
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@chapter Stream selection
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@c man begin STREAM SELECTION
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By default av tries to pick the "best" stream of each type present in input
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By default avconv tries to pick the "best" stream of each type present in input
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files and add them to each output file. For video, this means the highest
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resolution, for audio the highest channel count. For subtitle it's simply the
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first subtitle stream.
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@ -134,13 +134,16 @@ to the duration specified in seconds.
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Set the file size limit.
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@item -ss @var{position}
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Seek to given time position in seconds.
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@code{hh:mm:ss[.xxx]} syntax is also supported.
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When used as an input option (before @code{-i}), seeks in this input file to
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@var{position}. When used as an output option (before an output filename),
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decodes but discards input until the timestamps reach @var{position}. This is
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slower, but more accurate.
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@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
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@item -itsoffset @var{offset}
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Set the input time offset in seconds.
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@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
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This option affects all the input files that follow it.
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The offset is added to the timestamps of the input files.
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Specifying a positive offset means that the corresponding
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streams are delayed by 'offset' seconds.
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@ -683,7 +686,7 @@ It disables matching streams from already created mappings.
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For example, to map ALL streams from the first input file to output
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@example
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av -i INPUT -map 0 output
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avconv -i INPUT -map 0 output
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@end example
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For example, if you have two audio streams in the first input file,
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