adding ffmpeg specific version

1 files changed, 184 insertions, 0 deletions

diff --git a/ffmpeg1/libavfilter/audio.c b/ffmpeg1/libavfilter/audio.c
new file mode 100644
index 0000000..1075217
--- /dev/null
+++ b/ffmpeg1/libavfilter/audio.c
@@ -0,0 +1,184 @@
+/*
+ * Copyright (c) Stefano Sabatini | stefasab at gmail.com
+ * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavcodec/avcodec.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+int avfilter_ref_get_channels(AVFilterBufferRef *ref)
+{
+    return ref->audio ? ref->audio->channels : 0;
+}
+
+AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples)
+{
+    return ff_get_audio_buffer(link->dst->outputs[0], nb_samples);
+}
+
+AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples)
+{
+    AVFrame *frame = av_frame_alloc();
+    int channels = link->channels;
+    int buf_size, ret;
+
+    av_assert0(channels == av_get_channel_layout_nb_channels(link->channel_layout) || !av_get_channel_layout_nb_channels(link->channel_layout));
+
+    if (!frame)
+        return NULL;
+
+    buf_size = av_samples_get_buffer_size(NULL, channels, nb_samples,
+                                          link->format, 0);
+    if (buf_size < 0)
+        goto fail;
+
+    frame->buf[0] = av_buffer_alloc(buf_size);
+    if (!frame->buf[0])
+        goto fail;
+
+    frame->nb_samples = nb_samples;
+    ret = avcodec_fill_audio_frame(frame, channels, link->format,
+                                   frame->buf[0]->data, buf_size, 0);
+    if (ret < 0)
+        goto fail;
+
+    av_samples_set_silence(frame->extended_data, 0, nb_samples, channels,
+                           link->format);
+
+    frame->nb_samples     = nb_samples;
+    frame->format         = link->format;
+    av_frame_set_channels(frame, link->channels);
+    frame->channel_layout = link->channel_layout;
+    frame->sample_rate    = link->sample_rate;
+
+    return frame;
+
+fail:
+    av_buffer_unref(&frame->buf[0]);
+    av_frame_free(&frame);
+    return NULL;
+}
+
+AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
+{
+    AVFrame *ret = NULL;
+
+    if (link->dstpad->get_audio_buffer)
+        ret = link->dstpad->get_audio_buffer(link, nb_samples);
+
+    if (!ret)
+        ret = ff_default_get_audio_buffer(link, nb_samples);
+
+    return ret;
+}
+
+#if FF_API_AVFILTERBUFFER
+AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_channels(uint8_t **data,
+                                                                      int linesize,int perms,
+                                                                      int nb_samples,
+                                                                      enum AVSampleFormat sample_fmt,
+                                                                      int channels,
+                                                                      uint64_t channel_layout)
+{
+    int planes;
+    AVFilterBuffer    *samples    = av_mallocz(sizeof(*samples));
+    AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
+
+    if (!samples || !samplesref)
+        goto fail;
+
+    av_assert0(channels);
+    av_assert0(channel_layout == 0 ||
+               channels == av_get_channel_layout_nb_channels(channel_layout));
+
+    samplesref->buf         = samples;
+    samplesref->buf->free   = ff_avfilter_default_free_buffer;
+    if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
+        goto fail;
+
+    samplesref->audio->nb_samples     = nb_samples;
+    samplesref->audio->channel_layout = channel_layout;
+    samplesref->audio->channels       = channels;
+
+    planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
+
+    /* make sure the buffer gets read permission or it's useless for output */
+    samplesref->perms = perms | AV_PERM_READ;
+
+    samples->refcount  = 1;
+    samplesref->type   = AVMEDIA_TYPE_AUDIO;
+    samplesref->format = sample_fmt;
+
+    memcpy(samples->data, data,
+           FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
+    memcpy(samplesref->data, samples->data, sizeof(samples->data));
+
+    samples->linesize[0] = samplesref->linesize[0] = linesize;
+
+    if (planes > FF_ARRAY_ELEMS(samples->data)) {
+        samples->   extended_data = av_mallocz(sizeof(*samples->extended_data) *
+                                               planes);
+        samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
+                                               planes);
+
+        if (!samples->extended_data || !samplesref->extended_data)
+            goto fail;
+
+        memcpy(samples->   extended_data, data, sizeof(*data)*planes);
+        memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
+    } else {
+        samples->extended_data    = samples->data;
+        samplesref->extended_data = samplesref->data;
+    }
+
+    samplesref->pts = AV_NOPTS_VALUE;
+
+    return samplesref;
+
+fail:
+    if (samples && samples->extended_data != samples->data)
+        av_freep(&samples->extended_data);
+    if (samplesref) {
+        av_freep(&samplesref->audio);
+        if (samplesref->extended_data != samplesref->data)
+            av_freep(&samplesref->extended_data);
+    }
+    av_freep(&samplesref);
+    av_freep(&samples);
+    return NULL;
+}
+
+AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
+                                                             int linesize,int perms,
+                                                             int nb_samples,
+                                                             enum AVSampleFormat sample_fmt,
+                                                             uint64_t channel_layout)
+{
+    int channels = av_get_channel_layout_nb_channels(channel_layout);
+    return avfilter_get_audio_buffer_ref_from_arrays_channels(data, linesize, perms,
+                                                              nb_samples, sample_fmt,
+                                                              channels, channel_layout);
+}
+#endif