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avaudenc: fix audio encoder flushing according to libav docs

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* @param[in] frame AVFrame containing the raw audio data to be encoded.
 *                  May be NULL when flushing an encoder that has the
 *                  CODEC_CAP_DELAY capability set.

The AVFrame itself should be null, not the frame.data pointer

https://bugzilla.gnome.org/show_bug.cgi?id=724536
This commit is contained in:
Thiago Santos 2014-02-20 17:25:35 -03:00
parent 7f5132b34f
commit 845b874575
1 changed files with 87 additions and 77 deletions
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164
ext/libav/gstavaudenc.c
View file

@ -425,98 +425,108 @@ gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
ctx = ffmpegaudenc->context; ctx = ffmpegaudenc->context;
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer "); GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer %p size:%u", audio_in,
in_size);
memset (&pkt, 0, sizeof (pkt)); memset (&pkt, 0, sizeof (pkt));
memset (&frame, 0, sizeof (frame));
avcodec_get_frame_defaults (&frame);
info = gst_audio_encoder_get_audio_info (enc); if (audio_in != NULL) {
planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt); memset (&frame, 0, sizeof (frame));
avcodec_get_frame_defaults (&frame);
if (planar && info->channels > 1) { info = gst_audio_encoder_get_audio_info (enc);
gint channels, nsamples; planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt);
gint i, j;
nsamples = frame.nb_samples = in_size / info->bpf; if (planar && info->channels > 1) {
channels = info->channels; gint channels, nsamples;
gint i, j;
nsamples = frame.nb_samples = in_size / info->bpf;
channels = info->channels;
if (info->channels > AV_NUM_DATA_POINTERS) {
frame.extended_data = g_new (uint8_t *, info->channels);
} else {
frame.extended_data = frame.data;
}
frame.extended_data[0] = g_malloc (in_size);
frame.linesize[0] = in_size / channels;
for (i = 1; i < channels; i++)
frame.extended_data[i] = frame.extended_data[i - 1] + frame.linesize[0];
switch (info->finfo->width) {
case 8:{
const guint8 *idata = (const guint8 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint8 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 16:{
const guint16 *idata = (const guint16 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint16 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 32:{
const guint32 *idata = (const guint32 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint32 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 64:{
const guint64 *idata = (const guint64 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint64 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
if (info->channels > AV_NUM_DATA_POINTERS) {
frame.extended_data = g_new (uint8_t *, info->channels);
} else { } else {
frame.data[0] = audio_in;
frame.extended_data = frame.data; frame.extended_data = frame.data;
frame.linesize[0] = in_size;
frame.nb_samples = in_size / info->bpf;
} }
frame.extended_data[0] = g_malloc (in_size); /* we have a frame to feed the encoder */
frame.linesize[0] = in_size / channels; res = avcodec_encode_audio2 (ctx, &pkt, &frame, have_data);
for (i = 1; i < channels; i++)
frame.extended_data[i] = frame.extended_data[i - 1] + frame.linesize[0];
switch (info->finfo->width) { if (planar && info->channels > 1)
case 8:{ g_free (frame.data[0]);
const guint8 *idata = (const guint8 *) audio_in; if (frame.extended_data != frame.data)
g_free (frame.extended_data);
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint8 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 16:{
const guint16 *idata = (const guint16 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint16 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 32:{
const guint32 *idata = (const guint32 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint32 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 64:{
const guint64 *idata = (const guint64 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint64 *) frame.extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
} else { } else {
frame.data[0] = audio_in; /* flushing the encoder */
frame.extended_data = frame.data; res = avcodec_encode_audio2 (ctx, &pkt, NULL, have_data);
frame.linesize[0] = in_size;
frame.nb_samples = in_size / info->bpf;
} }
res = avcodec_encode_audio2 (ctx, &pkt, &frame, have_data);
if (planar && info->channels > 1)
g_free (frame.data[0]);
if (frame.extended_data != frame.data)
g_free (frame.extended_data);
if (res < 0) { if (res < 0) {
char error_str[128] = { 0, }; char error_str[128] = { 0, };