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-rw-r--r--ffmpeg/libavcodec/libopusdec.c198
1 files changed, 198 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/libopusdec.c b/ffmpeg/libavcodec/libopusdec.c
new file mode 100644
index 0000000..9b5cfe7
--- /dev/null
+++ b/ffmpeg/libavcodec/libopusdec.c
@@ -0,0 +1,198 @@
+/*
+ * 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 "libavutil/avassert.h"
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "vorbis.h"
+#include "mathops.h"
+#include "libopus.h"
+
+struct libopus_context {
+ OpusMSDecoder *dec;
+ int pre_skip;
+#ifndef OPUS_SET_GAIN
+ union { int i; double d; } gain;
+#endif
+};
+
+#define OPUS_HEAD_SIZE 19
+
+static av_cold int libopus_decode_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_arr[8] = { 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 = sign_extend(AV_RL16(avc->extradata + 16), 16);
+ channel_map = AV_RL8 (avc->extradata + 18);
+ }
+ 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_arr;
+ }
+
+ if (avc->channels > 2 && avc->channels <= 8) {
+ const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
+ int ch;
+
+ /* Remap channels from vorbis order to ffmpeg order */
+ for (ch = 0; ch < avc->channels; ch++)
+ mapping_arr[ch] = mapping[vorbis_offset[ch]];
+ mapping = mapping_arr;
+ }
+
+ 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;
+ avc->delay = 3840; /* Decoder delay (in samples) at 48kHz */
+
+ return 0;
+}
+
+static av_cold int libopus_decode_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_decode(AVCodecContext *avc, void *data,
+ int *got_frame_ptr, AVPacket *pkt)
+{
+ struct libopus_context *opus = avc->priv_data;
+ AVFrame *frame = data;
+ int ret, nb_samples;
+
+ frame->nb_samples = MAX_FRAME_SIZE;
+ if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
+ return ret;
+
+ if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
+ nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
+ (opus_int16 *)frame->data[0],
+ frame->nb_samples, 0);
+ else
+ nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
+ (float *)frame->data[0],
+ 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);
+ }
+
+#ifndef OPUS_SET_GAIN
+ {
+ int i = avc->channels * nb_samples;
+ if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ float *pcm = (float *)frame->data[0];
+ for (; i > 0; i--, pcm++)
+ *pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
+ } else {
+ int16_t *pcm = (int16_t *)frame->data[0];
+ for (; i > 0; i--, pcm++)
+ *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
+ }
+ }
+#endif
+
+ frame->nb_samples = nb_samples;
+ *got_frame_ptr = 1;
+
+ return pkt->size;
+}
+
+static void libopus_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_decode_init,
+ .close = libopus_decode_close,
+ .decode = libopus_decode,
+ .flush = libopus_flush,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+};
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