[FFmpeg-devel] [PATCH] api-h264-test: rename and expand

[Michael Niedermayer]

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
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