Hi, Thank you for the comment! I'm not sure if I fixed it right =/
Kind regards, Ludmila Glinskih ср, 26 авг. 2015 г. в 3:52, Michael Niedermayer <mich...@niedermayer.cc>: > On Tue, Aug 25, 2015 at 11:00:40PM +0300, Ludmila Glinskih wrote: > > Add support of floating point decoders. Add support of audio decoders. > > --- > > tests/api/Makefile | 2 +- > > tests/api/api-decode-test.c | 355 > +++++++++++++++++++++++++++++++++++++++++ > > tests/api/api-h264-test.c | 166 ------------------- > > tests/fate/api.mak | 12 +- > > tests/ref/fate/api-decode-h264 | 18 +++ > > tests/ref/fate/api-h264 | 18 --- > > 6 files changed, 383 insertions(+), 188 deletions(-) > > create mode 100644 tests/api/api-decode-test.c > > delete mode 100644 tests/api/api-h264-test.c > > create mode 100644 tests/ref/fate/api-decode-h264 > > delete mode 100644 tests/ref/fate/api-h264 > > > > diff --git a/tests/api/Makefile b/tests/api/Makefile > > index 27f499f..57a7422 100644 > > --- a/tests/api/Makefile > > +++ b/tests/api/Makefile > > @@ -1,5 +1,5 @@ > > APITESTPROGS-$(call ENCDEC, FLAC, FLAC) += api-flac > > -APITESTPROGS-$(call DEMDEC, H264, H264) += api-h264 > > +APITESTPROGS-yes += api-decode > > APITESTPROGS-yes += api-seek > > APITESTPROGS-$(call DEMDEC, H263, H263) += api-band > > APITESTPROGS += $(APITESTPROGS-yes) > > diff --git a/tests/api/api-decode-test.c b/tests/api/api-decode-test.c > > new file mode 100644 > > index 0000000..29c7dd7 > > --- /dev/null > > +++ b/tests/api/api-decode-test.c > > @@ -0,0 +1,355 @@ > > +/* > > + * Copyright (c) 2015 Ludmila Glinskih > > + * > > + * Permission is hereby granted, free of charge, to any person > obtaining a copy > > + * of this software and associated documentation files (the > "Software"), to deal > > + * in the Software without restriction, including without limitation > the rights > > + * to use, copy, modify, merge, publish, distribute, sublicense, and/or > sell > > + * copies of the Software, and to permit persons to whom the Software is > > + * furnished to do so, subject to the following conditions: > > + * > > + * The above copyright notice and this permission notice shall be > included in > > + * all copies or substantial portions of the Software. > > + * > > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, > EXPRESS OR > > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF > MERCHANTABILITY, > > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT > SHALL > > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR > OTHER > > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, > ARISING FROM, > > + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER > DEALINGS IN > > + * THE SOFTWARE. > > + */ > > + > > +/** > > + * Decode test. > > + */ > > + > > +#include "libavutil/adler32.h" > > +#include "libavcodec/avcodec.h" > > +#include "libavformat/avformat.h" > > +#include "libavutil/imgutils.h" > > +#include "libswresample/swresample.h" > > +#include "libavutil/opt.h" > > + > > +static int resample_and_print_data(AVCodecContext *ctx, AVFrame *fr, > int sample_fmt) > > +{ > > + struct SwrContext *swr_ctx; > > + int dst_nb_samples; > > + int dst_bufsize; > > + int dst_linesize = 0; > > + uint8_t **dst_data = NULL; > > + int result; > > + > > + swr_ctx = swr_alloc_set_opts(NULL, > > + fr->channel_layout, > > + sample_fmt, > > + fr->sample_rate, > > + fr->channel_layout, > > + ctx->sample_fmt, > > + fr->sample_rate, > > + 0, NULL); > > + if (!swr_ctx) { > > + av_log(NULL, AV_LOG_ERROR, "Could not allocate resampler > context\n"); > > + return -1; > > + } > > + result = swr_init(swr_ctx); > > + if (result < 0) { > > + av_log(NULL, AV_LOG_ERROR, "Can't initialize the resampling > context\n"); > > + return result; > > + } > > + dst_nb_samples = fr->nb_samples; > > + result = av_samples_alloc_array_and_samples(&dst_data, > &dst_linesize, fr->channels, > > + dst_nb_samples, > sample_fmt, 0); > > + if (result < 0) { > > + av_log(NULL, AV_LOG_ERROR, "Can't allocate buffer for samples > after resampling\n"); > > + return result; > > + } > > + > > > + result = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const > uint8_t **)fr->data, fr->nb_samples); > > + if (result < 0) { > > + av_log(NULL, AV_LOG_ERROR, "Error while resampling\n"); > > + return result; > > + } > > + > > + dst_bufsize = av_samples_get_buffer_size(&dst_linesize, > fr->channels, result, sample_fmt, 1); > > + if (dst_bufsize < 0) { > > + av_log(NULL, AV_LOG_ERROR, "Can'get buffer size after > resampling\n"); > > + return dst_bufsize; > > + } > > + > > + fwrite(dst_data[0], 1, dst_bufsize, stdout); > > this would mismatch on big endian > > > [...] > -- > Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB > > Frequently ignored answer#1 FFmpeg bugs should be sent to our bugtracker. > User > questions about the command line tools should be sent to the ffmpeg-user > ML. > And questions about how to use libav* should be sent to the libav-user ML. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel