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, 173 insertions, 0 deletions

diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h
new file mode 100644
index 0000000000..4609ebc284
--- /dev/null
+++ b/libavresample/audio_data.h
@@ -0,0 +1,173 @@
+/*
+ * 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
+ */
+
+#ifndef AVRESAMPLE_AUDIO_DATA_H
+#define AVRESAMPLE_AUDIO_DATA_H
+
+#include <stdint.h>
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/log.h"
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+
+/**
+ * Audio buffer used for intermediate storage between conversion phases.
+ */
+typedef struct AudioData {
+    const AVClass *class;               /**< AVClass for logging            */
+    uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers        */
+    uint8_t *buffer;                    /**< data buffer                    */
+    unsigned int buffer_size;           /**< allocated buffer size          */
+    int allocated_samples;              /**< number of samples the buffer can hold */
+    int nb_samples;                     /**< current number of samples      */
+    enum AVSampleFormat sample_fmt;     /**< sample format                  */
+    int channels;                       /**< channel count                  */
+    int allocated_channels;             /**< allocated channel count        */
+    int is_planar;                      /**< sample format is planar        */
+    int planes;                         /**< number of data planes          */
+    int sample_size;                    /**< bytes per sample               */
+    int stride;                         /**< sample byte offset within a plane */
+    int read_only;                      /**< data is read-only              */
+    int allow_realloc;                  /**< realloc is allowed             */
+    int ptr_align;                      /**< minimum data pointer alignment */
+    int samples_align;                  /**< allocated samples alignment    */
+    const char *name;                   /**< name for debug logging         */
+} AudioData;
+
+int ff_audio_data_set_channels(AudioData *a, int channels);
+
+/**
+ * Initialize AudioData using a given source.
+ *
+ * This does not allocate an internal buffer. It only sets the data pointers
+ * and audio parameters.
+ *
+ * @param a               AudioData struct
+ * @param src             source data pointers
+ * @param plane_size      plane size, in bytes.
+ *                        This can be 0 if unknown, but that will lead to
+ *                        optimized functions not being used in many cases,
+ *                        which could slow down some conversions.
+ * @param channels        channel count
+ * @param nb_samples      number of samples in the source data
+ * @param sample_fmt      sample format
+ * @param read_only       indicates if buffer is read only or read/write
+ * @param name            name for debug logging (can be NULL)
+ * @return                0 on success, negative AVERROR value on error
+ */
+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);
+
+/**
+ * Allocate AudioData.
+ *
+ * This allocates an internal buffer and sets audio parameters.
+ *
+ * @param channels        channel count
+ * @param nb_samples      number of samples to allocate space for
+ * @param sample_fmt      sample format
+ * @param name            name for debug logging (can be NULL)
+ * @return                newly allocated AudioData struct, or NULL on error
+ */
+AudioData *ff_audio_data_alloc(int channels, int nb_samples,
+                               enum AVSampleFormat sample_fmt,
+                               const char *name);
+
+/**
+ * Reallocate AudioData.
+ *
+ * The AudioData must have been previously allocated with ff_audio_data_alloc().
+ *
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to allocate space for
+ * @return            0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_realloc(AudioData *a, int nb_samples);
+
+/**
+ * Free AudioData.
+ *
+ * The AudioData must have been previously allocated with ff_audio_data_alloc().
+ *
+ * @param a  AudioData struct
+ */
+void ff_audio_data_free(AudioData **a);
+
+/**
+ * Copy data from one AudioData to another.
+ *
+ * @param out  output AudioData
+ * @param in   input AudioData
+ * @return     0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_copy(AudioData *out, AudioData *in);
+
+/**
+ * Append data from one AudioData to the end of another.
+ *
+ * @param dst         destination AudioData
+ * @param dst_offset  offset, in samples, to start writing, relative to the
+ *                    start of dst
+ * @param src         source AudioData
+ * @param src_offset  offset, in samples, to start copying, relative to the
+ *                    start of the src
+ * @param nb_samples  number of samples to copy
+ * @return            0 on success, negative AVERROR value on error
+ */
+int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
+                          int src_offset, int nb_samples);
+
+/**
+ * Drain samples from the start of the AudioData.
+ *
+ * Remaining samples are shifted to the start of the AudioData.
+ *
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to drain
+ */
+void ff_audio_data_drain(AudioData *a, int nb_samples);
+
+/**
+ * Add samples in AudioData to an AVAudioFifo.
+ *
+ * @param af          Audio FIFO Buffer
+ * @param a           AudioData struct
+ * @param offset      number of samples to skip from the start of the data
+ * @param nb_samples  number of samples to add to the FIFO
+ * @return            number of samples actually added to the FIFO, or
+ *                    negative AVERROR code on error
+ */
+int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
+                              int nb_samples);
+
+/**
+ * Read samples from an AVAudioFifo to AudioData.
+ *
+ * @param af          Audio FIFO Buffer
+ * @param a           AudioData struct
+ * @param nb_samples  number of samples to read from the FIFO
+ * @return            number of samples actually read from the FIFO, or
+ *                    negative AVERROR code on error
+ */
+int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
+
+#endif /* AVRESAMPLE_AUDIO_DATA_H */