Merge remote-tracking branch 'qatar/master'

* qatar/master:
  libopus: Remap channels using libopus' internal remapping.
  Opus decoder using libopus
  avcodec: document the use of AVCodecContext.delay for audio decoding
  vc1dec: add flush function for WMV9 and VC-1 decoders
  http: Increase buffer sizes to cope with longer URIs
  nutenc: const correctness for ff_put_v_trace/put_s_trace function arguments
  h264_refs: Fix debug tprintf argument types
  golomb: const correctness for get_ue()/get_se() function arguments
  get_bits: const correctness for get_bits_trace()/get_xbits_trace() arguments

Conflicts:
	Changelog
	libavcodec/Makefile
	libavcodec/version.h
	libavformat/http.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 216 insertions, 0 deletions

diff --git a/libavcodec/libopusdec.c b/libavcodec/libopusdec.c
new file mode 100644
index 0000000000..24488fe9dd
--- /dev/null
+++ b/libavcodec/libopusdec.c
@@ -0,0 +1,216 @@
+/*
+ * 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/common.h"
+#include "libavutil/avassert.h"
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "vorbis.h"
+#include "mathops.h"
+
+struct libopus_context {
+    OpusMSDecoder *dec;
+    AVFrame frame;
+    int pre_skip;
+#ifndef OPUS_SET_GAIN
+    union { int i; double d; } gain;
+#endif
+};
+
+static int 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);
+    }
+}
+
+#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 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 */
+    avcodec_get_frame_defaults(&opus->frame);
+    avc->coded_frame = &opus->frame;
+    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 *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;
+    }
+
+    if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
+        nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
+                                             (opus_int16 *)opus->frame.data[0],
+                                             opus->frame.nb_samples, 0);
+    else
+        nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
+                                                   (float *)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 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 *)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_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 },
+};