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authorAnton Khirnov <anton@khirnov.net>2014-06-24 07:51:18 +0200
committerAnton Khirnov <anton@khirnov.net>2014-06-26 16:06:54 +0200
commit5e7b125b6ae36893dfd9cb5661c99b67363cbb38 (patch)
tree259b529412a6616b602ffcf8f2fd67edd01ae305 /doc/examples
parentac85f631c9a9cc59aaca1c8dd6894fb1f701c594 (diff)
output example: use the new AVFrame API to allocate audio frames
Diffstat (limited to 'doc/examples')
-rw-r--r--doc/examples/output.c43
1 files changed, 24 insertions, 19 deletions
diff --git a/doc/examples/output.c b/doc/examples/output.c
index 3d1444912f..1f8ff9642f 100644
--- a/doc/examples/output.c
+++ b/doc/examples/output.c
@@ -59,7 +59,6 @@ typedef struct OutputStream {
/* audio output */
static float t, tincr, tincr2;
-static int16_t *samples;
static int audio_input_frame_size;
/*
@@ -122,20 +121,24 @@ static void open_audio(AVFormatContext *oc, AVStream *st)
audio_input_frame_size = 10000;
else
audio_input_frame_size = c->frame_size;
- samples = av_malloc(audio_input_frame_size *
- av_get_bytes_per_sample(c->sample_fmt) *
- c->channels);
}
/* 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];
- q = samples;
- for (j = 0; j < frame_size; j++) {
+ /* 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);
+
+ for (j = 0; j < frame->nb_samples; j++) {
v = (int)(sin(t) * 10000);
for (i = 0; i < nb_channels; i++)
*q++ = v;
@@ -149,18 +152,22 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame = av_frame_alloc();
- int got_packet;
+ int got_packet, ret;
av_init_packet(&pkt);
c = st->codec;
- get_audio_frame(samples, audio_input_frame_size, c->channels);
- frame->nb_samples = audio_input_frame_size;
- avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
- (uint8_t *)samples,
- audio_input_frame_size *
- av_get_bytes_per_sample(c->sample_fmt) *
- c->channels, 1);
+ frame->sample_rate = c->sample_rate;
+ frame->nb_samples = audio_input_frame_size;
+ 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);
avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (!got_packet)
@@ -179,8 +186,6 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
-
- av_free(samples);
}
/**************************************************************/