[FFmpeg-devel] [PATCH 2/2] doc/examples: add audio decoding/filtering example.
Clément Bœsch
ubitux at gmail.com
Tue Feb 21 09:15:15 CET 2012
On Mon, Feb 20, 2012 at 06:45:27PM +0100, Nicolas George wrote:
> Le duodi 2 ventôse, an CCXX, Clément Bœsch a écrit :
> > doc/examples/Makefile | 2 +-
> > doc/examples/filtering-audio.c | 244 ++++++++++++++++++++++++++++++++++++++++
> > 2 files changed, 245 insertions(+), 1 deletions(-)
> > create mode 100644 doc/examples/filtering-audio.c
>
> Good idea.
>
> >
> > diff --git a/doc/examples/Makefile b/doc/examples/Makefile
> > index b4d299f..135fa95 100644
> > --- a/doc/examples/Makefile
> > +++ b/doc/examples/Makefile
> > @@ -3,7 +3,7 @@ FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
> > CFLAGS+=-Wall $(shell pkg-config --cflags $(FFMPEG_LIBS))
> > LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
> >
> > -EXAMPLES=decoding_encoding filtering metadata muxing
> > +EXAMPLES=decoding_encoding filtering filtering-audio metadata muxing
>
> Aren't we being inconsistent with dashes and underscores?
>
Indeed, renamed.
> >
> > OBJS=$(addsuffix .o,$(EXAMPLES))
> >
> > diff --git a/doc/examples/filtering-audio.c b/doc/examples/filtering-audio.c
> > new file mode 100644
> > index 0000000..738b186
> > --- /dev/null
> > +++ b/doc/examples/filtering-audio.c
> > @@ -0,0 +1,244 @@
> > +/*
> > + * Copyright (c) 2010 Nicolas George
> > + * Copyright (c) 2011 Stefano Sabatini
> > + * Copyright (c) 2012 Clément Bœsch
> > + *
> > + * 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.
> > + */
> > +
> > +/**
> > + * @file
> > + * API example for audio decoding and filtering
> > + */
> > +
> > +#include <unistd.h>
> > +
> > +#include <libavcodec/avcodec.h>
> > +#include <libavformat/avformat.h>
> > +#include <libavfilter/asrc_abuffer.h>
> > +#include <libavfilter/avfiltergraph.h>
> > +#include <libavfilter/avcodec.h>
> > +#include <libavfilter/buffersink.h>
> > +
> > +const char *filter_descr = "aresample=8000,aconvert=s16:mono";
> > +const char *player = "ffplay -f s16le -ar 8000 -ac 1 -i /dev/stdin";
>
> For ffplay, I believe "-" works too, and may be more portable. And the -i is
> not necessary.
>
That's right, simplified.
> > +
> > +static AVFormatContext *fmt_ctx;
> > +static AVCodecContext *dec_ctx;
> > +AVFilterContext *buffersink_ctx;
> > +AVFilterContext *buffersrc_ctx;
> > +AVFilterGraph *filter_graph;
> > +static int audio_stream_index = -1;
> > +
> > +static int open_input_file(const char *filename)
> > +{
> > + int ret;
> > + AVCodec *dec;
> > +
> > + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
> > + return ret;
> > + }
> > +
> > + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
> > + return ret;
> > + }
> > +
> > + /* select the audio stream */
> > + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
> > + if (ret < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
> > + return ret;
> > + }
> > + audio_stream_index = ret;
> > + dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
> > +
> > + /* init the audio decoder */
> > + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
> > + return ret;
> > + }
> > +
> > + return 0;
> > +}
> > +
> > +static int init_filters(const char *filters_descr)
> > +{
> > + char args[512];
> > + int ret;
> > + AVFilter *buffersrc = avfilter_get_by_name("abuffer");
> > + AVFilter *buffersink = avfilter_get_by_name("abuffersink");
> > + AVFilterInOut *outputs = avfilter_inout_alloc();
> > + AVFilterInOut *inputs = avfilter_inout_alloc();
> > + const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
> > + const int packing_fmts[] = { AVFILTER_PACKED, -1 };
> > + const int64_t *chlayouts = avfilter_all_channel_layouts;
> > + AVABufferSinkParams *abuffersink_params;
>
> Some of the variables use the exact name "abuffer...", some use only
> "buffer", this is slightly inconsistent.
>
buffersrc and buffsersink respectively renamed into abuffersrc and abuffersink.
> > + const AVFilterLink *outlink;
> > +
> > + filter_graph = avfilter_graph_alloc();
> > +
> > + /* buffer audio source: the decoded frames from the decoder will be inserted here. */
> > + if (!dec_ctx->channel_layout)
> > + dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
> > + snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64":packed",
> > + dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
> > + ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
> > + args, NULL, filter_graph);
> > + if (ret < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
> > + return ret;
> > + }
> > +
> > + /* buffer audio sink: to terminate the filter chain. */
> > + abuffersink_params = av_abuffersink_params_alloc();
> > + abuffersink_params->sample_fmts = sample_fmts;
> > + abuffersink_params->channel_layouts = chlayouts;
> > + abuffersink_params->packing_fmts = packing_fmts;
> > + ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
> > + NULL, abuffersink_params, filter_graph);
>
> As a matter of curiosity, is there a reason to use a string argument for
> abuffersrc and a structured parameter for abuffersink?
>
None that I know of, except that abuffersink
(lavfi/sink_buffer.c:asink_init) only accepts a struct parameter while
abuffer (lavfi/asrc_abuffer.c:init) only accepts a string argument.
Maybe the two filters could accept two kind of inputs for consistency.
> > + if (ret < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
> > + return ret;
> > + }
> > +
> > + /* Endpoints for the filter graph. */
> > + outputs->name = av_strdup("in");
> > + outputs->filter_ctx = buffersrc_ctx;
> > + outputs->pad_idx = 0;
> > + outputs->next = NULL;
> > +
> > + inputs->name = av_strdup("out");
> > + inputs->filter_ctx = buffersink_ctx;
> > + inputs->pad_idx = 0;
> > + inputs->next = NULL;
> > +
> > + if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
> > + &inputs, &outputs, NULL)) < 0)
> > + return ret;
> > +
> > + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
> > + return ret;
> > +
> > + outlink = buffersink_ctx->inputs[0];
> > + // abuse args buffer to store channel layout string
>
> "reuse", maybe? And while I am nitpicking, the comment style is
> inconsistent.
>
Replaced "abuse" with "reuse". About the comment style, it was relative to
a single line and not an explanation of the following block; since the
line was already too long, I just put it above. The comment is reworked,
hopefully in a better way.
> > + av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
> > + av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
> > + (int)outlink->sample_rate,
> > + (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
> > + args);
> > +
> > + return 0;
> > +}
> > +
> > +static void print_samplesref(AVFilterBufferRef *samplesref)
> > +{
> > + const AVFilterBufferRefAudioProps *props = samplesref->audio;
> > + const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
> > + const uint16_t *p = (uint16_t*)samplesref->data[0];
> > + const uint16_t *p_end = p + n;
> > +
> > + while (p < p_end) {
> > + fputc(*p & 0xff, stdout);
> > + fputc(*p>>8 & 0xff, stdout);
> > + p++;
> > + }
> > + fflush(stdout);
> > +}
> > +
> > +int main(int argc, char **argv)
> > +{
> > + int ret;
> > + AVPacket packet;
> > + AVFrame frame;
> > + int got_frame;
> > +
> > + if (argc != 2) {
> > + fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
> > + exit(1);
> > + }
> > +
> > + avcodec_register_all();
> > + av_register_all();
> > + avfilter_register_all();
> > +
> > + if ((ret = open_input_file(argv[1])) < 0)
> > + goto end;
> > + if ((ret = init_filters(filter_descr)) < 0)
> > + goto end;
> > +
> > + /* read all packets */
> > + while (1) {
> > + AVFilterBufferRef *samplesref;
> > + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
> > + break;
> > +
> > + if (packet.stream_index == audio_stream_index) {
> > + avcodec_get_frame_defaults(&frame);
> > + got_frame = 0;
> > + ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
> > + av_free_packet(&packet);
> > + if (ret < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
> > + break;
> > + }
> > +
> > + if (got_frame) {
> > + const int bps = av_get_bytes_per_sample(dec_ctx->sample_fmt);
> > + const int decoded_data_size = frame.nb_samples * dec_ctx->channels * bps;
> > +
> > + /* push the audio data from decoded frame into the filtergraph */
> > + if (av_asrc_buffer_add_buffer(buffersrc_ctx,
> > + frame.data[0],
> > + decoded_data_size,
> > + dec_ctx->sample_rate,
> > + dec_ctx->sample_fmt,
> > + dec_ctx->channel_layout,
> > + 0, frame.pts, 0) < 0) {
> > + av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
> > + exit(1);
>
> Is there a reason to use exit here while the other places use break or goto?
>
Not at all, my eyes just catch a few exit() and I didn't though much about
this; obviously, a break is more appropriate here, so replaced.
> > + }
> > +
> > + /* pull filtered pictures from the filtergraph */
> > + while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
> > + av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
> > + if (samplesref) {
> > + print_samplesref(samplesref);
> > + avfilter_unref_buffer(samplesref);
> > + }
> > + }
> > + }
> > + }
> > + }
> > +end:
> > + avfilter_graph_free(&filter_graph);
> > + if (dec_ctx)
> > + avcodec_close(dec_ctx);
> > + avformat_close_input(&fmt_ctx);
> > +
> > + if (ret < 0 && ret != AVERROR_EOF) {
> > + char buf[1024];
> > + av_strerror(ret, buf, sizeof(buf));
> > + fprintf(stderr, "Error occurred: %s\n", buf);
> > + exit(1);
> > + }
> > +
> > + exit(0);
> > +}
>
> Apart from these little details, it seems really useful.
>
Great, then I'll push the attached version soon.
--
Clément B.
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From 6d6f41aafcfb855bf89977ad055fa524235c732e Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Cl=C3=A9ment=20B=C5=93sch?= <clement.boesch at smartjog.com>
Date: Mon, 20 Feb 2012 13:49:18 +0100
Subject: [PATCH 2/2] doc/examples: add audio decoding/filtering example.
Mostly based on doc/examples/filtering.c. lavfi API is still limited to
"buffer feeding" instead of "frame feeding" at the moment, so this
example code sticks with it.
---
doc/examples/Makefile | 2 +-
doc/examples/filtering_audio.c | 245 ++++++++++++++++++++++++++++++++++++++++
2 files changed, 246 insertions(+), 1 deletions(-)
create mode 100644 doc/examples/filtering_audio.c
diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index b4d299f..6b9902d 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -3,7 +3,7 @@ FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
CFLAGS+=-Wall $(shell pkg-config --cflags $(FFMPEG_LIBS))
LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
-EXAMPLES=decoding_encoding filtering metadata muxing
+EXAMPLES=decoding_encoding filtering filtering_audio metadata muxing
OBJS=$(addsuffix .o,$(EXAMPLES))
diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c
new file mode 100644
index 0000000..988dbfe
--- /dev/null
+++ b/doc/examples/filtering_audio.c
@@ -0,0 +1,245 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2012 Clément Bœsch
+ *
+ * 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.
+ */
+
+/**
+ * @file
+ * API example for audio decoding and filtering
+ */
+
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/asrc_abuffer.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/avcodec.h>
+#include <libavfilter/buffersink.h>
+
+const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
+
+static AVFormatContext *fmt_ctx;
+static AVCodecContext *dec_ctx;
+AVFilterContext *buffersink_ctx;
+AVFilterContext *buffersrc_ctx;
+AVFilterGraph *filter_graph;
+static int audio_stream_index = -1;
+
+static int open_input_file(const char *filename)
+{
+ int ret;
+ AVCodec *dec;
+
+ if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+ return ret;
+ }
+
+ if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+ return ret;
+ }
+
+ /* select the audio stream */
+ ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
+ return ret;
+ }
+ audio_stream_index = ret;
+ dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+
+ /* init the audio decoder */
+ if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int init_filters(const char *filters_descr)
+{
+ char args[512];
+ int ret;
+ AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
+ AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+ AVFilterInOut *outputs = avfilter_inout_alloc();
+ AVFilterInOut *inputs = avfilter_inout_alloc();
+ const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+ const int packing_fmts[] = { AVFILTER_PACKED, -1 };
+ const int64_t *chlayouts = avfilter_all_channel_layouts;
+ AVABufferSinkParams *abuffersink_params;
+ const AVFilterLink *outlink;
+
+ filter_graph = avfilter_graph_alloc();
+
+ /* buffer audio source: the decoded frames from the decoder will be inserted here. */
+ if (!dec_ctx->channel_layout)
+ dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
+ snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64":packed",
+ dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
+ ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
+ args, NULL, filter_graph);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
+ return ret;
+ }
+
+ /* buffer audio sink: to terminate the filter chain. */
+ abuffersink_params = av_abuffersink_params_alloc();
+ abuffersink_params->sample_fmts = sample_fmts;
+ abuffersink_params->channel_layouts = chlayouts;
+ abuffersink_params->packing_fmts = packing_fmts;
+ ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
+ NULL, abuffersink_params, filter_graph);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+ return ret;
+ }
+
+ /* Endpoints for the filter graph. */
+ outputs->name = av_strdup("in");
+ outputs->filter_ctx = buffersrc_ctx;
+ outputs->pad_idx = 0;
+ outputs->next = NULL;
+
+ inputs->name = av_strdup("out");
+ inputs->filter_ctx = buffersink_ctx;
+ inputs->pad_idx = 0;
+ inputs->next = NULL;
+
+ if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
+ &inputs, &outputs, NULL)) < 0)
+ return ret;
+
+ if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+ return ret;
+
+ /* Print summary of the sink buffer
+ * Note: args buffer is reused to store channel layout string */
+ outlink = buffersink_ctx->inputs[0];
+ av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
+ av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
+ (int)outlink->sample_rate,
+ (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
+ args);
+
+ return 0;
+}
+
+static void print_samplesref(AVFilterBufferRef *samplesref)
+{
+ const AVFilterBufferRefAudioProps *props = samplesref->audio;
+ const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
+ const uint16_t *p = (uint16_t*)samplesref->data[0];
+ const uint16_t *p_end = p + n;
+
+ while (p < p_end) {
+ fputc(*p & 0xff, stdout);
+ fputc(*p>>8 & 0xff, stdout);
+ p++;
+ }
+ fflush(stdout);
+}
+
+int main(int argc, char **argv)
+{
+ int ret;
+ AVPacket packet;
+ AVFrame frame;
+ int got_frame;
+
+ if (argc != 2) {
+ fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
+ exit(1);
+ }
+
+ avcodec_register_all();
+ av_register_all();
+ avfilter_register_all();
+
+ if ((ret = open_input_file(argv[1])) < 0)
+ goto end;
+ if ((ret = init_filters(filter_descr)) < 0)
+ goto end;
+
+ /* read all packets */
+ while (1) {
+ AVFilterBufferRef *samplesref;
+ if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+ break;
+
+ if (packet.stream_index == audio_stream_index) {
+ avcodec_get_frame_defaults(&frame);
+ got_frame = 0;
+ ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
+ av_free_packet(&packet);
+ if (ret < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+ break;
+ }
+
+ if (got_frame) {
+ const int bps = av_get_bytes_per_sample(dec_ctx->sample_fmt);
+ const int decoded_data_size = frame.nb_samples * dec_ctx->channels * bps;
+
+ /* push the audio data from decoded frame into the filtergraph */
+ if (av_asrc_buffer_add_buffer(buffersrc_ctx,
+ frame.data[0],
+ decoded_data_size,
+ dec_ctx->sample_rate,
+ dec_ctx->sample_fmt,
+ dec_ctx->channel_layout,
+ 0, frame.pts, 0) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+ break;
+ }
+
+ /* pull filtered audio from the filtergraph */
+ while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
+ av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
+ if (samplesref) {
+ print_samplesref(samplesref);
+ avfilter_unref_buffer(samplesref);
+ }
+ }
+ }
+ }
+ }
+end:
+ avfilter_graph_free(&filter_graph);
+ if (dec_ctx)
+ avcodec_close(dec_ctx);
+ avformat_close_input(&fmt_ctx);
+
+ if (ret < 0 && ret != AVERROR_EOF) {
+ char buf[1024];
+ av_strerror(ret, buf, sizeof(buf));
+ fprintf(stderr, "Error occurred: %s\n", buf);
+ exit(1);
+ }
+
+ exit(0);
+}
--
1.7.9
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