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avauddec: Add support for planar audio formats

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This commit is contained in:
Sebastian Dröge 2012-12-04 20:45:28 +01:00
parent 47647e1cac
commit e092f3d411
3 changed files with 89 additions and 11 deletions
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93
ext/libav/gstavauddec.c
View file

@ -229,14 +229,20 @@ gst_ffmpegauddec_get_buffer (AVCodecContext * context, AVFrame * frame)
{
GstFFMpegAudDec *ffmpegdec;
GstAudioInfo *info;
BufferInfo *buffer_info = g_slice_new (BufferInfo);
BufferInfo *buffer_info;
ffmpegdec = (GstFFMpegAudDec *) context->opaque;
if (G_UNLIKELY (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)))
goto negotiate_failed;
/* Always use the default allocator for planar audio formats because
* we will have to copy and deinterleave later anyway */
if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt))
goto fallback;
info = gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER (ffmpegdec));
buffer_info = g_slice_new (BufferInfo);
buffer_info->buffer =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (ffmpegdec),
frame->nb_samples * info->bpf);
@ -451,22 +457,87 @@ gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
if (len >= 0 && have_data > 0) {
BufferInfo *buffer_info = frame.opaque;
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (buffer_info) {
*outbuf = gst_buffer_ref (buffer_info->buffer);
} else {
*outbuf = gst_buffer_new_and_alloc (frame.linesize[0]);
gst_buffer_fill (*outbuf, 0, frame.data[0], frame.linesize[0]);
}
ffmpegdec->context->release_buffer (ffmpegdec->context, &frame);
if (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)) {
gst_buffer_unref (*outbuf);
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
len = -1;
goto beach;
}
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (buffer_info) {
*outbuf = gst_buffer_ref (buffer_info->buffer);
} else if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)) {
gint i, j;
gint nsamples, channels;
GstMapInfo minfo;
channels = ffmpegdec->info.channels;
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), frame.linesize[0] * channels);
gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
nsamples = frame.nb_samples;
switch (ffmpegdec->info.finfo->width) {
case 8:{
guint8 *odata = minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint8 *) frame.data[j])[i];
}
odata += channels;
}
break;
}
case 16:{
guint16 *odata = (guint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint16 *) frame.data[j])[i];
}
odata += channels;
}
break;
}
case 32:{
guint32 *odata = (guint32 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint32 *) frame.data[j])[i];
}
odata += channels;
}
break;
}
case 64:{
guint64 *odata = (guint64 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint64 *) frame.data[j])[i];
}
odata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (*outbuf, &minfo);
} else {
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), frame.linesize[0]);
gst_buffer_fill (*outbuf, 0, frame.data[0], frame.linesize[0]);
}
ffmpegdec->context->release_buffer (ffmpegdec->context, &frame);
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %d", have_data);
/* Reorder channels to the GStreamer channel order */

2
ext/libav/gstavcodecmap.c
View file

@ -1747,6 +1747,8 @@ gst_ffmpeg_pixfmt_to_caps (enum PixelFormat pix_fmt, AVCodecContext * context,
GstAudioFormat
gst_ffmpeg_smpfmt_to_audioformat (enum AVSampleFormat sample_fmt)
{
sample_fmt = av_get_packed_sample_fmt (sample_fmt);
switch (sample_fmt) {
case AV_SAMPLE_FMT_S16:
return GST_AUDIO_FORMAT_S16;

5
ext/libav/gstavutils.c
View file

@ -46,16 +46,21 @@ av_smp_format_depth (enum AVSampleFormat smp_fmt)
gint depth = -1;
switch (smp_fmt) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_U8P:
depth = 1;
break;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P:
depth = 2;
break;
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_S32P:
case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_FLTP:
depth = 4;
break;
case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_DBLP:
depth = 8;
break;
default: