Merge remote-tracking branch 'qatar/master'

* qatar/master:
  FATE: use updated reference for aac-latm_stereo_to_51
  avconv: use libavresample
  Add libavresample
  FATE: avoid channel mixing in lavf-dv_fmt

Conflicts:
	Changelog
	Makefile
	cmdutils.c
	configure
	doc/APIchanges
	ffmpeg.c
	tests/lavf-regression.sh
	tests/ref/lavf/dv_fmt
	tests/ref/seek/lavf_dv

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

1 files changed, 345 insertions, 0 deletions

diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c
new file mode 100644
index 0000000000..3f82c50ef0
--- /dev/null
+++ b/libavresample/audio_data.c
@@ -0,0 +1,345 @@
+/*
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * 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 <stdint.h>
+
+#include "libavutil/mem.h"
+#include "audio_data.h"
+
+static const AVClass audio_data_class = {
+    .class_name = "AudioData",
+    .item_name  = av_default_item_name,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+/*
+ * Calculate alignment for data pointers.
+ */
+static void calc_ptr_alignment(AudioData *a)
+{
+    int p;
+    int min_align = 128;
+
+    for (p = 0; p < a->planes; p++) {
+        int cur_align = 128;
+        while ((intptr_t)a->data[p] % cur_align)
+            cur_align >>= 1;
+        if (cur_align < min_align)
+            min_align = cur_align;
+    }
+    a->ptr_align = min_align;
+}
+
+int ff_audio_data_set_channels(AudioData *a, int channels)
+{
+    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
+        channels > a->allocated_channels)
+        return AVERROR(EINVAL);
+
+    a->channels  = channels;
+    a->planes    = a->is_planar ? channels : 1;
+
+    calc_ptr_alignment(a);
+
+    return 0;
+}
+
+int ff_audio_data_init(AudioData *a, void **src, int plane_size, int channels,
+                       int nb_samples, enum AVSampleFormat sample_fmt,
+                       int read_only, const char *name)
+{
+    int p;
+
+    memset(a, 0, sizeof(*a));
+    a->class = &audio_data_class;
+
+    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
+        av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
+        return AVERROR(EINVAL);
+    }
+
+    a->sample_size = av_get_bytes_per_sample(sample_fmt);
+    if (!a->sample_size) {
+        av_log(a, AV_LOG_ERROR, "invalid sample format\n");
+        return AVERROR(EINVAL);
+    }
+    a->is_planar = av_sample_fmt_is_planar(sample_fmt);
+    a->planes    = a->is_planar ? channels : 1;
+    a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
+
+    for (p = 0; p < (a->is_planar ? channels : 1); p++) {
+        if (!src[p]) {
+            av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
+            return AVERROR(EINVAL);
+        }
+        a->data[p] = src[p];
+    }
+    a->allocated_samples  = nb_samples * !read_only;
+    a->nb_samples         = nb_samples;
+    a->sample_fmt         = sample_fmt;
+    a->channels           = channels;
+    a->allocated_channels = channels;
+    a->read_only          = read_only;
+    a->allow_realloc      = 0;
+    a->name               = name ? name : "{no name}";
+
+    calc_ptr_alignment(a);
+    a->samples_align = plane_size / a->stride;
+
+    return 0;
+}
+
+AudioData *ff_audio_data_alloc(int channels, int nb_samples,
+                               enum AVSampleFormat sample_fmt, const char *name)
+{
+    AudioData *a;
+    int ret;
+
+    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
+        return NULL;
+
+    a = av_mallocz(sizeof(*a));
+    if (!a)
+        return NULL;
+
+    a->sample_size = av_get_bytes_per_sample(sample_fmt);
+    if (!a->sample_size) {
+        av_free(a);
+        return NULL;
+    }
+    a->is_planar = av_sample_fmt_is_planar(sample_fmt);
+    a->planes    = a->is_planar ? channels : 1;
+    a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
+
+    a->class              = &audio_data_class;
+    a->sample_fmt         = sample_fmt;
+    a->channels           = channels;
+    a->allocated_channels = channels;
+    a->read_only          = 0;
+    a->allow_realloc      = 1;
+    a->name               = name ? name : "{no name}";
+
+    if (nb_samples > 0) {
+        ret = ff_audio_data_realloc(a, nb_samples);
+        if (ret < 0) {
+            av_free(a);
+            return NULL;
+        }
+        return a;
+    } else {
+        calc_ptr_alignment(a);
+        return a;
+    }
+}
+
+int ff_audio_data_realloc(AudioData *a, int nb_samples)
+{
+    int ret, new_buf_size, plane_size, p;
+
+    /* check if buffer is already large enough */
+    if (a->allocated_samples >= nb_samples)
+        return 0;
+
+    /* validate that the output is not read-only and realloc is allowed */
+    if (a->read_only || !a->allow_realloc)
+        return AVERROR(EINVAL);
+
+    new_buf_size = av_samples_get_buffer_size(&plane_size,
+                                              a->allocated_channels, nb_samples,
+                                              a->sample_fmt, 0);
+    if (new_buf_size < 0)
+        return new_buf_size;
+
+    /* if there is already data in the buffer and the sample format is planar,
+       allocate a new buffer and copy the data, otherwise just realloc the
+       internal buffer and set new data pointers */
+    if (a->nb_samples > 0 && a->is_planar) {
+        uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
+
+        ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
+                               nb_samples, a->sample_fmt, 0);
+        if (ret < 0)
+            return ret;
+
+        for (p = 0; p < a->planes; p++)
+            memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
+
+        av_freep(&a->buffer);
+        memcpy(a->data, new_data, sizeof(new_data));
+        a->buffer = a->data[0];
+    } else {
+        av_freep(&a->buffer);
+        a->buffer = av_malloc(new_buf_size);
+        if (!a->buffer)
+            return AVERROR(ENOMEM);
+        ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
+                                     a->allocated_channels, nb_samples,
+                                     a->sample_fmt, 0);
+        if (ret < 0)
+            return ret;
+    }
+    a->buffer_size       = new_buf_size;
+    a->allocated_samples = nb_samples;
+
+    calc_ptr_alignment(a);
+    a->samples_align = plane_size / a->stride;
+
+    return 0;
+}
+
+void ff_audio_data_free(AudioData **a)
+{
+    if (!*a)
+        return;
+    av_free((*a)->buffer);
+    av_freep(a);
+}
+
+int ff_audio_data_copy(AudioData *dst, AudioData *src)
+{
+    int ret, p;
+
+    /* validate input/output compatibility */
+    if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
+        return AVERROR(EINVAL);
+
+    /* if the input is empty, just empty the output */
+    if (!src->nb_samples) {
+        dst->nb_samples = 0;
+        return 0;
+    }
+
+    /* reallocate output if necessary */
+    ret = ff_audio_data_realloc(dst, src->nb_samples);
+    if (ret < 0)
+        return ret;
+
+    /* copy data */
+    for (p = 0; p < src->planes; p++)
+        memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
+    dst->nb_samples = src->nb_samples;
+
+    return 0;
+}
+
+int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
+                          int src_offset, int nb_samples)
+{
+    int ret, p, dst_offset2, dst_move_size;
+
+    /* validate input/output compatibility */
+    if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
+        av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
+        return AVERROR(EINVAL);
+    }
+
+    /* validate offsets are within the buffer bounds */
+    if (dst_offset < 0 || dst_offset > dst->nb_samples ||
+        src_offset < 0 || src_offset > src->nb_samples) {
+        av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
+               src_offset, dst_offset);
+        return AVERROR(EINVAL);
+    }
+
+    /* check offsets and sizes to see if we can just do nothing and return */
+    if (nb_samples > src->nb_samples - src_offset)
+        nb_samples = src->nb_samples - src_offset;
+    if (nb_samples <= 0)
+        return 0;
+
+    /* validate that the output is not read-only */
+    if (dst->read_only) {
+        av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
+        return AVERROR(EINVAL);
+    }
+
+    /* reallocate output if necessary */
+    ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
+    if (ret < 0) {
+        av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
+        return ret;
+    }
+
+    dst_offset2   = dst_offset + nb_samples;
+    dst_move_size = dst->nb_samples - dst_offset;
+
+    for (p = 0; p < src->planes; p++) {
+        if (dst_move_size > 0) {
+            memmove(dst->data[p] + dst_offset2 * dst->stride,
+                    dst->data[p] + dst_offset  * dst->stride,
+                    dst_move_size * dst->stride);
+        }
+        memcpy(dst->data[p] + dst_offset * dst->stride,
+               src->data[p] + src_offset * src->stride,
+               nb_samples * src->stride);
+    }
+    dst->nb_samples += nb_samples;
+
+    return 0;
+}
+
+void ff_audio_data_drain(AudioData *a, int nb_samples)
+{
+    if (a->nb_samples <= nb_samples) {
+        /* drain the whole buffer */
+        a->nb_samples = 0;
+    } else {
+        int p;
+        int move_offset = a->stride * nb_samples;
+        int move_size   = a->stride * (a->nb_samples - nb_samples);
+
+        for (p = 0; p < a->planes; p++)
+            memmove(a->data[p], a->data[p] + move_offset, move_size);
+
+        a->nb_samples -= nb_samples;
+    }
+}
+
+int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
+                              int nb_samples)
+{
+    uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
+    int offset_size, p;
+
+    if (offset >= a->nb_samples)
+        return 0;
+    offset_size = offset * a->stride;
+    for (p = 0; p < a->planes; p++)
+        offset_data[p] = a->data[p] + offset_size;
+
+    return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
+}
+
+int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
+{
+    int ret;
+
+    if (a->read_only)
+        return AVERROR(EINVAL);
+
+    ret = ff_audio_data_realloc(a, nb_samples);
+    if (ret < 0)
+        return ret;
+
+    ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
+    if (ret >= 0)
+        a->nb_samples = ret;
+    return ret;
+}