[FFmpeg-cvslog] lavfi: move audio-related functions to a separate file.

Anton Khirnov git at videolan.org
Thu May 10 23:33:13 CEST 2012


ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Mon May  7 11:21:38 2012 +0200| [0b45334a5880d6e2a4b3642adcd5feab8a27a150] | committer: Anton Khirnov

lavfi: move audio-related functions to a separate file.

This is easier to follow than having them randomly scattered in
avfilter.c and defaults.c.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0b45334a5880d6e2a4b3642adcd5feab8a27a150
---

 libavfilter/Makefile   |    1 +
 libavfilter/audio.c    |  209 ++++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/avfilter.c |  127 +-----------------------------
 libavfilter/defaults.c |   64 ---------------
 libavfilter/internal.h |    4 +
 5 files changed, 215 insertions(+), 190 deletions(-)

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e786b6d..49a47d3 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -10,6 +10,7 @@ HEADERS = avfilter.h                                                    \
           vsrc_buffer.h                                                 \
 
 OBJS = allfilters.o                                                     \
+       audio.o                                                          \
        avfilter.o                                                       \
        avfiltergraph.o                                                  \
        buffersink.o                                                     \
diff --git a/libavfilter/audio.c b/libavfilter/audio.c
new file mode 100644
index 0000000..3e12c69
--- /dev/null
+++ b/libavfilter/audio.c
@@ -0,0 +1,209 @@
+/*
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/audioconvert.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
+                                            int nb_samples)
+{
+    return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
+}
+
+AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
+                                               int nb_samples)
+{
+    AVFilterBufferRef *samplesref = NULL;
+    uint8_t **data;
+    int planar      = av_sample_fmt_is_planar(link->format);
+    int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+    int planes      = planar ? nb_channels : 1;
+    int linesize;
+
+    if (!(data = av_mallocz(sizeof(*data) * planes)))
+        goto fail;
+
+    if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
+        goto fail;
+
+    samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
+                                                           nb_samples, link->format,
+                                                           link->channel_layout);
+    if (!samplesref)
+        goto fail;
+
+    av_freep(&data);
+
+fail:
+    if (data)
+        av_freep(&data[0]);
+    av_freep(&data);
+    return samplesref;
+}
+
+AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
+                                       int nb_samples)
+{
+    AVFilterBufferRef *ret = NULL;
+
+    if (link->dstpad->get_audio_buffer)
+        ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
+
+    if (!ret)
+        ret = ff_default_get_audio_buffer(link, perms, nb_samples);
+
+    if (ret)
+        ret->type = AVMEDIA_TYPE_AUDIO;
+
+    return ret;
+}
+
+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 planes;
+    AVFilterBuffer    *samples    = av_mallocz(sizeof(*samples));
+    AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
+
+    if (!samples || !samplesref)
+        goto fail;
+
+    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->planar         = av_sample_fmt_is_planar(sample_fmt);
+
+    planes = samplesref->audio->planar ? av_get_channel_layout_nb_channels(channel_layout) : 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;
+    }
+
+    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;
+}
+
+void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+    ff_filter_samples(link->dst->outputs[0], samplesref);
+}
+
+/* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */
+void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+{
+    AVFilterLink *outlink = NULL;
+
+    if (inlink->dst->output_count)
+        outlink = inlink->dst->outputs[0];
+
+    if (outlink) {
+        outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE,
+                                                       samplesref->audio->nb_samples);
+        outlink->out_buf->pts                = samplesref->pts;
+        outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;
+        ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));
+        avfilter_unref_buffer(outlink->out_buf);
+        outlink->out_buf = NULL;
+    }
+    avfilter_unref_buffer(samplesref);
+    inlink->cur_buf = NULL;
+}
+
+void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+    AVFilterPad *dst = link->dstpad;
+
+    FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1);
+
+    if (!(filter_samples = dst->filter_samples))
+        filter_samples = ff_default_filter_samples;
+
+    /* prepare to copy the samples if the buffer has insufficient permissions */
+    if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
+        dst->rej_perms & samplesref->perms) {
+        int  i, planar = av_sample_fmt_is_planar(samplesref->format);
+        int planes = !planar ? 1:
+                     av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);
+
+        av_log(link->dst, AV_LOG_DEBUG,
+               "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
+               samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
+
+        link->cur_buf = ff_default_get_audio_buffer(link, dst->min_perms,
+                                                    samplesref->audio->nb_samples);
+        link->cur_buf->pts                = samplesref->pts;
+        link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;
+
+        /* Copy actual data into new samples buffer */
+        for (i = 0; i < planes; i++)
+            memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], samplesref->linesize[0]);
+
+        avfilter_unref_buffer(samplesref);
+    } else
+        link->cur_buf = samplesref;
+
+    filter_samples(link, link->cur_buf);
+}
+
diff --git a/libavfilter/avfilter.c b/libavfilter/avfilter.c
index 6a530f8..bd898e3 100644
--- a/libavfilter/avfilter.c
+++ b/libavfilter/avfilter.c
@@ -27,7 +27,6 @@
 #include "libavutil/imgutils.h"
 #include "libavcodec/avcodec.h"
 
-#include "audio.h"
 #include "avfilter.h"
 #include "internal.h"
 
@@ -277,7 +276,7 @@ static void ff_dlog_ref(void *ctx, AVFilterBufferRef *ref, int end)
     av_dlog(ctx, "]%s", end ? "\n" : "");
 }
 
-static void ff_dlog_link(void *ctx, AVFilterLink *link, int end)
+void ff_dlog_link(void *ctx, AVFilterLink *link, int end)
 {
     if (link->type == AVMEDIA_TYPE_VIDEO) {
         av_dlog(ctx,
@@ -301,8 +300,6 @@ static void ff_dlog_link(void *ctx, AVFilterLink *link, int end)
     }
 }
 
-#define FF_DPRINTF_START(ctx, func) av_dlog(NULL, "%-16s: ", #func)
-
 AVFilterBufferRef *avfilter_get_video_buffer(AVFilterLink *link, int perms, int w, int h)
 {
     AVFilterBufferRef *ret = NULL;
@@ -368,91 +365,6 @@ fail:
     return NULL;
 }
 
-AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
-                                       int nb_samples)
-{
-    AVFilterBufferRef *ret = NULL;
-
-    if (link->dstpad->get_audio_buffer)
-        ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
-
-    if (!ret)
-        ret = ff_default_get_audio_buffer(link, perms, nb_samples);
-
-    if (ret)
-        ret->type = AVMEDIA_TYPE_AUDIO;
-
-    return ret;
-}
-
-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 planes;
-    AVFilterBuffer    *samples    = av_mallocz(sizeof(*samples));
-    AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
-
-    if (!samples || !samplesref)
-        goto fail;
-
-    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->planar         = av_sample_fmt_is_planar(sample_fmt);
-
-    planes = samplesref->audio->planar ? av_get_channel_layout_nb_channels(channel_layout) : 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;
-    }
-
-    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;
-}
-
 int avfilter_request_frame(AVFilterLink *link)
 {
     FF_DPRINTF_START(NULL, request_frame); ff_dlog_link(NULL, link, 1);
@@ -572,43 +484,6 @@ void avfilter_draw_slice(AVFilterLink *link, int y, int h, int slice_dir)
     draw_slice(link, y, h, slice_dir);
 }
 
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
-{
-    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
-    AVFilterPad *dst = link->dstpad;
-
-    FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1);
-
-    if (!(filter_samples = dst->filter_samples))
-        filter_samples = ff_default_filter_samples;
-
-    /* prepare to copy the samples if the buffer has insufficient permissions */
-    if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
-        dst->rej_perms & samplesref->perms) {
-        int  i, planar = av_sample_fmt_is_planar(samplesref->format);
-        int planes = !planar ? 1:
-                     av_get_channel_layout_nb_channels(samplesref->audio->channel_layout);
-
-        av_log(link->dst, AV_LOG_DEBUG,
-               "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
-               samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
-
-        link->cur_buf = ff_default_get_audio_buffer(link, dst->min_perms,
-                                                    samplesref->audio->nb_samples);
-        link->cur_buf->pts                = samplesref->pts;
-        link->cur_buf->audio->sample_rate = samplesref->audio->sample_rate;
-
-        /* Copy actual data into new samples buffer */
-        for (i = 0; i < planes; i++)
-            memcpy(link->cur_buf->extended_data[i], samplesref->extended_data[i], samplesref->linesize[0]);
-
-        avfilter_unref_buffer(samplesref);
-    } else
-        link->cur_buf = samplesref;
-
-    filter_samples(link, link->cur_buf);
-}
-
 #define MAX_REGISTERED_AVFILTERS_NB 64
 
 static AVFilter *registered_avfilters[MAX_REGISTERED_AVFILTERS_NB + 1];
diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c
index c25d37f..caf6442 100644
--- a/libavfilter/defaults.c
+++ b/libavfilter/defaults.c
@@ -23,7 +23,6 @@
 #include "libavutil/imgutils.h"
 #include "libavutil/samplefmt.h"
 
-#include "audio.h"
 #include "avfilter.h"
 #include "internal.h"
 
@@ -59,37 +58,6 @@ AVFilterBufferRef *avfilter_default_get_video_buffer(AVFilterLink *link, int per
     return picref;
 }
 
-AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
-                                               int nb_samples)
-{
-    AVFilterBufferRef *samplesref = NULL;
-    uint8_t **data;
-    int planar      = av_sample_fmt_is_planar(link->format);
-    int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
-    int planes      = planar ? nb_channels : 1;
-    int linesize;
-
-    if (!(data = av_mallocz(sizeof(*data) * planes)))
-        goto fail;
-
-    if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
-        goto fail;
-
-    samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
-                                                           nb_samples, link->format,
-                                                           link->channel_layout);
-    if (!samplesref)
-        goto fail;
-
-    av_freep(&data);
-
-fail:
-    if (data)
-        av_freep(&data[0]);
-    av_freep(&data);
-    return samplesref;
-}
-
 void avfilter_default_start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
 {
     AVFilterLink *outlink = NULL;
@@ -134,27 +102,6 @@ void avfilter_default_end_frame(AVFilterLink *inlink)
     }
 }
 
-/* FIXME: samplesref is same as link->cur_buf. Need to consider removing the redundant parameter. */
-void ff_default_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
-{
-    AVFilterLink *outlink = NULL;
-
-    if (inlink->dst->output_count)
-        outlink = inlink->dst->outputs[0];
-
-    if (outlink) {
-        outlink->out_buf = ff_default_get_audio_buffer(inlink, AV_PERM_WRITE,
-                                                       samplesref->audio->nb_samples);
-        outlink->out_buf->pts                = samplesref->pts;
-        outlink->out_buf->audio->sample_rate = samplesref->audio->sample_rate;
-        ff_filter_samples(outlink, avfilter_ref_buffer(outlink->out_buf, ~0));
-        avfilter_unref_buffer(outlink->out_buf);
-        outlink->out_buf = NULL;
-    }
-    avfilter_unref_buffer(samplesref);
-    inlink->cur_buf = NULL;
-}
-
 /**
  * default config_link() implementation for output video links to simplify
  * the implementation of one input one output video filters */
@@ -235,18 +182,7 @@ void avfilter_null_end_frame(AVFilterLink *link)
     avfilter_end_frame(link->dst->outputs[0]);
 }
 
-void ff_null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
-{
-    ff_filter_samples(link->dst->outputs[0], samplesref);
-}
-
 AVFilterBufferRef *avfilter_null_get_video_buffer(AVFilterLink *link, int perms, int w, int h)
 {
     return avfilter_get_video_buffer(link->dst->outputs[0], perms, w, h);
 }
-
-AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
-                                            int nb_samples)
-{
-    return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
-}
diff --git a/libavfilter/internal.h b/libavfilter/internal.h
index 0630e9b..a5b3f78 100644
--- a/libavfilter/internal.h
+++ b/libavfilter/internal.h
@@ -55,4 +55,8 @@ void ff_avfilter_default_free_buffer(AVFilterBuffer *buf);
 /** Tell is a format is contained in the provided list terminated by -1. */
 int ff_fmt_is_in(int fmt, const int *fmts);
 
+#define FF_DPRINTF_START(ctx, func) av_dlog(NULL, "%-16s: ", #func)
+
+void ff_dlog_link(void *ctx, AVFilterLink *link, int end);
+
 #endif /* AVFILTER_INTERNAL_H */



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