[FFmpeg-devel] [PATCH v2] examples: add flac_test
wm4
nfxjfg at googlemail.com
Tue Apr 14 21:58:33 CEST 2015
On Tue, 14 Apr 2015 04:03:40 +0300
Ludmila Glinskih <lglinskih at gmail.com> wrote:
> This is a simple test for the FLAC codec.
> It generates an increasing tone, encodes it, decodes it back and
> compares with the original one byte-by-byte.
> ---
> configure | 2 +
> doc/Makefile | 1 +
> doc/examples/Makefile | 1 +
> doc/examples/flac_test.c | 295 +++++++++++++++++++++++++++++++++++++++++++++++
> 4 files changed, 299 insertions(+)
> create mode 100644 doc/examples/flac_test.c
>
> diff --git a/configure b/configure
> index bc59271..5650ef8 100755
> --- a/configure
> +++ b/configure
> @@ -1329,6 +1329,7 @@ EXAMPLE_LIST="
> filter_audio_example
> filtering_audio_example
> filtering_video_example
> + flac_test_example
> metadata_example
> muxing_example
> qsvdec_example
> @@ -2679,6 +2680,7 @@ extract_mvs_example_deps="avcodec avformat avutil"
> filter_audio_example_deps="avfilter avutil"
> filtering_audio_example_deps="avfilter avcodec avformat avutil"
> filtering_video_example_deps="avfilter avcodec avformat avutil"
> +flac_test_example_deps="avcodec avutil"
> metadata_example_deps="avformat avutil"
> muxing_example_deps="avcodec avformat avutil swscale"
> qsvdec_example_deps="avcodec avutil libmfx h264_qsv_decoder vaapi_x11"
> diff --git a/doc/Makefile b/doc/Makefile
> index 4573531..f462acc 100644
> --- a/doc/Makefile
> +++ b/doc/Makefile
> @@ -45,6 +45,7 @@ DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
> DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
> DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
> DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
> +DOC_EXAMPLES-$(CONFIG_FLAC_TEST_EXAMPLE) += flac_test
> DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
> DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
> DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
> diff --git a/doc/examples/Makefile b/doc/examples/Makefile
> index 9699f11..72a2fb6 100644
> --- a/doc/examples/Makefile
> +++ b/doc/examples/Makefile
> @@ -18,6 +18,7 @@ EXAMPLES= avio_list_dir \
> extract_mvs \
> filtering_video \
> filtering_audio \
> + flac_test \
> metadata \
> muxing \
> remuxing \
> diff --git a/doc/examples/flac_test.c b/doc/examples/flac_test.c
> new file mode 100644
> index 0000000..392c50c
> --- /dev/null
> +++ b/doc/examples/flac_test.c
> @@ -0,0 +1,295 @@
> +/*
> + * Copyright (c) 2015 Ludmila Glinskih
> + * Copyright (c) 2001 Fabrice Bellard
> + *
> + * 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.
> + */
> +
> +/*
> + * FLAC codec test.
> + * Encodes raw data to FLAC format and decodes it back to raw. Compares raw-data
> + * after that.
> + */
> +
> +#include <libavcodec/avcodec.h>
> +#include <libavutil/common.h>
> +#include <libavutil/samplefmt.h>
> +
> +#define NUMBER_OF_FRAMES 200
> +#define NAME_BUFF_SIZE 100
> +
> +/* generate i-th frame of test audio */
> +static int generate_raw_frame(uint16_t *frame_data, int i, int sample_rate,
> + int channels, int frame_size)
> +{
> + double t, tincr, tincr2;
> + int j, k;
> +
> + t = 0.0;
> + tincr = 2 * M_PI * 440.0 / sample_rate;
> + tincr2 = tincr / sample_rate;
> + for (j = 0; j < frame_size; j++)
> + {
> + frame_data[channels * j] = (int)(sin(t) * 10000);
> + for (k = 1; k < channels; k++)
> + frame_data[channels * j + k] = frame_data[channels * j] * 2;
> + t = i * tincr + (i * (i + 1) / 2.0 * tincr2);
> + }
> + return 0;
> +}
This was mentioned before: using floating point in tests causes
problems which can be avoided by using integers only. This includes the
sin() function. (Maybe generate some sort of square wave instead? Or
just silence? I don't know.)
> +static int init_encoder(AVCodec *enc, AVCodecContext **enc_ctx,
> + int64_t ch_layout, int sample_rate)
> +{
> + AVCodecContext *ctx;
> + int result;
> + char name_buff[NAME_BUFF_SIZE];
> +
> + av_get_channel_layout_string(name_buff, NAME_BUFF_SIZE, 0, ch_layout);
> + av_log(NULL, AV_LOG_INFO, "channel layout: %s, sample rate: %i\n", name_buff, sample_rate);
> +
> + ctx = avcodec_alloc_context3(enc);
> + if (!ctx)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate encoder context\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + ctx->sample_fmt = AV_SAMPLE_FMT_S16;
> + ctx->sample_rate = sample_rate;
> + ctx->channel_layout = ch_layout;
> +
> + result = avcodec_open2(ctx, enc, NULL);
> + if (result < 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't open encoder\n");
> + return AVERROR_UNKNOWN;
In this particular case, it would probably make sense to forward the
error code. (Also affects some other lines in the patch.) I don't know
how important this is for API tests, or what exactly we want, though.
> + }
> +
> + *enc_ctx = ctx;
> + return 0;
> +}
> +
> +static int init_decoder(AVCodec *dec, AVCodecContext **dec_ctx,
> + int64_t ch_layout)
> +{
> + AVCodecContext *ctx;
> + int result;
> +
> + ctx = avcodec_alloc_context3(dec);
> + if (!ctx)
> + {
> + av_log(NULL, AV_LOG_ERROR , "Can't allocate decoder context\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + ctx->request_sample_fmt = AV_SAMPLE_FMT_S16;
> + /* XXX: FLAC ignores it for some reason */
> + ctx->request_channel_layout = ch_layout;
> + ctx->channel_layout = ch_layout;
Only some decoders can change the output, and then only in some cases.
Normally, the API user is supposed to use libraries like libswresample
to convert data to the required format. These fields (including
request_sample_fmt) merely expose additional decoder features. They
don't have to be present.
> +
> + result = avcodec_open2(ctx, dec, NULL);
> + if (result < 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't open decoder\n");
> + return AVERROR_UNKNOWN;
> + }
> +
> + *dec_ctx = ctx;
> + return 0;
> +}
> +
> +static int run_test(AVCodec *enc, AVCodec *dec, AVCodecContext *enc_ctx,
> + AVCodecContext *dec_ctx)
> +{
> + AVPacket enc_pkt;
> + AVFrame *in_frame, *out_frame;
> + uint8_t *raw_in = NULL, *raw_out = NULL;
> + int in_offset = 0, out_offset = 0;
> + int frame_data_size = 0;
> + int result = 0;
> + int got_output = 0;
> + int i = 0;
> +
> + in_frame = av_frame_alloc();
> + if (!in_frame)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate input frame\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + in_frame->nb_samples = enc_ctx->frame_size;
> + in_frame->format = enc_ctx->sample_fmt;
> + in_frame->channel_layout = enc_ctx->channel_layout;
> + if (av_frame_get_buffer(in_frame, 32) != 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate a buffer for input frame\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + out_frame = av_frame_alloc();
> + if (!out_frame)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate output frame\n");
> + return AVERROR(ENOMEM);
This leaks in_frame on error. But it might be ok in such a test. We
have to decide whether it is. (I'd say it's ok.)
> + }
> +
> + raw_in = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
> + if (!raw_in)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_in\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + raw_out = av_malloc(in_frame->linesize[0] * NUMBER_OF_FRAMES);
> + if (!raw_out)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for raw_out\n");
> + return AVERROR(ENOMEM);
> + }
> +
> + for (i = 0; i < NUMBER_OF_FRAMES; i++)
> + {
> + av_init_packet(&enc_pkt);
> + enc_pkt.data = NULL;
> + enc_pkt.size = 0;
> +
> + generate_raw_frame((uint16_t*)(in_frame->data[0]), i, enc_ctx->sample_rate,
> + enc_ctx->channels, enc_ctx->frame_size);
> + memcpy(raw_in + in_offset, in_frame->data[0], in_frame->linesize[0]);
> + in_offset += in_frame->linesize[0];
> + result = avcodec_encode_audio2(enc_ctx, &enc_pkt, in_frame, &got_output);
> + if (result < 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Error encoding audio frame\n");
> + return AVERROR_UNKNOWN;
> + }
> +
> + /* if we get an encoded packet, feed it straight to the decoder */
> + if (got_output)
> + {
> + result = avcodec_decode_audio4(dec_ctx, out_frame, &got_output, &enc_pkt);
> + if (result < 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Error decoding audio packet\n");
> + return AVERROR_UNKNOWN;
> + }
> +
> + if (got_output)
> + {
> + if (result != enc_pkt.size)
> + {
> + av_log(NULL, AV_LOG_INFO, "Decoder consumed only part of a packet, it is allowed to do so -- need to update this test\n");
The message probably lacks an "if" ("if it is allowed").
> + return AVERROR_UNKNOWN;
> + }
> +
> + if (in_frame->nb_samples != out_frame->nb_samples)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different number of samples\n");
> + return AVERROR_UNKNOWN;
> + }
> +
> + if (in_frame->channel_layout != out_frame->channel_layout)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different channel layout\n");
> + return AVERROR_UNKNOWN;
> + }
> +
> + if (in_frame->format != out_frame->format)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Error frames before and after decoding has different sample format\n");
> + return AVERROR_UNKNOWN;
> + }
> + memcpy(raw_out + out_offset, out_frame->data[0], out_frame->linesize[0]);
> + out_offset += out_frame->linesize[0];
> + }
> + }
> + av_free_packet(&enc_pkt);
> + }
> +
> + if (memcmp(raw_in, raw_out, frame_data_size * NUMBER_OF_FRAMES) != 0)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Output differs\n");
> + return 1;
> + }
> +
> + av_log(NULL, AV_LOG_INFO, "OK\n");
> +
> + av_free(raw_in);
> + av_free(raw_out);
> + av_frame_free(&in_frame);
> + av_frame_free(&out_frame);
> + return 0;
> +}
> +
> +static int close_encoder(AVCodecContext *enc_ctx)
> +{
> + avcodec_close(enc_ctx);
> + av_free(enc_ctx);
> + return 0;
> +}
> +
> +static int close_decoder(AVCodecContext *dec_ctx)
> +{
> + avcodec_close(dec_ctx);
> + av_free(dec_ctx);
> + return 0;
> +}
> +
> +int main(void)
> +{
> + AVCodec *enc = NULL, *dec = NULL;
> + AVCodecContext *enc_ctx = NULL, *dec_ctx = NULL;
> + uint64_t channel_layouts[] = {AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1_BACK, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_STEREO_DOWNMIX};
> + int sample_rates[] = {8000, 44100, 48000, 192000};
> + int cl, sr;
> +
> + avcodec_register_all();
> +
> + enc = avcodec_find_encoder(AV_CODEC_ID_FLAC);
> + if (!enc)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't find encoder\n");
> + return 1;
> + }
> +
> + dec = avcodec_find_decoder(AV_CODEC_ID_FLAC);
> + if (!dec)
> + {
> + av_log(NULL, AV_LOG_ERROR, "Can't find decoder\n");
> + return 1;
> + }
> +
> + for (cl = 0; cl < FF_ARRAY_ELEMS(channel_layouts); cl++)
> + {
> + for (sr = 0; sr < FF_ARRAY_ELEMS(sample_rates); sr++)
> + {
> + if (init_encoder(enc, &enc_ctx, channel_layouts[cl], sample_rates[sr]) != 0)
> + return 1;
> + if (init_decoder(dec, &dec_ctx, channel_layouts[cl]) != 0)
> + return 1;
> + if (run_test(enc, dec, enc_ctx, dec_ctx) != 0)
> + return 1;
> + close_encoder(enc_ctx);
> + close_decoder(dec_ctx);
> + }
> + }
> +
> + return 0;
> +}
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