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https://github.com/FFmpeg/FFmpeg.git
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alacenc: consolidate bitstream writing into a single function.
Simplifies use of verbatim mode.
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@ -59,6 +59,7 @@ typedef struct AlacLPCContext {
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typedef struct AlacEncodeContext {
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int frame_size; /**< current frame size */
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int verbatim; /**< current frame verbatim mode flag */
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int compression_level;
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int min_prediction_order;
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int max_prediction_order;
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@ -118,7 +119,7 @@ static void encode_scalar(AlacEncodeContext *s, int x,
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}
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}
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static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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static void write_frame_header(AlacEncodeContext *s)
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{
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int encode_fs = 0;
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@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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put_bits(&s->pbctx, 16, 0); // Seems to be zero
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put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
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put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
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put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
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put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
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if (encode_fs)
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put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
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}
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@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s)
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}
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}
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static void write_compressed_frame(AlacEncodeContext *s)
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static int write_frame(AlacEncodeContext *s, uint8_t *data, int size,
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const int16_t *samples)
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{
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int i, j;
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int prediction_type = 0;
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PutBitContext *pb = &s->pbctx;
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init_put_bits(pb, data, size);
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if (s->verbatim) {
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write_frame_header(s);
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for (i = 0; i < s->frame_size * s->avctx->channels; i++)
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put_sbits(pb, 16, *samples++);
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} else {
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init_sample_buffers(s, samples);
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write_frame_header(s);
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if (s->avctx->channels == 2)
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alac_stereo_decorrelation(s);
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put_bits(&s->pbctx, 8, s->interlacing_shift);
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put_bits(&s->pbctx, 8, s->interlacing_leftweight);
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put_bits(pb, 8, s->interlacing_shift);
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put_bits(pb, 8, s->interlacing_leftweight);
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for (i = 0; i < s->avctx->channels; i++) {
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calc_predictor_params(s, i);
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put_bits(&s->pbctx, 4, prediction_type);
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put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
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put_bits(pb, 4, prediction_type);
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put_bits(pb, 4, s->lpc[i].lpc_quant);
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put_bits(&s->pbctx, 3, s->rc.rice_modifier);
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put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
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put_bits(pb, 3, s->rc.rice_modifier);
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put_bits(pb, 5, s->lpc[i].lpc_order);
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// predictor coeff. table
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for (j = 0; j < s->lpc[i].lpc_order; j++)
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put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
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put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
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}
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// apply lpc and entropy coding to audio samples
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@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *s)
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alac_entropy_coder(s);
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}
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}
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put_bits(pb, 3, 7);
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flush_put_bits(pb);
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return put_bits_count(pb) >> 3;
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}
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static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
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@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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int buf_size, void *data)
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{
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AlacEncodeContext *s = avctx->priv_data;
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PutBitContext *pb = &s->pbctx;
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int i, out_bytes, verbatim_flag = 0;
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int max_frame_size;
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int out_bytes, max_frame_size;
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s->frame_size = avctx->frame_size;
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@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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return AVERROR(EINVAL);
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}
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verbatim:
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init_put_bits(pb, frame, buf_size);
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/* use verbatim mode for compression_level 0 */
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s->verbatim = !s->compression_level;
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if (s->compression_level == 0 || verbatim_flag) {
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// Verbatim mode
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const int16_t *samples = data;
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write_frame_header(s, 1);
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for (i = 0; i < s->frame_size * avctx->channels; i++) {
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put_sbits(pb, 16, *samples++);
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}
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} else {
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init_sample_buffers(s, data);
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write_frame_header(s, 0);
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write_compressed_frame(s);
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}
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put_bits(pb, 3, 7);
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flush_put_bits(pb);
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out_bytes = put_bits_count(pb) >> 3;
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out_bytes = write_frame(s, frame, buf_size, data);
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if (out_bytes > max_frame_size) {
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/* frame too large. use verbatim mode */
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if (verbatim_flag || s->compression_level == 0) {
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/* still too large. must be an error. */
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av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
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return AVERROR_BUG;
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}
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verbatim_flag = 1;
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goto verbatim;
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s->verbatim = 1;
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out_bytes = write_frame(s, frame, buf_size, data);
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}
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return out_bytes;
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