Merge commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d'

* commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d':
  examples/avcodec: split audio encoding into a separate example

Merged-by: Clément Bœsch <u@pkh.me>

6 files changed, 241 insertions, 193 deletions

diff --git a/configure b/configure
index a84b12656f..bf1d2fe1e5 100755
--- a/configure
+++ b/configure
@@ -1463,6 +1463,7 @@ EXAMPLE_LIST="
     avio_reading_example
     decoding_encoding_example
     demuxing_decoding_example
+    encode_audio_example
     extract_mvs_example
     filter_audio_example
     filtering_audio_example
@@ -3169,6 +3170,7 @@ avio_dir_cmd_deps="avformat avutil"
 avio_reading_deps="avformat avcodec avutil"
 decoding_encoding_example_deps="avcodec avformat avutil"
 demuxing_decoding_example_deps="avcodec avformat avutil"
+encode_audio_example_deps="avcodec avutil"
 extract_mvs_example_deps="avcodec avformat avutil"
 filter_audio_example_deps="avfilter avutil"
 filtering_audio_example_deps="avfilter avcodec avformat avutil"
diff --git a/doc/Makefile b/doc/Makefile
index a180fbed21..396d9e08cf 100644
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -40,6 +40,7 @@ DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
 DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE)      += encode_audio
 DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
 DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
diff --git a/doc/examples/.gitignore b/doc/examples/.gitignore
index 430d3bfbfc..1a6209ba36 100644
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -2,6 +2,7 @@
 /avio_reading
 /decoding_encoding
 /demuxing_decoding
+/encode_audio
 /extract_mvs
 /filter_audio
 /filtering_audio
diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index af3815995a..f76e5a293c 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -15,6 +15,7 @@ EXAMPLES=       avio_dir_cmd                       \
                 avio_reading                       \
                 decoding_encoding                  \
                 demuxing_decoding                  \
+                encode_audio                       \
                 extract_mvs                        \
                 filtering_video                    \
                 filtering_audio                    \
@@ -31,7 +32,7 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
-decoding_encoding: LDLIBS += -lm
+encode_audio:      LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
 
diff --git a/doc/examples/decoding_encoding.c b/doc/examples/decoding_encoding.c
index 1c5a78a44d..f863bdf4a2 100644
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -44,197 +44,6 @@
 #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("Encode audio file %s\n", filename);
-
-    /* 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);
-    if (!c) {
-        fprintf(stderr, "Could not allocate audio codec context\n");
-        exit(1);
-    }
-
-    /* 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 sample format %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);
-    if (buffer_size < 0) {
-        fprintf(stderr, "Could not get sample buffer size\n");
-        exit(1);
-    }
-    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);
-        }
-    }
-
-    /* get the delayed frames */
-    for (got_output = 1; got_output; i++) {
-        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
-        if (ret < 0) {
-            fprintf(stderr, "Error encoding 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.
  */
@@ -646,7 +455,6 @@ int main(int argc, char **argv)
     if (!strcmp(output_type, "h264")) {
         video_encode_example("test.h264", AV_CODEC_ID_H264);
     } else if (!strcmp(output_type, "mp2")) {
-        audio_encode_example("test.mp2");
         audio_decode_example("test.pcm", "test.mp2");
     } else if (!strcmp(output_type, "mpg")) {
         video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
diff --git a/doc/examples/encode_audio.c b/doc/examples/encode_audio.c
new file mode 100644
index 0000000000..5932521b2c
--- /dev/null
+++ b/doc/examples/encode_audio.c
@@ -0,0 +1,235 @@
+/*
+ * 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.
+ */
+
+/**
+ * @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);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* 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 sample format %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);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
+    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);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding 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);
+}