mirror of https://git.ffmpeg.org/ffmpeg.git
alacenc: store current frame size in AlacEncodeContext.
This avoids an indirection and will simplify implementation of encode2()
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65d15aec77
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@ -58,6 +58,7 @@ typedef struct AlacLPCContext {
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} AlacLPCContext;
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typedef struct AlacEncodeContext {
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int frame_size; /**< current frame size */
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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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@ -82,7 +83,7 @@ static void init_sample_buffers(AlacEncodeContext *s,
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for (ch = 0; ch < s->avctx->channels; ch++) {
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const int16_t *sptr = input_samples + ch;
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for (i = 0; i < s->avctx->frame_size; i++) {
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for (i = 0; i < s->frame_size; i++) {
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s->sample_buf[ch][i] = *sptr;
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sptr += s->avctx->channels;
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}
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@ -124,7 +125,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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put_bits(&s->pbctx, 1, 1); // 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_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
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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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static void calc_predictor_params(AlacEncodeContext *s, int ch)
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@ -144,7 +145,7 @@ static void calc_predictor_params(AlacEncodeContext *s, int ch)
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s->lpc[ch].lpc_coeff[5] = -25;
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} else {
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opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
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s->avctx->frame_size,
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s->frame_size,
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s->min_prediction_order,
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s->max_prediction_order,
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ALAC_MAX_LPC_PRECISION, coefs, shift,
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@ -193,7 +194,7 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
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static void alac_stereo_decorrelation(AlacEncodeContext *s)
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{
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int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
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int i, mode, n = s->avctx->frame_size;
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int i, mode, n = s->frame_size;
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int32_t tmp;
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mode = estimate_stereo_mode(left, right, n);
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@ -238,7 +239,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
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if (lpc.lpc_order == 31) {
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s->predictor_buf[0] = s->sample_buf[ch][0];
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for (i = 1; i < s->avctx->frame_size; i++) {
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for (i = 1; i < s->frame_size; i++) {
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s->predictor_buf[i] = s->sample_buf[ch][i ] -
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s->sample_buf[ch][i - 1];
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}
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@ -258,7 +259,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
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residual[i] = samples[i] - samples[i-1];
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// perform lpc on remaining samples
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for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
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for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {
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int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
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for (j = 0; j < lpc.lpc_order; j++) {
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@ -300,7 +301,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
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int sign_modifier = 0, i, k;
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int32_t *samples = s->predictor_buf;
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for (i = 0; i < s->avctx->frame_size;) {
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for (i = 0; i < s->frame_size;) {
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int x;
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k = av_log2((history >> 9) + 3);
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@ -320,12 +321,12 @@ static void alac_entropy_coder(AlacEncodeContext *s)
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if (x > 0xFFFF)
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history = 0xFFFF;
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if (history < 128 && i < s->avctx->frame_size) {
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if (history < 128 && i < s->frame_size) {
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unsigned int block_size = 0;
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k = 7 - av_log2(history) + ((history + 16) >> 6);
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while (*samples == 0 && i < s->avctx->frame_size) {
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while (*samples == 0 && i < s->frame_size) {
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samples++;
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i++;
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block_size++;
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@ -369,7 +370,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
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// TODO: determine when this will actually help. for now it's not used.
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if (prediction_type == 15) {
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// 2nd pass 1st order filter
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for (j = s->avctx->frame_size - 1; j > 0; j--)
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for (j = s->frame_size - 1; j > 0; j--)
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s->predictor_buf[j] -= s->predictor_buf[j - 1];
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}
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@ -398,7 +399,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
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int ret;
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uint8_t *alac_extradata;
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avctx->frame_size = DEFAULT_FRAME_SIZE;
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avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
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if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
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av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
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@ -519,8 +520,10 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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int i, out_bytes, verbatim_flag = 0;
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int max_frame_size;
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s->frame_size = avctx->frame_size;
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if (avctx->frame_size < DEFAULT_FRAME_SIZE)
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max_frame_size = get_max_frame_size(avctx->frame_size, avctx->channels,
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max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
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DEFAULT_SAMPLE_SIZE);
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else
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max_frame_size = s->max_coded_frame_size;
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@ -537,7 +540,7 @@ verbatim:
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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 < avctx->frame_size * avctx->channels; i++) {
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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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