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, 75 insertions, 0 deletions

diff --git a/libavresample/internal.h b/libavresample/internal.h
new file mode 100644
index 0000000000..49ea6a668e
--- /dev/null
+++ b/libavresample/internal.h
@@ -0,0 +1,75 @@
+/*
+ * 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_INTERNAL_H
+#define AVRESAMPLE_INTERNAL_H
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avresample.h"
+#include "audio_convert.h"
+#include "audio_data.h"
+#include "audio_mix.h"
+#include "resample.h"
+
+struct AVAudioResampleContext {
+    const AVClass *av_class;        /**< AVClass for logging and AVOptions  */
+
+    uint64_t in_channel_layout;                 /**< input channel layout   */
+    enum AVSampleFormat in_sample_fmt;          /**< input sample format    */
+    int in_sample_rate;                         /**< input sample rate      */
+    uint64_t out_channel_layout;                /**< output channel layout  */
+    enum AVSampleFormat out_sample_fmt;         /**< output sample format   */
+    int out_sample_rate;                        /**< output sample rate     */
+    enum AVSampleFormat internal_sample_fmt;    /**< internal sample format */
+    enum AVMixCoeffType mix_coeff_type;         /**< mixing coefficient type */
+    double center_mix_level;                    /**< center mix level       */
+    double surround_mix_level;                  /**< surround mix level     */
+    double lfe_mix_level;                       /**< lfe mix level          */
+    int force_resampling;                       /**< force resampling       */
+    int filter_size;                            /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
+    int phase_shift;                            /**< log2 of the number of entries in the resampling polyphase filterbank */
+    int linear_interp;                          /**< if 1 then the resampling FIR filter will be linearly interpolated */
+    double cutoff;                              /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
+
+    int in_channels;        /**< number of input channels                   */
+    int out_channels;       /**< number of output channels                  */
+    int resample_channels;  /**< number of channels used for resampling     */
+    int downmix_needed;     /**< downmixing is needed                       */
+    int upmix_needed;       /**< upmixing is needed                         */
+    int mixing_needed;      /**< either upmixing or downmixing is needed    */
+    int resample_needed;    /**< resampling is needed                       */
+    int in_convert_needed;  /**< input sample format conversion is needed   */
+    int out_convert_needed; /**< output sample format conversion is needed  */
+
+    AudioData *in_buffer;           /**< buffer for converted input         */
+    AudioData *resample_out_buffer; /**< buffer for output from resampler   */
+    AudioData *out_buffer;          /**< buffer for converted output        */
+    AVAudioFifo *out_fifo;          /**< FIFO for output samples            */
+
+    AudioConvert *ac_in;        /**< input sample format conversion context  */
+    AudioConvert *ac_out;       /**< output sample format conversion context */
+    ResampleContext *resample;  /**< resampling context                      */
+    AudioMix *am;               /**< channel mixing context                  */
+};
+
+#endif /* AVRESAMPLE_INTERNAL_H */