Merge remote-tracking branch 'qatar/master'

* qatar/master: (25 commits)
  rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC
  ape: Use unsigned integer maths
  arm: dsputil: fix overreads in put/avg_pixels functions
  h264: K&R formatting cosmetics for header files (part II/II)
  h264: K&R formatting cosmetics for header files (part I/II)
  rtmp: Implement check bandwidth notification.
  rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player.
  rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin.
  rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream.
  cmdutils: Add fallback case to switch in check_stream_specifier().
  sctp: be consistent with socket option level
  configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags.
  vcr1enc: drop pointless empty encode_init() wrapper function
  vcr1: drop pointless write-only AVCodecContext member from VCR1Context
  vcr1: group encoder code together to save #ifdefs
  vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments
  mov: make one comment slightly more specific
  lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX
  lavfi: move audio-related functions to a separate file.
  lavfi: remove some audio-related function from public API.
  ...

Conflicts:
	cmdutils.c
	libavcodec/h264.h
	libavcodec/h264_mvpred.h
	libavcodec/vcr1.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/defaults.c
	libavfilter/internal.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 291 insertions, 0 deletions

diff --git a/libavfilter/audio.c b/libavfilter/audio.c
new file mode 100644
index 0000000000..31f6796437
--- /dev/null
+++ b/libavfilter/audio.c
@@ -0,0 +1,291 @@
+/*
+ * 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/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;
+    int linesize[8] = {0};
+    uint8_t *data[8] = {0};
+    int ch, nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+
+    /* right now we don't support more than 8 channels */
+    av_assert0(nb_channels <= 8);
+
+    /* Calculate total buffer size, round to multiple of 16 to be SIMD friendly */
+    if (av_samples_alloc(data, linesize,
+                         nb_channels, nb_samples,
+                         av_get_alt_sample_fmt(link->format, link->planar),
+                         16) < 0)
+        return NULL;
+
+    for (ch = 1; link->planar && ch < nb_channels; ch++)
+        linesize[ch] = linesize[0];
+    samplesref =
+        avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
+                                                  nb_samples, link->format,
+                                                  link->channel_layout, link->planar);
+    if (!samplesref) {
+        av_free(data[0]);
+        return NULL;
+    }
+
+    return samplesref;
+}
+
+static AVFilterBufferRef *ff_default_get_audio_buffer_alt(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_alt(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[8], int linesize[8], int perms,
+                                          int nb_samples, enum AVSampleFormat sample_fmt,
+                                          uint64_t channel_layout, int planar)
+{
+    AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer));
+    AVFilterBufferRef *samplesref = av_mallocz(sizeof(AVFilterBufferRef));
+
+    if (!samples || !samplesref)
+        goto fail;
+
+    samplesref->buf = samples;
+    samplesref->buf->free = ff_avfilter_default_free_buffer;
+    if (!(samplesref->audio = av_mallocz(sizeof(AVFilterBufferRefAudioProps))))
+        goto fail;
+
+    samplesref->audio->nb_samples     = nb_samples;
+    samplesref->audio->channel_layout = channel_layout;
+    samplesref->audio->planar         = planar;
+
+    /* 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,     sizeof(samples->data));
+    memcpy(samples->linesize,    linesize, sizeof(samples->linesize));
+    memcpy(samplesref->data,     data,     sizeof(samplesref->data));
+    memcpy(samplesref->linesize, linesize, sizeof(samplesref->linesize));
+
+    return samplesref;
+
+fail:
+    if (samplesref && samplesref->audio)
+        av_freep(&samplesref->audio);
+    av_freep(&samplesref);
+    av_freep(&samples);
+    return NULL;
+}
+
+AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_alt(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;
+    int64_t pts;
+
+    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; samplesref->data[i] && i < 8; i++)
+            memcpy(link->cur_buf->data[i], samplesref->data[i], samplesref->linesize[0]);
+        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;
+
+    pts = link->cur_buf->pts;
+    filter_samples(link, link->cur_buf);
+    ff_update_link_current_pts(link, pts);
+}