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avaudenc: Implement draining

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This commit is contained in:
Sebastian Dröge 2012-11-20 11:51:13 +01:00
parent e281493f29
commit 5ffd21f1ee
1 changed files with 34 additions and 6 deletions
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40
ext/libav/gstavaudenc.c
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@ -170,7 +170,7 @@ gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc)
ffmpegaudenc->context = avcodec_alloc_context ();
ffmpegaudenc->opened = FALSE;
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), FALSE);
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), TRUE);
}
static void
@ -355,7 +355,7 @@ gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
static GstFlowReturn
gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
guint8 * audio_in, guint in_size)
guint8 * audio_in, guint in_size, gint * have_data)
{
AVCodecContext *ctx;
gint res;
@ -363,7 +363,6 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
GstAudioInfo *info;
AVPacket pkt;
AVFrame frame;
gint got_output;
ctx = ffmpegaudenc->context;
@ -377,14 +376,14 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
frame.linesize[0] = in_size;
frame.nb_samples = in_size / info->bpf;
res = avcodec_encode_audio2 (ctx, &pkt, &frame, &got_output);
res = avcodec_encode_audio2 (ctx, &pkt, &frame, have_data);
if (res < 0) {
GST_ERROR_OBJECT (ffmpegaudenc, "Failed to encode buffer: %d", res);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (ffmpegaudenc, "got output size %d", res);
if (got_output) {
if (*have_data) {
GstBuffer *outbuf;
GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", pkt.size);
@ -403,6 +402,29 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
return ret;
}
static void
gst_ffmpegaudenc_drain (GstFFMpegAudEnc * ffmpegaudenc)
{
GstFFMpegAudEncClass *oclass;
oclass = (GstFFMpegAudEncClass *) (G_OBJECT_GET_CLASS (ffmpegaudenc));
if (oclass->in_plugin->capabilities & CODEC_CAP_DELAY) {
gint have_data, try = 0;
GST_LOG_OBJECT (ffmpegaudenc,
"codec has delay capabilities, calling until libav has drained everything");
do {
GstFlowReturn ret;
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, NULL, 0, &have_data);
if (ret != GST_FLOW_OK || have_data == 0)
break;
} while (try++ < 10);
}
}
static GstFlowReturn
gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
{
@ -411,12 +433,18 @@ gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
GstFlowReturn ret;
guint8 *in_data;
GstMapInfo map;
gint have_data;
ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
if (G_UNLIKELY (!ffmpegaudenc->opened))
goto not_negotiated;
if (!inbuf) {
gst_ffmpegaudenc_drain (ffmpegaudenc);
return GST_FLOW_OK;
}
inbuf = gst_buffer_ref (inbuf);
GST_DEBUG_OBJECT (ffmpegaudenc,
@ -427,7 +455,7 @@ gst_ffmpegaudenc_handle_frame (GstAudioEncoder * encoder, GstBuffer * inbuf)
gst_buffer_map (inbuf, &map, GST_MAP_READ);
in_data = map.data;
size = map.size;
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size);
ret = gst_ffmpegaudenc_encode_audio (ffmpegaudenc, in_data, size, &have_data);
gst_buffer_unmap (inbuf, &map);
gst_buffer_unref (inbuf);