making act segmenter

1 files changed, 628 insertions, 0 deletions

diff --git a/ffmpeg/libavcodec/alac.c b/ffmpeg/libavcodec/alac.c
new file mode 100644
index 0000000..0018b9a
--- /dev/null
+++ b/ffmpeg/libavcodec/alac.c
@@ -0,0 +1,628 @@
+/*
+ * ALAC (Apple Lossless Audio Codec) decoder
+ * Copyright (c) 2005 David Hammerton
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * ALAC (Apple Lossless Audio Codec) decoder
+ * @author 2005 David Hammerton
+ * @see http://crazney.net/programs/itunes/alac.html
+ *
+ * Note: This decoder expects a 36-byte QuickTime atom to be
+ * passed through the extradata[_size] fields. This atom is tacked onto
+ * the end of an 'alac' stsd atom and has the following format:
+ *
+ * 32bit  atom size
+ * 32bit  tag                  ("alac")
+ * 32bit  tag version          (0)
+ * 32bit  samples per frame    (used when not set explicitly in the frames)
+ *  8bit  compatible version   (0)
+ *  8bit  sample size
+ *  8bit  history mult         (40)
+ *  8bit  initial history      (10)
+ *  8bit  rice param limit     (14)
+ *  8bit  channels
+ * 16bit  maxRun               (255)
+ * 32bit  max coded frame size (0 means unknown)
+ * 32bit  average bitrate      (0 means unknown)
+ * 32bit  samplerate
+ */
+
+#include "libavutil/channel_layout.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "bytestream.h"
+#include "internal.h"
+#include "unary.h"
+#include "mathops.h"
+#include "alac_data.h"
+
+#define ALAC_EXTRADATA_SIZE 36
+
+typedef struct {
+    AVCodecContext *avctx;
+    GetBitContext gb;
+    int channels;
+
+    int32_t *predict_error_buffer[2];
+    int32_t *output_samples_buffer[2];
+    int32_t *extra_bits_buffer[2];
+
+    uint32_t max_samples_per_frame;
+    uint8_t  sample_size;
+    uint8_t  rice_history_mult;
+    uint8_t  rice_initial_history;
+    uint8_t  rice_limit;
+
+    int extra_bits;     /**< number of extra bits beyond 16-bit */
+    int nb_samples;     /**< number of samples in the current frame */
+
+    int direct_output;
+} ALACContext;
+
+static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
+{
+    unsigned int x = get_unary_0_9(gb);
+
+    if (x > 8) { /* RICE THRESHOLD */
+        /* use alternative encoding */
+        x = get_bits_long(gb, bps);
+    } else if (k != 1) {
+        int extrabits = show_bits(gb, k);
+
+        /* multiply x by 2^k - 1, as part of their strange algorithm */
+        x = (x << k) - x;
+
+        if (extrabits > 1) {
+            x += extrabits - 1;
+            skip_bits(gb, k);
+        } else
+            skip_bits(gb, k - 1);
+    }
+    return x;
+}
+
+static int rice_decompress(ALACContext *alac, int32_t *output_buffer,
+                            int nb_samples, int bps, int rice_history_mult)
+{
+    int i;
+    unsigned int history = alac->rice_initial_history;
+    int sign_modifier = 0;
+
+    for (i = 0; i < nb_samples; i++) {
+        int k;
+        unsigned int x;
+
+        if(get_bits_left(&alac->gb) <= 0)
+            return -1;
+
+        /* calculate rice param and decode next value */
+        k = av_log2((history >> 9) + 3);
+        k = FFMIN(k, alac->rice_limit);
+        x = decode_scalar(&alac->gb, k, bps);
+        x += sign_modifier;
+        sign_modifier = 0;
+        output_buffer[i] = (x >> 1) ^ -(x & 1);
+
+        /* update the history */
+        if (x > 0xffff)
+            history = 0xffff;
+        else
+            history +=         x * rice_history_mult -
+                       ((history * rice_history_mult) >> 9);
+
+        /* special case: there may be compressed blocks of 0 */
+        if ((history < 128) && (i + 1 < nb_samples)) {
+            int block_size;
+
+            /* calculate rice param and decode block size */
+            k = 7 - av_log2(history) + ((history + 16) >> 6);
+            k = FFMIN(k, alac->rice_limit);
+            block_size = decode_scalar(&alac->gb, k, 16);
+
+            if (block_size > 0) {
+                if (block_size >= nb_samples - i) {
+                    av_log(alac->avctx, AV_LOG_ERROR,
+                           "invalid zero block size of %d %d %d\n", block_size,
+                           nb_samples, i);
+                    block_size = nb_samples - i - 1;
+                }
+                memset(&output_buffer[i + 1], 0,
+                       block_size * sizeof(*output_buffer));
+                i += block_size;
+            }
+            if (block_size <= 0xffff)
+                sign_modifier = 1;
+            history = 0;
+        }
+    }
+    return 0;
+}
+
+static inline int sign_only(int v)
+{
+    return v ? FFSIGN(v) : 0;
+}
+
+static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
+                           int nb_samples, int bps, int16_t *lpc_coefs,
+                           int lpc_order, int lpc_quant)
+{
+    int i;
+    int32_t *pred = buffer_out;
+
+    /* first sample always copies */
+    *buffer_out = *error_buffer;
+
+    if (nb_samples <= 1)
+        return;
+
+    if (!lpc_order) {
+        memcpy(&buffer_out[1], &error_buffer[1],
+               (nb_samples - 1) * sizeof(*buffer_out));
+        return;
+    }
+
+    if (lpc_order == 31) {
+        /* simple 1st-order prediction */
+        for (i = 1; i < nb_samples; i++) {
+            buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
+                                        bps);
+        }
+        return;
+    }
+
+    /* read warm-up samples */
+    for (i = 1; i <= lpc_order && i < nb_samples; i++)
+        buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
+
+    /* NOTE: 4 and 8 are very common cases that could be optimized. */
+
+    for (; i < nb_samples; i++) {
+        int j;
+        int val = 0;
+        int error_val = error_buffer[i];
+        int error_sign;
+        int d = *pred++;
+
+        /* LPC prediction */
+        for (j = 0; j < lpc_order; j++)
+            val += (pred[j] - d) * lpc_coefs[j];
+        val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
+        val += d + error_val;
+        buffer_out[i] = sign_extend(val, bps);
+
+        /* adapt LPC coefficients */
+        error_sign = sign_only(error_val);
+        if (error_sign) {
+            for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
+                int sign;
+                val  = d - pred[j];
+                sign = sign_only(val) * error_sign;
+                lpc_coefs[j] -= sign;
+                val *= sign;
+                error_val -= (val >> lpc_quant) * (j + 1);
+            }
+        }
+    }
+}
+
+static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
+                               int decorr_shift, int decorr_left_weight)
+{
+    int i;
+
+    for (i = 0; i < nb_samples; i++) {
+        int32_t a, b;
+
+        a = buffer[0][i];
+        b = buffer[1][i];
+
+        a -= (b * decorr_left_weight) >> decorr_shift;
+        b += a;
+
+        buffer[0][i] = b;
+        buffer[1][i] = a;
+    }
+}
+
+static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
+                              int extra_bits, int channels, int nb_samples)
+{
+    int i, ch;
+
+    for (ch = 0; ch < channels; ch++)
+        for (i = 0; i < nb_samples; i++)
+            buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
+}
+
+static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
+                          int channels)
+{
+    ALACContext *alac = avctx->priv_data;
+    int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;
+    uint32_t output_samples;
+    int i, ch;
+
+    skip_bits(&alac->gb, 4);  /* element instance tag */
+    skip_bits(&alac->gb, 12); /* unused header bits */
+
+    /* the number of output samples is stored in the frame */
+    has_size = get_bits1(&alac->gb);
+
+    alac->extra_bits = get_bits(&alac->gb, 2) << 3;
+    bps = alac->sample_size - alac->extra_bits + channels - 1;
+    if (bps > 32U) {
+        av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    /* whether the frame is compressed */
+    is_compressed = !get_bits1(&alac->gb);
+
+    if (has_size)
+        output_samples = get_bits_long(&alac->gb, 32);
+    else
+        output_samples = alac->max_samples_per_frame;
+    if (!output_samples || output_samples > alac->max_samples_per_frame) {
+        av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %d\n",
+               output_samples);
+        return AVERROR_INVALIDDATA;
+    }
+    if (!alac->nb_samples) {
+        /* get output buffer */
+        frame->nb_samples = output_samples;
+        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+            return ret;
+    } else if (output_samples != alac->nb_samples) {
+        av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %u != %d\n",
+               output_samples, alac->nb_samples);
+        return AVERROR_INVALIDDATA;
+    }
+    alac->nb_samples = output_samples;
+    if (alac->direct_output) {
+        for (ch = 0; ch < channels; ch++)
+            alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch];
+    }
+
+    if (is_compressed) {
+        int16_t lpc_coefs[2][32];
+        int lpc_order[2];
+        int prediction_type[2];
+        int lpc_quant[2];
+        int rice_history_mult[2];
+
+        decorr_shift       = get_bits(&alac->gb, 8);
+        decorr_left_weight = get_bits(&alac->gb, 8);
+
+        for (ch = 0; ch < channels; ch++) {
+            prediction_type[ch]   = get_bits(&alac->gb, 4);
+            lpc_quant[ch]         = get_bits(&alac->gb, 4);
+            rice_history_mult[ch] = get_bits(&alac->gb, 3);
+            lpc_order[ch]         = get_bits(&alac->gb, 5);
+
+            /* read the predictor table */
+            for (i = lpc_order[ch] - 1; i >= 0; i--)
+                lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
+        }
+
+        if (alac->extra_bits) {
+            for (i = 0; i < alac->nb_samples; i++) {
+                if(get_bits_left(&alac->gb) <= 0)
+                    return -1;
+                for (ch = 0; ch < channels; ch++)
+                    alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
+            }
+        }
+        for (ch = 0; ch < channels; ch++) {
+            int ret=rice_decompress(alac, alac->predict_error_buffer[ch],
+                            alac->nb_samples, bps,
+                            rice_history_mult[ch] * alac->rice_history_mult / 4);
+            if(ret<0)
+                return ret;
+
+            /* adaptive FIR filter */
+            if (prediction_type[ch] == 15) {
+                /* Prediction type 15 runs the adaptive FIR twice.
+                 * The first pass uses the special-case coef_num = 31, while
+                 * the second pass uses the coefs from the bitstream.
+                 *
+                 * However, this prediction type is not currently used by the
+                 * reference encoder.
+                 */
+                lpc_prediction(alac->predict_error_buffer[ch],
+                               alac->predict_error_buffer[ch],
+                               alac->nb_samples, bps, NULL, 31, 0);
+            } else if (prediction_type[ch] > 0) {
+                av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
+                       prediction_type[ch]);
+            }
+            lpc_prediction(alac->predict_error_buffer[ch],
+                           alac->output_samples_buffer[ch], alac->nb_samples,
+                           bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
+        }
+    } else {
+        /* not compressed, easy case */
+        for (i = 0; i < alac->nb_samples; i++) {
+            if(get_bits_left(&alac->gb) <= 0)
+                return -1;
+            for (ch = 0; ch < channels; ch++) {
+                alac->output_samples_buffer[ch][i] =
+                         get_sbits_long(&alac->gb, alac->sample_size);
+            }
+        }
+        alac->extra_bits   = 0;
+        decorr_shift       = 0;
+        decorr_left_weight = 0;
+    }
+
+    if (channels == 2 && decorr_left_weight) {
+        decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
+                           decorr_shift, decorr_left_weight);
+    }
+
+    if (alac->extra_bits) {
+        append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
+                          alac->extra_bits, channels, alac->nb_samples);
+    }
+
+    if(av_sample_fmt_is_planar(avctx->sample_fmt)) {
+    switch(alac->sample_size) {
+    case 16: {
+        for (ch = 0; ch < channels; ch++) {
+            int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
+            for (i = 0; i < alac->nb_samples; i++)
+                *outbuffer++ = alac->output_samples_buffer[ch][i];
+        }}
+        break;
+    case 24: {
+        for (ch = 0; ch < channels; ch++) {
+            for (i = 0; i < alac->nb_samples; i++)
+                alac->output_samples_buffer[ch][i] <<= 8;
+        }}
+        break;
+    }
+    }else{
+        switch(alac->sample_size) {
+        case 16: {
+            int16_t *outbuffer = ((int16_t *)frame->extended_data[0]) + ch_index;
+            for (i = 0; i < alac->nb_samples; i++) {
+                for (ch = 0; ch < channels; ch++)
+                    *outbuffer++ = alac->output_samples_buffer[ch][i];
+                outbuffer += alac->channels - channels;
+            }
+            }
+            break;
+        case 24: {
+            int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
+            for (i = 0; i < alac->nb_samples; i++) {
+                for (ch = 0; ch < channels; ch++)
+                    *outbuffer++ = alac->output_samples_buffer[ch][i] << 8;
+                outbuffer += alac->channels - channels;
+            }
+            }
+            break;
+        case 32: {
+            int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
+            for (i = 0; i < alac->nb_samples; i++) {
+                for (ch = 0; ch < channels; ch++)
+                    *outbuffer++ = alac->output_samples_buffer[ch][i];
+                outbuffer += alac->channels - channels;
+            }
+            }
+            break;
+        }
+    }
+
+    return 0;
+}
+
+static int alac_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    ALACContext *alac = avctx->priv_data;
+    AVFrame *frame    = data;
+    enum AlacRawDataBlockType element;
+    int channels;
+    int ch, ret, got_end;
+
+    init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8);
+
+    got_end = 0;
+    alac->nb_samples = 0;
+    ch = 0;
+    while (get_bits_left(&alac->gb) >= 3) {
+        element = get_bits(&alac->gb, 3);
+        if (element == TYPE_END) {
+            got_end = 1;
+            break;
+        }
+        if (element > TYPE_CPE && element != TYPE_LFE) {
+            av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d\n", element);
+            return AVERROR_PATCHWELCOME;
+        }
+
+        channels = (element == TYPE_CPE) ? 2 : 1;
+        if (   ch + channels > alac->channels
+            || ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels
+        ) {
+            av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        ret = decode_element(avctx, frame,
+                             ff_alac_channel_layout_offsets[alac->channels - 1][ch],
+                             channels);
+        if (ret < 0 && get_bits_left(&alac->gb))
+            return ret;
+
+        ch += channels;
+    }
+    if (!got_end) {
+        av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
+        av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
+               avpkt->size * 8 - get_bits_count(&alac->gb));
+    }
+
+    *got_frame_ptr = 1;
+
+    return avpkt->size;
+}
+
+static av_cold int alac_decode_close(AVCodecContext *avctx)
+{
+    ALACContext *alac = avctx->priv_data;
+
+    int ch;
+    for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
+        av_freep(&alac->predict_error_buffer[ch]);
+        if (!alac->direct_output)
+            av_freep(&alac->output_samples_buffer[ch]);
+        av_freep(&alac->extra_bits_buffer[ch]);
+    }
+
+    return 0;
+}
+
+static int allocate_buffers(ALACContext *alac)
+{
+    int ch;
+    int buf_size;
+
+    if (alac->max_samples_per_frame > INT_MAX / sizeof(int32_t))
+        goto buf_alloc_fail;
+    buf_size = alac->max_samples_per_frame * sizeof(int32_t);
+
+    for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
+                         buf_size, buf_alloc_fail);
+
+        alac->direct_output = alac->sample_size > 16 && av_sample_fmt_is_planar(alac->avctx->sample_fmt);
+        if (!alac->direct_output) {
+            FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch],
+                             buf_size, buf_alloc_fail);
+        }
+
+        FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
+                         buf_size, buf_alloc_fail);
+    }
+    return 0;
+buf_alloc_fail:
+    alac_decode_close(alac->avctx);
+    return AVERROR(ENOMEM);
+}
+
+static int alac_set_info(ALACContext *alac)
+{
+    GetByteContext gb;
+
+    bytestream2_init(&gb, alac->avctx->extradata,
+                     alac->avctx->extradata_size);
+
+    bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
+
+    alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
+    if (!alac->max_samples_per_frame || alac->max_samples_per_frame > INT_MAX) {
+        av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %u\n",
+               alac->max_samples_per_frame);
+        return AVERROR_INVALIDDATA;
+    }
+    bytestream2_skipu(&gb, 1);  // compatible version
+    alac->sample_size          = bytestream2_get_byteu(&gb);
+    alac->rice_history_mult    = bytestream2_get_byteu(&gb);
+    alac->rice_initial_history = bytestream2_get_byteu(&gb);
+    alac->rice_limit           = bytestream2_get_byteu(&gb);
+    alac->channels             = bytestream2_get_byteu(&gb);
+    bytestream2_get_be16u(&gb); // maxRun
+    bytestream2_get_be32u(&gb); // max coded frame size
+    bytestream2_get_be32u(&gb); // average bitrate
+    bytestream2_get_be32u(&gb); // samplerate
+
+    return 0;
+}
+
+static av_cold int alac_decode_init(AVCodecContext * avctx)
+{
+    int ret;
+    int req_packed;
+    ALACContext *alac = avctx->priv_data;
+    alac->avctx = avctx;
+
+    /* initialize from the extradata */
+    if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) {
+        av_log(avctx, AV_LOG_ERROR, "extradata is too small\n");
+        return AVERROR_INVALIDDATA;
+    }
+    if (alac_set_info(alac)) {
+        av_log(avctx, AV_LOG_ERROR, "set_info failed\n");
+        return -1;
+    }
+
+    req_packed = LIBAVCODEC_VERSION_MAJOR < 55 && !av_sample_fmt_is_planar(avctx->request_sample_fmt);
+    switch (alac->sample_size) {
+    case 16: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P;
+             break;
+    case 24:
+    case 32: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S32P;
+             break;
+    default: avpriv_request_sample(avctx, "Sample depth %d", alac->sample_size);
+             return AVERROR_PATCHWELCOME;
+    }
+    avctx->bits_per_raw_sample = alac->sample_size;
+
+    if (alac->channels < 1) {
+        av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
+        alac->channels = avctx->channels;
+    } else {
+        if (alac->channels > ALAC_MAX_CHANNELS)
+            alac->channels = avctx->channels;
+        else
+            avctx->channels = alac->channels;
+    }
+    if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
+               avctx->channels);
+        return AVERROR_PATCHWELCOME;
+    }
+    avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];
+
+    if ((ret = allocate_buffers(alac)) < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+AVCodec ff_alac_decoder = {
+    .name           = "alac",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ALAC,
+    .priv_data_size = sizeof(ALACContext),
+    .init           = alac_decode_init,
+    .close          = alac_decode_close,
+    .decode         = alac_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+};