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-rw-r--r--libavcodec/libopus_dec.c221
1 files changed, 221 insertions, 0 deletions
diff --git a/libavcodec/libopus_dec.c b/libavcodec/libopus_dec.c
new file mode 100644
index 0000000000..a5c6b41385
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+++ b/libavcodec/libopus_dec.c
@@ -0,0 +1,221 @@
+/*
+ * 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"),
+};