[FFmpeg-cvslog] lavf/output-example: use new audio encoding API correctly.

[Anton Khirnov]

ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Fri Feb 24 08:59:38 2012 +0100| [5ff42e3138998ef5207ca793735409105897c6f2] | committer: Anton Khirnov

lavf/output-example: use new audio encoding API correctly.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5ff42e3138998ef5207ca793735409105897c6f2
---

 libavformat/output-example.c |   44 ++++++++++++++---------------------------
 1 files changed, 15 insertions(+), 29 deletions(-)

diff --git a/libavformat/output-example.c b/libavformat/output-example.c
index 38ce377..86324b4 100644
--- a/libavformat/output-example.c
+++ b/libavformat/output-example.c
@@ -53,8 +53,6 @@ static int sws_flags = SWS_BICUBIC;
 
 static float t, tincr, tincr2;
 static int16_t *samples;
-static uint8_t *audio_outbuf;
-static int audio_outbuf_size;
 static int audio_input_frame_size;
 
 /*
@@ -112,27 +110,12 @@ static void open_audio(AVFormatContext *oc, AVStream *st)
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    audio_outbuf_size = 10000;
-    audio_outbuf = av_malloc(audio_outbuf_size);
-
-    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
-       support to compute the input frame size in samples */
-    if (c->frame_size <= 1) {
-        audio_input_frame_size = audio_outbuf_size / c->channels;
-        switch(st->codec->codec_id) {
-        case CODEC_ID_PCM_S16LE:
-        case CODEC_ID_PCM_S16BE:
-        case CODEC_ID_PCM_U16LE:
-        case CODEC_ID_PCM_U16BE:
-            audio_input_frame_size >>= 1;
-            break;
-        default:
-            break;
-        }
-    } else {
+    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
+        audio_input_frame_size = 10000;
+    else
         audio_input_frame_size = c->frame_size;
-    }
-    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
+    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
@@ -156,19 +139,23 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt;
-    av_init_packet(&pkt);
+    AVFrame *frame = avcodec_alloc_frame();
+    int got_packet;
 
+    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);
 
-    pkt.size = avcodec_encode_audio2(c, audio_outbuf, audio_outbuf_size, samples);
+    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (!got_packet)
+        return;
 
-    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
-        pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
-    pkt.flags |= AV_PKT_FLAG_KEY;
     pkt.stream_index= st->index;
-    pkt.data= audio_outbuf;
 
     /* write the compressed frame in the media file */
     if (av_interleaved_write_frame(oc, &pkt) != 0) {
@@ -182,7 +169,6 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
 
     av_free(samples);
-    av_free(audio_outbuf);
 }
 
 /**************************************************************/