[FFmpeg-cvslog] examples/avcodec: split audio encoding into a separate example
Anton Khirnov
git at videolan.org
Wed Mar 29 14:29:49 EEST 2017
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Wed Oct 19 21:05:22 2016 +0200| [40aaa8dadfd1c69ff4460d04750e1403b5535a6d] | committer: Anton Khirnov
examples/avcodec: split audio encoding into a separate example
The four examples (audio/video encoding/decoding) are completely
independent so it makes little sense to have them all in one file.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=40aaa8dadfd1c69ff4460d04750e1403b5535a6d
---
configure | 2 +
doc/Makefile | 3 +-
doc/examples/avcodec.c | 170 -----------------------------------
doc/examples/encode_audio.c | 211 ++++++++++++++++++++++++++++++++++++++++++++
4 files changed, 215 insertions(+), 171 deletions(-)
diff --git a/configure b/configure
index 8c2b4fd..28567bb 100755
--- a/configure
+++ b/configure
@@ -1210,6 +1210,7 @@ COMPONENT_LIST="
EXAMPLE_LIST="
avcodec_example
+ encode_audio_example
filter_audio_example
metadata_example
output_example
@@ -2435,6 +2436,7 @@ scale_vaapi_filter_deps="vaapi VAProcPipelineParameterBuffer"
# examples
avcodec_example_deps="avcodec avutil"
+encode_audio_example_deps="avcodec avutil"
filter_audio_example_deps="avfilter avutil"
metadata_example_deps="avformat avutil"
output_example_deps="avcodec avformat avutil swscale"
diff --git a/doc/Makefile b/doc/Makefile
index c464a48..738e601 100644
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -17,12 +17,13 @@ DOCS-$(CONFIG_TEXI2HTML) += $(HTMLPAGES)
DOCS = $(DOCS-yes)
DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec
+DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
DOC_EXAMPLES-$(CONFIG_OUTPUT_EXAMPLE) += output
DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
-ALL_DOC_EXAMPLES = avcodec filter_audio metadata output transcode_aac
+ALL_DOC_EXAMPLES = avcodec encode_audio filter_audio metadata output transcode_aac
DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(EXESUF))
ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES:%=doc/examples/%$(EXESUF))
diff --git a/doc/examples/avcodec.c b/doc/examples/avcodec.c
index 8fee552..63812d9 100644
--- a/doc/examples/avcodec.c
+++ b/doc/examples/avcodec.c
@@ -47,175 +47,6 @@
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
-/* check that a given sample format is supported by the encoder */
-static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
-{
- const enum AVSampleFormat *p = codec->sample_fmts;
-
- while (*p != AV_SAMPLE_FMT_NONE) {
- if (*p == sample_fmt)
- return 1;
- p++;
- }
- return 0;
-}
-
-/* just pick the highest supported samplerate */
-static int select_sample_rate(AVCodec *codec)
-{
- const int *p;
- int best_samplerate = 0;
-
- if (!codec->supported_samplerates)
- return 44100;
-
- p = codec->supported_samplerates;
- while (*p) {
- best_samplerate = FFMAX(*p, best_samplerate);
- p++;
- }
- return best_samplerate;
-}
-
-/* select layout with the highest channel count */
-static int select_channel_layout(AVCodec *codec)
-{
- const uint64_t *p;
- uint64_t best_ch_layout = 0;
- int best_nb_channels = 0;
-
- if (!codec->channel_layouts)
- return AV_CH_LAYOUT_STEREO;
-
- p = codec->channel_layouts;
- while (*p) {
- int nb_channels = av_get_channel_layout_nb_channels(*p);
-
- if (nb_channels > best_nb_channels) {
- best_ch_layout = *p;
- best_nb_channels = nb_channels;
- }
- p++;
- }
- return best_ch_layout;
-}
-
-/*
- * Audio encoding example
- */
-static void audio_encode_example(const char *filename)
-{
- AVCodec *codec;
- AVCodecContext *c= NULL;
- AVFrame *frame;
- AVPacket pkt;
- int i, j, k, ret, got_output;
- int buffer_size;
- FILE *f;
- uint16_t *samples;
- float t, tincr;
-
- printf("Audio encoding\n");
-
- /* find the MP2 encoder */
- codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
- if (!codec) {
- fprintf(stderr, "codec not found\n");
- exit(1);
- }
-
- c = avcodec_alloc_context3(codec);
-
- /* put sample parameters */
- c->bit_rate = 64000;
-
- /* check that the encoder supports s16 pcm input */
- c->sample_fmt = AV_SAMPLE_FMT_S16;
- if (!check_sample_fmt(codec, c->sample_fmt)) {
- fprintf(stderr, "encoder does not support %s",
- av_get_sample_fmt_name(c->sample_fmt));
- exit(1);
- }
-
- /* select other audio parameters supported by the encoder */
- c->sample_rate = select_sample_rate(codec);
- c->channel_layout = select_channel_layout(codec);
- c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
-
- /* open it */
- if (avcodec_open2(c, codec, NULL) < 0) {
- fprintf(stderr, "could not open codec\n");
- exit(1);
- }
-
- f = fopen(filename, "wb");
- if (!f) {
- fprintf(stderr, "could not open %s\n", filename);
- exit(1);
- }
-
- /* frame containing input raw audio */
- frame = av_frame_alloc();
- if (!frame) {
- fprintf(stderr, "could not allocate audio frame\n");
- exit(1);
- }
-
- frame->nb_samples = c->frame_size;
- frame->format = c->sample_fmt;
- frame->channel_layout = c->channel_layout;
-
- /* the codec gives us the frame size, in samples,
- * we calculate the size of the samples buffer in bytes */
- buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
- c->sample_fmt, 0);
- samples = av_malloc(buffer_size);
- if (!samples) {
- fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
- buffer_size);
- exit(1);
- }
- /* setup the data pointers in the AVFrame */
- ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
- (const uint8_t*)samples, buffer_size, 0);
- if (ret < 0) {
- fprintf(stderr, "could not setup audio frame\n");
- exit(1);
- }
-
- /* encode a single tone sound */
- t = 0;
- tincr = 2 * M_PI * 440.0 / c->sample_rate;
- for(i=0;i<200;i++) {
- av_init_packet(&pkt);
- pkt.data = NULL; // packet data will be allocated by the encoder
- pkt.size = 0;
-
- for (j = 0; j < c->frame_size; j++) {
- samples[2*j] = (int)(sin(t) * 10000);
-
- for (k = 1; k < c->channels; k++)
- samples[2*j + k] = samples[2*j];
- t += tincr;
- }
- /* encode the samples */
- ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
- if (ret < 0) {
- fprintf(stderr, "error encoding audio frame\n");
- exit(1);
- }
- if (got_output) {
- fwrite(pkt.data, 1, pkt.size, f);
- av_packet_unref(&pkt);
- }
- }
- fclose(f);
-
- av_freep(&samples);
- av_frame_free(&frame);
- avcodec_free_context(&c);
-}
-
/*
* Audio decoding.
*/
@@ -575,7 +406,6 @@ int main(int argc, char **argv)
avcodec_register_all();
if (argc <= 1) {
- audio_encode_example("/tmp/test.mp2");
audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
video_encode_example("/tmp/test.mpg");
diff --git a/doc/examples/encode_audio.c b/doc/examples/encode_audio.c
new file mode 100644
index 0000000..cabe589
--- /dev/null
+++ b/doc/examples/encode_audio.c
@@ -0,0 +1,211 @@
+/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio encoding with libavcodec API example.
+ *
+ * @example encode_audio.c
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/frame.h"
+#include "libavutil/samplefmt.h"
+
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
+{
+ const enum AVSampleFormat *p = codec->sample_fmts;
+
+ while (*p != AV_SAMPLE_FMT_NONE) {
+ if (*p == sample_fmt)
+ return 1;
+ p++;
+ }
+ return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(AVCodec *codec)
+{
+ const int *p;
+ int best_samplerate = 0;
+
+ if (!codec->supported_samplerates)
+ return 44100;
+
+ p = codec->supported_samplerates;
+ while (*p) {
+ best_samplerate = FFMAX(*p, best_samplerate);
+ p++;
+ }
+ return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(AVCodec *codec)
+{
+ const uint64_t *p;
+ uint64_t best_ch_layout = 0;
+ int best_nb_channels = 0;
+
+ if (!codec->channel_layouts)
+ return AV_CH_LAYOUT_STEREO;
+
+ p = codec->channel_layouts;
+ while (*p) {
+ int nb_channels = av_get_channel_layout_nb_channels(*p);
+
+ if (nb_channels > best_nb_channels) {
+ best_ch_layout = *p;
+ best_nb_channels = nb_channels;
+ }
+ p++;
+ }
+ return best_ch_layout;
+}
+
+int main(int argc, char **argv)
+{
+ const char *filename;
+ AVCodec *codec;
+ AVCodecContext *c= NULL;
+ AVFrame *frame;
+ AVPacket pkt;
+ int i, j, k, ret, got_output;
+ int buffer_size;
+ FILE *f;
+ uint16_t *samples;
+ float t, tincr;
+
+ if (argc <= 1) {
+ fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
+ return 0;
+ }
+ filename = argv[1];
+
+ /* register all the codecs */
+ avcodec_register_all();
+
+ /* find the MP2 encoder */
+ codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+ if (!codec) {
+ fprintf(stderr, "codec not found\n");
+ exit(1);
+ }
+
+ c = avcodec_alloc_context3(codec);
+
+ /* put sample parameters */
+ c->bit_rate = 64000;
+
+ /* check that the encoder supports s16 pcm input */
+ c->sample_fmt = AV_SAMPLE_FMT_S16;
+ if (!check_sample_fmt(codec, c->sample_fmt)) {
+ fprintf(stderr, "encoder does not support %s",
+ av_get_sample_fmt_name(c->sample_fmt));
+ exit(1);
+ }
+
+ /* select other audio parameters supported by the encoder */
+ c->sample_rate = select_sample_rate(codec);
+ c->channel_layout = select_channel_layout(codec);
+ c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
+
+ /* open it */
+ if (avcodec_open2(c, codec, NULL) < 0) {
+ fprintf(stderr, "could not open codec\n");
+ exit(1);
+ }
+
+ f = fopen(filename, "wb");
+ if (!f) {
+ fprintf(stderr, "could not open %s\n", filename);
+ exit(1);
+ }
+
+ /* frame containing input raw audio */
+ frame = av_frame_alloc();
+ if (!frame) {
+ fprintf(stderr, "could not allocate audio frame\n");
+ exit(1);
+ }
+
+ frame->nb_samples = c->frame_size;
+ frame->format = c->sample_fmt;
+ frame->channel_layout = c->channel_layout;
+
+ /* the codec gives us the frame size, in samples,
+ * we calculate the size of the samples buffer in bytes */
+ buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
+ c->sample_fmt, 0);
+ samples = av_malloc(buffer_size);
+ if (!samples) {
+ fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
+ buffer_size);
+ exit(1);
+ }
+ /* setup the data pointers in the AVFrame */
+ ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+ (const uint8_t*)samples, buffer_size, 0);
+ if (ret < 0) {
+ fprintf(stderr, "could not setup audio frame\n");
+ exit(1);
+ }
+
+ /* encode a single tone sound */
+ t = 0;
+ tincr = 2 * M_PI * 440.0 / c->sample_rate;
+ for(i=0;i<200;i++) {
+ av_init_packet(&pkt);
+ pkt.data = NULL; // packet data will be allocated by the encoder
+ pkt.size = 0;
+
+ for (j = 0; j < c->frame_size; j++) {
+ samples[2*j] = (int)(sin(t) * 10000);
+
+ for (k = 1; k < c->channels; k++)
+ samples[2*j + k] = samples[2*j];
+ t += tincr;
+ }
+ /* encode the samples */
+ ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
+ if (ret < 0) {
+ fprintf(stderr, "error encoding audio frame\n");
+ exit(1);
+ }
+ if (got_output) {
+ fwrite(pkt.data, 1, pkt.size, f);
+ av_packet_unref(&pkt);
+ }
+ }
+ fclose(f);
+
+ av_freep(&samples);
+ av_frame_free(&frame);
+ avcodec_free_context(&c);
+}
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