2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 * This file is part of Libav.
6 * Libav is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * Libav is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with Libav; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #ifndef AVRESAMPLE_INTERNAL_H
22 #define AVRESAMPLE_INTERNAL_H
24 #include "libavutil/audio_fifo.h"
25 #include "libavutil/log.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/samplefmt.h"
28 #include "avresample.h"
29 #include "audio_convert.h"
30 #include "audio_data.h"
31 #include "audio_mix.h"
34 struct AVAudioResampleContext {
35 const AVClass *av_class; /**< AVClass for logging and AVOptions */
37 uint64_t in_channel_layout; /**< input channel layout */
38 enum AVSampleFormat in_sample_fmt; /**< input sample format */
39 int in_sample_rate; /**< input sample rate */
40 uint64_t out_channel_layout; /**< output channel layout */
41 enum AVSampleFormat out_sample_fmt; /**< output sample format */
42 int out_sample_rate; /**< output sample rate */
43 enum AVSampleFormat internal_sample_fmt; /**< internal sample format */
44 enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */
45 double center_mix_level; /**< center mix level */
46 double surround_mix_level; /**< surround mix level */
47 double lfe_mix_level; /**< lfe mix level */
48 int force_resampling; /**< force resampling */
49 int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
50 int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
51 int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
52 double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
54 int in_channels; /**< number of input channels */
55 int out_channels; /**< number of output channels */
56 int resample_channels; /**< number of channels used for resampling */
57 int downmix_needed; /**< downmixing is needed */
58 int upmix_needed; /**< upmixing is needed */
59 int mixing_needed; /**< either upmixing or downmixing is needed */
60 int resample_needed; /**< resampling is needed */
61 int in_convert_needed; /**< input sample format conversion is needed */
62 int out_convert_needed; /**< output sample format conversion is needed */
64 AudioData *in_buffer; /**< buffer for converted input */
65 AudioData *resample_out_buffer; /**< buffer for output from resampler */
66 AudioData *out_buffer; /**< buffer for converted output */
67 AVAudioFifo *out_fifo; /**< FIFO for output samples */
69 AudioConvert *ac_in; /**< input sample format conversion context */
70 AudioConvert *ac_out; /**< output sample format conversion context */
71 ResampleContext *resample; /**< resampling context */
72 AudioMix *am; /**< channel mixing context */
75 #endif /* AVRESAMPLE_INTERNAL_H */