diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index e26c25da1626051c4fc9554797444b4cf59c3e8f..5eb01354992cd9b868959e647e95fed547243405 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -156,15 +156,8 @@ static AVStream *add_stream(OutputStream *ost, AVFormatContext *oc,
static float t, tincr, tincr2;
-AVFrame *audio_frame;
-static uint8_t **src_samples_data;
-static int src_samples_linesize;
static int src_nb_samples;
-static int max_dst_nb_samples;
-uint8_t **dst_samples_data;
-int dst_samples_linesize;
-int dst_samples_size;
int samples_count;
struct SwrContext *swr_ctx = NULL;
@@ -176,13 +169,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
c = st->codec;
- /* allocate and init a re-usable frame */
- audio_frame = av_frame_alloc();
- if (!audio_frame) {
- fprintf(stderr, "Could not allocate audio frame\n");
- exit(1);
- }
-
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
@@ -196,20 +182,10 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
/* increment frequency by 110 Hz per second */
tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
- src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
- 10000 : c->frame_size;
-
- ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
- src_nb_samples, AV_SAMPLE_FMT_S16, 0);
- if (ret < 0) {
- fprintf(stderr, "Could not allocate source samples\n");
- exit(1);
- }
-
- /* compute the number of converted samples: buffering is avoided
- * ensuring that the output buffer will contain at least all the
- * converted input samples */
- max_dst_nb_samples = src_nb_samples;
+ if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
+ src_nb_samples = 10000;
+ else
+ src_nb_samples = c->frame_size;
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
@@ -232,29 +208,25 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
-
- ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
- max_dst_nb_samples, c->sample_fmt, 0);
- if (ret < 0) {
- fprintf(stderr, "Could not allocate destination samples\n");
- exit(1);
- }
- } else {
- dst_samples_data = src_samples_data;
}
- dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
- c->sample_fmt, 0);
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
-static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
+static void get_audio_frame(AVFrame *frame, int nb_channels)
{
- int j, i, v;
- int16_t *q;
+ int j, i, v, ret;
+ int16_t *q = (int16_t*)frame->data[0];
+
+ /* when we pass a frame to the encoder, it may keep a reference to it
+ * internally;
+ * make sure we do not overwrite it here
+ */
+ ret = av_frame_make_writable(frame);
+ if (ret < 0)
+ exit(1);
- q = samples;
- for (j = 0; j < frame_size; j++) {
+ for (j = 0; j < frame->nb_samples; j++) {
v = (int)(sin(t) * 10000);
for (i = 0; i < nb_channels; i++)
*q++ = v;
@@ -267,50 +239,63 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
- int got_packet, ret, dst_nb_samples;
+ AVFrame *frame = av_frame_alloc();
+ int got_packet, ret;
+ int dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
if (!flush) {
- get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+ frame->sample_rate = c->sample_rate;
+ frame->nb_samples = src_nb_samples;
+ frame->format = AV_SAMPLE_FMT_S16;
+ frame->channel_layout = c->channel_layout;
+ ret = av_frame_get_buffer(frame, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate an audio frame.\n");
+ exit(1);
+ }
+
+ get_audio_frame(frame, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
+ AVFrame *tmp_frame = av_frame_alloc();
+
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
- if (dst_nb_samples > max_dst_nb_samples) {
- av_free(dst_samples_data[0]);
- ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
- dst_nb_samples, c->sample_fmt, 0);
- if (ret < 0)
- exit(1);
- max_dst_nb_samples = dst_nb_samples;
- dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
- c->sample_fmt, 0);
+ tmp_frame->sample_rate = c->sample_rate;
+ tmp_frame->nb_samples = dst_nb_samples;
+ tmp_frame->format = c->sample_fmt;
+ tmp_frame->channel_layout = c->channel_layout;
+ ret = av_frame_get_buffer(tmp_frame, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate an audio frame.\n");
+ exit(1);
}
/* convert to destination format */
ret = swr_convert(swr_ctx,
- dst_samples_data, dst_nb_samples,
- (const uint8_t **)src_samples_data, src_nb_samples);
+ tmp_frame->data, dst_nb_samples,
+ (const uint8_t **)frame->data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
+ av_frame_free(&frame);
+ frame = tmp_frame;
} else {
dst_nb_samples = src_nb_samples;
}
- audio_frame->nb_samples = dst_nb_samples;
- audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
- avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
- dst_samples_data[0], dst_samples_size, 0);
+ frame->nb_samples = dst_nb_samples;
+ frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
samples_count += dst_nb_samples;
}
- ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
+ ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
@@ -333,13 +318,6 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
- if (dst_samples_data != src_samples_data) {
- av_free(dst_samples_data[0]);
- av_free(dst_samples_data);
- }
- av_free(src_samples_data[0]);
- av_free(src_samples_data);
- av_frame_free(&audio_frame);
}
/**************************************************************/