FFmpeg/libavfilter/af_ashowinfo.c
Michael Niedermayer 6b783d2ace Merge commit '7046bd9bc9ce04521edf453c9b603d84d69c7512'
* commit '7046bd9bc9ce04521edf453c9b603d84d69c7512':
  lavfi: Move avcodec header to the only filter needing it

Conflicts:
	libavfilter/avfilter.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2015-05-20 19:14:11 +02:00

264 lines
9.3 KiB
C

/*
* Copyright (c) 2011 Stefano Sabatini
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* filter for showing textual audio frame information
*/
#include <inttypes.h>
#include <stddef.h>
#include "libavutil/adler32.h"
#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/downmix_info.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem.h"
#include "libavutil/replaygain.h"
#include "libavutil/timestamp.h"
#include "libavutil/samplefmt.h"
#include "libavcodec/avcodec.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct AShowInfoContext {
/**
* Scratch space for individual plane checksums for planar audio
*/
uint32_t *plane_checksums;
} AShowInfoContext;
static av_cold void uninit(AVFilterContext *ctx)
{
AShowInfoContext *s = ctx->priv;
av_freep(&s->plane_checksums);
}
static void dump_matrixenc(AVFilterContext *ctx, AVFrameSideData *sd)
{
enum AVMatrixEncoding enc;
av_log(ctx, AV_LOG_INFO, "matrix encoding: ");
if (sd->size < sizeof(enum AVMatrixEncoding)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
enc = *(enum AVMatrixEncoding *)sd->data;
switch (enc) {
case AV_MATRIX_ENCODING_NONE: av_log(ctx, AV_LOG_INFO, "none"); break;
case AV_MATRIX_ENCODING_DOLBY: av_log(ctx, AV_LOG_INFO, "Dolby Surround"); break;
case AV_MATRIX_ENCODING_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
case AV_MATRIX_ENCODING_DPLIIX: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIx"); break;
case AV_MATRIX_ENCODING_DPLIIZ: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic IIz"); break;
case AV_MATRIX_ENCODING_DOLBYEX: av_log(ctx, AV_LOG_INFO, "Dolby EX"); break;
case AV_MATRIX_ENCODING_DOLBYHEADPHONE: av_log(ctx, AV_LOG_INFO, "Dolby Headphone"); break;
default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
}
}
static void dump_downmix(AVFilterContext *ctx, AVFrameSideData *sd)
{
AVDownmixInfo *di;
av_log(ctx, AV_LOG_INFO, "downmix: ");
if (sd->size < sizeof(*di)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
di = (AVDownmixInfo *)sd->data;
av_log(ctx, AV_LOG_INFO, "preferred downmix type - ");
switch (di->preferred_downmix_type) {
case AV_DOWNMIX_TYPE_LORO: av_log(ctx, AV_LOG_INFO, "Lo/Ro"); break;
case AV_DOWNMIX_TYPE_LTRT: av_log(ctx, AV_LOG_INFO, "Lt/Rt"); break;
case AV_DOWNMIX_TYPE_DPLII: av_log(ctx, AV_LOG_INFO, "Dolby Pro Logic II"); break;
default: av_log(ctx, AV_LOG_WARNING, "unknown"); break;
}
av_log(ctx, AV_LOG_INFO, " Mix levels: center %f (%f ltrt) - "
"surround %f (%f ltrt) - lfe %f",
di->center_mix_level, di->center_mix_level_ltrt,
di->surround_mix_level, di->surround_mix_level_ltrt,
di->lfe_mix_level);
}
static void print_gain(AVFilterContext *ctx, const char *str, int32_t gain)
{
av_log(ctx, AV_LOG_INFO, "%s - ", str);
if (gain == INT32_MIN)
av_log(ctx, AV_LOG_INFO, "unknown");
else
av_log(ctx, AV_LOG_INFO, "%f", gain / 100000.0f);
av_log(ctx, AV_LOG_INFO, ", ");
}
static void print_peak(AVFilterContext *ctx, const char *str, uint32_t peak)
{
av_log(ctx, AV_LOG_INFO, "%s - ", str);
if (!peak)
av_log(ctx, AV_LOG_INFO, "unknown");
else
av_log(ctx, AV_LOG_INFO, "%f", (float)peak / UINT32_MAX);
av_log(ctx, AV_LOG_INFO, ", ");
}
static void dump_replaygain(AVFilterContext *ctx, AVFrameSideData *sd)
{
AVReplayGain *rg;
av_log(ctx, AV_LOG_INFO, "replaygain: ");
if (sd->size < sizeof(*rg)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
rg = (AVReplayGain*)sd->data;
print_gain(ctx, "track gain", rg->track_gain);
print_peak(ctx, "track peak", rg->track_peak);
print_gain(ctx, "album gain", rg->album_gain);
print_peak(ctx, "album peak", rg->album_peak);
}
static void dump_audio_service_type(AVFilterContext *ctx, AVFrameSideData *sd)
{
enum AVAudioServiceType *ast;
av_log(ctx, AV_LOG_INFO, "audio service type: ");
if (sd->size < sizeof(*ast)) {
av_log(ctx, AV_LOG_INFO, "invalid data");
return;
}
ast = (enum AVAudioServiceType*)sd->data;
switch (*ast) {
case AV_AUDIO_SERVICE_TYPE_MAIN: av_log(ctx, AV_LOG_INFO, "Main Audio Service"); break;
case AV_AUDIO_SERVICE_TYPE_EFFECTS: av_log(ctx, AV_LOG_INFO, "Effects"); break;
case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Visually Impaired"); break;
case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED: av_log(ctx, AV_LOG_INFO, "Hearing Impaired"); break;
case AV_AUDIO_SERVICE_TYPE_DIALOGUE: av_log(ctx, AV_LOG_INFO, "Dialogue"); break;
case AV_AUDIO_SERVICE_TYPE_COMMENTARY: av_log(ctx, AV_LOG_INFO, "Commentary"); break;
case AV_AUDIO_SERVICE_TYPE_EMERGENCY: av_log(ctx, AV_LOG_INFO, "Emergency"); break;
case AV_AUDIO_SERVICE_TYPE_VOICE_OVER: av_log(ctx, AV_LOG_INFO, "Voice Over"); break;
case AV_AUDIO_SERVICE_TYPE_KARAOKE: av_log(ctx, AV_LOG_INFO, "Karaoke"); break;
default: av_log(ctx, AV_LOG_INFO, "unknown"); break;
}
}
static void dump_unknown(AVFilterContext *ctx, AVFrameSideData *sd)
{
av_log(ctx, AV_LOG_INFO, "unknown side data type: %d, size %d bytes", sd->type, sd->size);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
AShowInfoContext *s = ctx->priv;
char chlayout_str[128];
uint32_t checksum = 0;
int channels = inlink->channels;
int planar = av_sample_fmt_is_planar(buf->format);
int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
int data_size = buf->nb_samples * block_align;
int planes = planar ? channels : 1;
int i;
void *tmp_ptr = av_realloc_array(s->plane_checksums, channels, sizeof(*s->plane_checksums));
if (!tmp_ptr)
return AVERROR(ENOMEM);
s->plane_checksums = tmp_ptr;
for (i = 0; i < planes; i++) {
uint8_t *data = buf->extended_data[i];
s->plane_checksums[i] = av_adler32_update(0, data, data_size);
checksum = i ? av_adler32_update(checksum, data, data_size) :
s->plane_checksums[0];
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
buf->channel_layout);
av_log(ctx, AV_LOG_INFO,
"n:%"PRId64" pts:%s pts_time:%s pos:%"PRId64" "
"fmt:%s channels:%d chlayout:%s rate:%d nb_samples:%d "
"checksum:%08"PRIX32" ",
inlink->frame_count,
av_ts2str(buf->pts), av_ts2timestr(buf->pts, &inlink->time_base),
av_frame_get_pkt_pos(buf),
av_get_sample_fmt_name(buf->format), av_frame_get_channels(buf), chlayout_str,
buf->sample_rate, buf->nb_samples,
checksum);
av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
for (i = 0; i < planes; i++)
av_log(ctx, AV_LOG_INFO, "%08"PRIX32" ", s->plane_checksums[i]);
av_log(ctx, AV_LOG_INFO, "]\n");
for (i = 0; i < buf->nb_side_data; i++) {
AVFrameSideData *sd = buf->side_data[i];
av_log(ctx, AV_LOG_INFO, " side data - ");
switch (sd->type) {
case AV_FRAME_DATA_MATRIXENCODING: dump_matrixenc (ctx, sd); break;
case AV_FRAME_DATA_DOWNMIX_INFO: dump_downmix (ctx, sd); break;
case AV_FRAME_DATA_REPLAYGAIN: dump_replaygain(ctx, sd); break;
case AV_FRAME_DATA_AUDIO_SERVICE_TYPE: dump_audio_service_type(ctx, sd); break;
default: dump_unknown (ctx, sd); break;
}
av_log(ctx, AV_LOG_INFO, "\n");
}
return ff_filter_frame(inlink->dst->outputs[0], buf);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_ashowinfo = {
.name = "ashowinfo",
.description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
.priv_size = sizeof(AShowInfoContext),
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
};