Add libavresample

This is a new library for audio sample format, channel layout, and sample rate
conversion.

1 files changed, 89 insertions, 0 deletions

diff --git a/libavresample/options.c b/libavresample/options.c
new file mode 100644
index 0000000000..0be1a26117
--- /dev/null
+++ b/libavresample/options.c
@@ -0,0 +1,89 @@
+/*
+ * 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 "libavutil/mathematics.h"
+#include "libavutil/opt.h"
+#include "avresample.h"
+#include "internal.h"
+#include "audio_mix.h"
+
+/**
+ * @file
+ * Options definition for AVAudioResampleContext.
+ */
+
+#define OFFSET(x) offsetof(AVAudioResampleContext, x)
+#define PARAM AV_OPT_FLAG_AUDIO_PARAM
+
+static const AVOption options[] = {
+    { "in_channel_layout",      "Input Channel Layout",     OFFSET(in_channel_layout),      AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
+    { "in_sample_fmt",          "Input Sample Format",      OFFSET(in_sample_fmt),          AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
+    { "in_sample_rate",         "Input Sample Rate",        OFFSET(in_sample_rate),         AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
+    { "out_channel_layout",     "Output Channel Layout",    OFFSET(out_channel_layout),     AV_OPT_TYPE_INT64,  { 0                     }, INT64_MIN,            INT64_MAX,              PARAM },
+    { "out_sample_fmt",         "Output Sample Format",     OFFSET(out_sample_fmt),         AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_S16     }, AV_SAMPLE_FMT_U8,     AV_SAMPLE_FMT_NB-1,     PARAM },
+    { "out_sample_rate",        "Output Sample Rate",       OFFSET(out_sample_rate),        AV_OPT_TYPE_INT,    { 48000                 }, 1,                    INT_MAX,                PARAM },
+    { "internal_sample_fmt",    "Internal Sample Format",   OFFSET(internal_sample_fmt),    AV_OPT_TYPE_INT,    { AV_SAMPLE_FMT_FLTP    }, AV_SAMPLE_FMT_NONE,   AV_SAMPLE_FMT_NB-1,     PARAM },
+    { "mix_coeff_type",         "Mixing Coefficient Type",  OFFSET(mix_coeff_type),         AV_OPT_TYPE_INT,    { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q6, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
+        { "q6",  "16-bit 10.6 Fixed-Point",  0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q6  }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
+        { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
+        { "flt", "Floating-Point",           0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
+    { "center_mix_level",       "Center Mix Level",         OFFSET(center_mix_level),       AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
+    { "surround_mix_level",     "Surround Mix Level",       OFFSET(surround_mix_level),     AV_OPT_TYPE_DOUBLE, { M_SQRT1_2             }, -32.0,                32.0,                   PARAM },
+    { "lfe_mix_level",          "LFE Mix Level",            OFFSET(lfe_mix_level),          AV_OPT_TYPE_DOUBLE, { 0.0                   }, -32.0,                32.0,                   PARAM },
+    { "force_resampling",       "Force Resampling",         OFFSET(force_resampling),       AV_OPT_TYPE_INT,    { 0                     }, 0,                    1,                      PARAM },
+    { "filter_size",            "Resampling Filter Size",   OFFSET(filter_size),            AV_OPT_TYPE_INT,    { 16                    }, 0,                    32, /* ??? */           PARAM },
+    { "phase_shift",            "Resampling Phase Shift",   OFFSET(phase_shift),            AV_OPT_TYPE_INT,    { 10                    }, 0,                    30, /* ??? */           PARAM },
+    { "linear_interp",          "Use Linear Interpolation", OFFSET(linear_interp),          AV_OPT_TYPE_INT,    { 0                     }, 0,                    1,                      PARAM },
+    { "cutoff",                 "Cutoff Frequency Ratio",   OFFSET(cutoff),                 AV_OPT_TYPE_DOUBLE, { 0.8                   }, 0.0,                  1.0,                    PARAM },
+    { NULL },
+};
+
+static const AVClass av_resample_context_class = {
+    .class_name = "AVAudioResampleContext",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVAudioResampleContext *avresample_alloc_context(void)
+{
+    AVAudioResampleContext *avr;
+
+    avr = av_mallocz(sizeof(*avr));
+    if (!avr)
+        return NULL;
+
+    avr->av_class = &av_resample_context_class;
+    av_opt_set_defaults(avr);
+
+    avr->am = av_mallocz(sizeof(*avr->am));
+    if (!avr->am) {
+        av_free(avr);
+        return NULL;
+    }
+    avr->am->avr = avr;
+
+    return avr;
+}
+
+const AVClass *avresample_get_class(void)
+{
+    return &av_resample_context_class;
+}