aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/libopus_dec.c
blob: a5c6b413853ddd90d39b61e384d300b63af520a2 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
/*
 * Opus decoder using libopus
 * Copyright (c) 2012 Nicolas George
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <opus.h>
#include <opus_multistream.h>
#include "avcodec.h"
#include "internal.h"
#include "vorbis.h"
#include "libavutil/avassert.h"
#include "libavutil/intreadwrite.h"

struct libopus_context {
    OpusMSDecoder *dec;
    AVFrame frame;
    int pre_skip;
#ifndef OPUS_SET_GAIN
    union { int i; double d; } gain;
#endif
};

static int ff_opus_error_to_averror(int err)
{
    switch (err) {
        case OPUS_BAD_ARG:          return AVERROR(EINVAL);
        case OPUS_BUFFER_TOO_SMALL: return AVERROR_BUFFER_TOO_SMALL;
        case OPUS_INTERNAL_ERROR:   return AVERROR(EFAULT);
        case OPUS_INVALID_PACKET:   return AVERROR_INVALIDDATA;
        case OPUS_UNIMPLEMENTED:    return AVERROR(ENOSYS);
        case OPUS_INVALID_STATE:    return AVERROR_EXTERNAL;
        case OPUS_ALLOC_FAIL:       return AVERROR(ENOMEM);
        default:                    return AVERROR(EINVAL);
    }
}

static inline void reorder(uint8_t *data, unsigned channels, unsigned bps,
                           unsigned samples, const uint8_t *map)
{
    uint8_t tmp[8 * 4];
    unsigned i;

    av_assert1(channels * bps <= sizeof(tmp));
    for (; samples > 0; samples--) {
        for (i = 0; i < channels; i++)
            memcpy(tmp + bps * i, data + bps * map[i], bps);
        memcpy(data, tmp, bps * channels);
        data += bps * channels;
    }
}

#define OPUS_HEAD_SIZE 19

static av_cold int libopus_dec_init(AVCodecContext *avc)
{
    struct libopus_context *opus = avc->priv_data;
    int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
    uint8_t mapping_stereo[] = { 0, 1 }, *mapping;

    avc->sample_rate = 48000;
    avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
                      AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
    avc->channel_layout = avc->channels > 8 ? 0 :
                          ff_vorbis_channel_layouts[avc->channels - 1];

    if (avc->extradata_size >= OPUS_HEAD_SIZE) {
        opus->pre_skip = AV_RL16(avc->extradata + 10);
        gain_db        = AV_RL16(avc->extradata + 16);
        channel_map    = AV_RL8 (avc->extradata + 18);
        gain_db -= (gain_db & 0x8000) << 1; /* signed */
    }
    if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
        nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0];
        nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
        if (nb_streams + nb_coupled != avc->channels)
            av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
        mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
    } else {
        if (avc->channels > 2 || channel_map) {
            av_log(avc, AV_LOG_ERROR,
                   "No channel mapping for %d channels.\n", avc->channels);
            return AVERROR(EINVAL);
        }
        nb_streams = 1;
        nb_coupled = avc->channels > 1;
        mapping = mapping_stereo;
    }

    opus->dec = opus_multistream_decoder_create(
        avc->sample_rate, avc->channels,
        nb_streams, nb_coupled, mapping, &ret);
    if (!opus->dec) {
        av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
               opus_strerror(ret));
        return ff_opus_error_to_averror(ret);
    }

#ifdef OPUS_SET_GAIN
    ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
    if (ret != OPUS_OK)
        av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
               opus_strerror(ret));
#else
    {
        double gain_lin = pow(10, gain_db / (20.0 * 256));
        if (avc->sample_fmt == AV_SAMPLE_FMT_FLT)
            opus->gain.d = gain_lin;
        else
            opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX);
    }
#endif

    avc->internal->skip_samples = opus->pre_skip;
    avcodec_get_frame_defaults(&opus->frame);
    avc->coded_frame = &opus->frame;
    return 0;
}

static av_cold int libopus_dec_close(AVCodecContext *avc)
{
    struct libopus_context *opus = avc->priv_data;

    opus_multistream_decoder_destroy(opus->dec);
    return 0;
}

#define MAX_FRAME_SIZE (960*6)

static int libopus_dec_decode(AVCodecContext *avc, void *frame,
                              int *got_frame_ptr, AVPacket *pkt)
{
    struct libopus_context *opus = avc->priv_data;
    int ret, nb_samples;

    opus->frame.nb_samples = MAX_FRAME_SIZE;
    ret = avc->get_buffer(avc, &opus->frame);
    if (ret < 0) {
        av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }

    nb_samples = avc->sample_fmt == AV_SAMPLE_FMT_S16 ?
                 opus_multistream_decode      (opus->dec, pkt->data, pkt->size,
                                               (void *)opus->frame.data[0],
                                               opus->frame.nb_samples, 0) :
                 opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
                                               (void *)opus->frame.data[0],
                                               opus->frame.nb_samples, 0);
    if (nb_samples < 0) {
        av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
               opus_strerror(nb_samples));
        return ff_opus_error_to_averror(nb_samples);
    }

    if (avc->channels > 3 && avc->channels <= 8) {
        const uint8_t *m = ff_vorbis_channel_layout_offsets[avc->channels - 1];
        if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
            reorder(opus->frame.data[0], avc->channels, 2, nb_samples, m);
        else
            reorder(opus->frame.data[0], avc->channels, 4, nb_samples, m);
    }

#ifndef OPUS_SET_GAIN
    {
        int i = avc->channels * nb_samples;
        if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
            float *pcm = (float *)opus->frame.data[0];
            for (; i > 0; i--, pcm++)
                *pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
        } else {
            int16_t *pcm = (int16_t *)opus->frame.data[0];
            for (; i > 0; i--, pcm++)
                *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
        }
    }
#endif

    opus->frame.nb_samples = nb_samples;
    *(AVFrame *)frame = opus->frame;
    *got_frame_ptr = 1;
    return pkt->size;
}

static void libopus_dec_flush(AVCodecContext *avc)
{
    struct libopus_context *opus = avc->priv_data;

    opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE);
    /* The stream can have been extracted by a tool that is not Opus-aware.
       Therefore, any packet can become the first of the stream. */
    avc->internal->skip_samples = opus->pre_skip;
}

AVCodec ff_libopus_decoder = {
    .name           = "libopus",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_OPUS,
    .priv_data_size = sizeof(struct libopus_context),
    .init           = libopus_dec_init,
    .close          = libopus_dec_close,
    .decode         = libopus_dec_decode,
    .flush          = libopus_dec_flush,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("libopus Opus"),
};