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https://gitee.com/openharmony/third_party_ffmpeg
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Simplify the FLAC decoder now that it no longer has to parse arbitrary raw FLAC
data thanks to the recently added FLAC parser. Patch by Michael Chinen [mchinen at gmail] Originally committed as revision 25915 to svn://svn.ffmpeg.org/ffmpeg/trunk
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475ae04a27
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@ -61,10 +61,6 @@ typedef struct FLACContext {
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int got_streaminfo; ///< indicates if the STREAMINFO has been read
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int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
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uint8_t *bitstream;
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unsigned int bitstream_size;
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unsigned int bitstream_index;
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unsigned int allocated_bitstream_size;
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} FLACContext;
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static void allocate_buffers(FLACContext *s);
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@ -140,20 +136,10 @@ static void allocate_buffers(FLACContext *s)
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assert(s->max_blocksize);
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if (s->max_framesize == 0 && s->max_blocksize) {
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s->max_framesize = ff_flac_get_max_frame_size(s->max_blocksize,
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s->channels, s->bps);
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}
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for (i = 0; i < s->channels; i++) {
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s->decoded[i] = av_realloc(s->decoded[i],
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sizeof(int32_t)*s->max_blocksize);
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}
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if (s->allocated_bitstream_size < s->max_framesize)
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s->bitstream= av_fast_realloc(s->bitstream,
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&s->allocated_bitstream_size,
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s->max_framesize);
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}
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void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
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@ -563,7 +549,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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FLACContext *s = avctx->priv_data;
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int i, j = 0, input_buf_size = 0, bytes_read = 0;
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int i, j = 0, bytes_read = 0;
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int16_t *samples_16 = data;
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int32_t *samples_32 = data;
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int alloc_data_size= *data_size;
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@ -575,41 +561,13 @@ static int flac_decode_frame(AVCodecContext *avctx,
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s->max_framesize =
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ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
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FLAC_MAX_CHANNELS, 32);
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s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
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}
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if (1 && s->max_framesize) { //FIXME truncated
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if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
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buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
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input_buf_size= buf_size;
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if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
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return -1;
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if (s->allocated_bitstream_size < s->bitstream_size + buf_size)
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s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
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if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) {
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memmove(s->bitstream, &s->bitstream[s->bitstream_index],
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s->bitstream_size);
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s->bitstream_index=0;
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}
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memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size],
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buf, buf_size);
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buf= &s->bitstream[s->bitstream_index];
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buf_size += s->bitstream_size;
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s->bitstream_size= buf_size;
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if (buf_size < s->max_framesize && input_buf_size) {
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return input_buf_size;
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}
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}
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/* check that there is at least the smallest decodable amount of data.
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this amount corresponds to the smallest valid FLAC frame possible.
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FF F8 69 02 00 00 9A 00 00 34 46 */
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if (buf_size < 11)
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goto end;
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return buf_size;
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/* check for inline header */
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if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
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@ -617,26 +575,13 @@ static int flac_decode_frame(AVCodecContext *avctx,
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av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
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return -1;
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}
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bytes_read = get_metadata_size(buf, buf_size);
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goto end;
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}
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/* check for frame sync code and resync stream if necessary */
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if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
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const uint8_t *buf_end = buf + buf_size;
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av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
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while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8)
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buf++;
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bytes_read = buf_size - (buf_end - buf);
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goto end; // we may not have enough bits left to decode a frame, so try next time
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return get_metadata_size(buf, buf_size);
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}
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/* decode frame */
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init_get_bits(&s->gb, buf, buf_size*8);
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if (decode_frame(s) < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
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s->bitstream_size=0;
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s->bitstream_index=0;
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return -1;
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}
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bytes_read = (get_bits_count(&s->gb)+7)/8;
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@ -646,7 +591,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
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if (output_size > alloc_data_size) {
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av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
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"allocated data size\n");
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goto end;
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return -1;
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}
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*data_size = output_size;
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@ -684,20 +629,16 @@ static int flac_decode_frame(AVCodecContext *avctx,
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DECORRELATE( (a-=b>>1) + b, a)
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}
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end:
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if (bytes_read > buf_size) {
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av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
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s->bitstream_size=0;
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s->bitstream_index=0;
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return -1;
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}
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if (bytes_read < buf_size) {
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av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
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buf_size - bytes_read, buf_size);
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}
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if (s->bitstream_size) {
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s->bitstream_index += bytes_read;
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s->bitstream_size -= bytes_read;
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return input_buf_size;
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} else
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return bytes_read;
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return bytes_read;
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}
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static av_cold int flac_decode_close(AVCodecContext *avctx)
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@ -708,19 +649,10 @@ static av_cold int flac_decode_close(AVCodecContext *avctx)
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for (i = 0; i < s->channels; i++) {
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av_freep(&s->decoded[i]);
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}
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av_freep(&s->bitstream);
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return 0;
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}
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static void flac_flush(AVCodecContext *avctx)
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{
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FLACContext *s = avctx->priv_data;
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s->bitstream_size=
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s->bitstream_index= 0;
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}
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AVCodec flac_decoder = {
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"flac",
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AVMEDIA_TYPE_AUDIO,
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@ -730,9 +662,5 @@ AVCodec flac_decoder = {
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NULL,
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flac_decode_close,
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flac_decode_frame,
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CODEC_CAP_DELAY | CODEC_CAP_SUBFRAMES, /* FIXME: add a FLAC parser so that
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we will not need to use either
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of these capabilities */
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.flush= flac_flush,
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.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
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};
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