On Wed, 24 Apr 2013 18:50:36 +0200, Luca Barbato <[email protected]> wrote: > On 04/24/2013 05:29 PM, Anton Khirnov wrote: > > Integrate the code in the packet reading function, instead of inserting > > sleeps in many places. > > This is simpler to follow and should work better. > > --- > > avconv.c | 28 +++++++++++----------------- > > 1 file changed, 11 insertions(+), 17 deletions(-) > > > > diff --git a/avconv.c b/avconv.c > > index 658dadc..7aee0cb 100644 > > --- a/avconv.c > > +++ b/avconv.c > > @@ -1046,16 +1046,6 @@ static void do_streamcopy(InputStream *ist, > > OutputStream *ost, const AVPacket *p > > ost->st->codec->frame_number++; > > } > > > > -static void rate_emu_sleep(InputStream *ist) > > -{ > > - if (input_files[ist->file_index]->rate_emu) { > > - int64_t pts = av_rescale(ist->last_dts, 1000000, AV_TIME_BASE); > > - int64_t now = av_gettime() - ist->start; > > - if (pts > now) > > - av_usleep(pts - now); > > - } > > -} > > - > > int guess_input_channel_layout(InputStream *ist) > > { > > AVCodecContext *dec = ist->st->codec; > > @@ -1104,8 +1094,6 @@ static int decode_audio(InputStream *ist, AVPacket > > *pkt, int *got_output) > > pkt->pts = AV_NOPTS_VALUE; > > } > > > > - rate_emu_sleep(ist); > > - > > resample_changed = ist->resample_sample_fmt != > > decoded_frame->format || > > ist->resample_channels != avctx->channels > > || > > ist->resample_channel_layout != > > decoded_frame->channel_layout || > > @@ -1199,8 +1187,6 @@ static int decode_video(InputStream *ist, AVPacket > > *pkt, int *got_output) > > pre_process_video_frame(ist, (AVPicture *)decoded_frame, > > &buffer_to_free); > > #endif > > > > - rate_emu_sleep(ist); > > - > > if (ist->st->sample_aspect_ratio.num) > > decoded_frame->sample_aspect_ratio = ist->st->sample_aspect_ratio; > > > > @@ -1260,8 +1246,6 @@ static int transcode_subtitles(InputStream *ist, > > AVPacket *pkt, int *got_output) > > if (!*got_output) > > return ret; > > > > - rate_emu_sleep(ist); > > - > > for (i = 0; i < nb_output_streams; i++) { > > OutputStream *ost = output_streams[i]; > > > > @@ -1349,7 +1333,6 @@ static int output_packet(InputStream *ist, const > > AVPacket *pkt) > > > > /* handle stream copy */ > > if (!ist->decoding_needed) { > > - rate_emu_sleep(ist); > > ist->last_dts = ist->next_dts; > > switch (ist->st->codec->codec_type) { > > case AVMEDIA_TYPE_AUDIO: > > @@ -2056,6 +2039,17 @@ static int get_input_packet_mt(InputFile *f, > > AVPacket *pkt) > > > > static int get_input_packet(InputFile *f, AVPacket *pkt) > > { > > + if (f->rate_emu) { > > + int i; > > + for (i = 0; i < f->nb_streams; i++) { > > + InputStream *ist = input_streams[f->ist_index + i]; > > + int64_t pts = av_rescale(ist->last_dts, 1000000, AV_TIME_BASE); > > + int64_t now = av_gettime() - ist->start; > > + if (pts > now) > > + return AVERROR(EAGAIN); > > + } > > + } > > Should be tested, surely it would be more precise but it is busy wait vs > sleep so might cause unwanted effects. might be better get the timing > for all the streams and wait for the least amount before returning eagain. >
There should be no such problems, there is a sleep in the reading loop if there was an eagain on all inputs. But it should behave better if there are multiple inputs. And of course i tested it and it appears to work fine. -- Anton Khirnov _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
