From fc9cf0b2a6a0bd3933fcef216860c594b767834e Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Wed, 1 Feb 2012 21:21:24 -0500 Subject: alacenc: pretty-printing and other cosmetics --- libavcodec/alacenc.c | 135 ++++++++++++++++++++++++--------------------------- 1 file changed, 64 insertions(+), 71 deletions(-) (limited to 'libavcodec/alacenc.c') diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 934541e39f..9e632aeca1 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -119,12 +119,12 @@ static void encode_scalar(AlacEncodeContext *s, int x, static void write_frame_header(AlacEncodeContext *s, int is_verbatim) { - put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 - put_bits(&s->pbctx, 16, 0); // Seems to be zero - put_bits(&s->pbctx, 1, 1); // Sample count is in the header - put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field - put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim - put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame + put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 + put_bits(&s->pbctx, 16, 0); // Seems to be zero + put_bits(&s->pbctx, 1, 1); // Sample count is in the header + put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field + put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim + put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame } static void calc_predictor_params(AlacEncodeContext *s, int ch) @@ -167,8 +167,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* calculate sum of 2nd order residual for each channel */ sum[0] = sum[1] = sum[2] = sum[3] = 0; for (i = 2; i < n; i++) { - lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2]; - rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2]; + lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2]; + rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2]; sum[2] += FFABS((lt + rt) >> 1); sum[3] += FFABS(lt - rt); sum[0] += FFABS(lt); @@ -184,9 +184,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) /* return mode with lowest score */ best = 0; for (i = 1; i < 4; i++) { - if (score[i] < score[best]) { + if (score[i] < score[best]) best = i; - } } return best; } @@ -199,40 +198,35 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) mode = estimate_stereo_mode(left, right, n); - switch(mode) - { - case ALAC_CHMODE_LEFT_RIGHT: - s->interlacing_leftweight = 0; - s->interlacing_shift = 0; - break; - - case ALAC_CHMODE_LEFT_SIDE: - for (i = 0; i < n; i++) { - right[i] = left[i] - right[i]; - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 0; - break; - - case ALAC_CHMODE_RIGHT_SIDE: - for (i = 0; i < n; i++) { - tmp = right[i]; - right[i] = left[i] - right[i]; - left[i] = tmp + (right[i] >> 31); - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 31; - break; - - default: - for (i = 0; i < n; i++) { - tmp = left[i]; - left[i] = (tmp + right[i]) >> 1; - right[i] = tmp - right[i]; - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 1; - break; + switch (mode) { + case ALAC_CHMODE_LEFT_RIGHT: + s->interlacing_leftweight = 0; + s->interlacing_shift = 0; + break; + case ALAC_CHMODE_LEFT_SIDE: + for (i = 0; i < n; i++) + right[i] = left[i] - right[i]; + s->interlacing_leftweight = 1; + s->interlacing_shift = 0; + break; + case ALAC_CHMODE_RIGHT_SIDE: + for (i = 0; i < n; i++) { + tmp = right[i]; + right[i] = left[i] - right[i]; + left[i] = tmp + (right[i] >> 31); + } + s->interlacing_leftweight = 1; + s->interlacing_shift = 31; + break; + default: + for (i = 0; i < n; i++) { + tmp = left[i]; + left[i] = (tmp + right[i]) >> 1; + right[i] = tmp - right[i]; + } + s->interlacing_leftweight = 1; + s->interlacing_shift = 1; + break; } } @@ -244,8 +238,10 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) if (lpc.lpc_order == 31) { s->predictor_buf[0] = s->sample_buf[ch][0]; - for (i = 1; i < s->avctx->frame_size; i++) - s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; + for (i = 1; i < s->avctx->frame_size; i++) { + s->predictor_buf[i] = s->sample_buf[ch][i ] - + s->sample_buf[ch][i - 1]; + } return; } @@ -267,7 +263,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) for (j = 0; j < lpc.lpc_order; j++) { sum += (samples[lpc.lpc_order-j] - samples[0]) * - lpc.lpc_coeff[j]; + lpc.lpc_coeff[j]; } sum >>= lpc.lpc_quant; @@ -276,21 +272,20 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) s->write_sample_size); res_val = residual[i]; - if(res_val) { + if (res_val) { int index = lpc.lpc_order - 1; int neg = (res_val < 0); - while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { - int val = samples[0] - samples[lpc.lpc_order - index]; + while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) { + int val = samples[0] - samples[lpc.lpc_order - index]; int sign = (val ? FFSIGN(val) : 0); - if(neg) - sign*=-1; + if (neg) + sign *= -1; lpc.lpc_coeff[index] -= sign; val *= sign; - res_val -= ((val >> lpc.lpc_quant) * - (lpc.lpc_order - index)); + res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index); index--; } } @@ -310,16 +305,16 @@ static void alac_entropy_coder(AlacEncodeContext *s) k = av_log2((history >> 9) + 3); - x = -2*(*samples)-1; - x ^= (x>>31); + x = -2 * (*samples) -1; + x ^= x >> 31; samples++; i++; encode_scalar(s, x - sign_modifier, k, s->write_sample_size); - history += x * s->rc.history_mult - - ((history * s->rc.history_mult) >> 9); + history += x * s->rc.history_mult - + ((history * s->rc.history_mult) >> 9); sign_modifier = 0; if (x > 0xFFFF) @@ -336,9 +331,7 @@ static void alac_entropy_coder(AlacEncodeContext *s) block_size++; } encode_scalar(s, block_size, k, 16); - sign_modifier = (block_size <= 0xFFFF); - history = 0; } @@ -356,7 +349,6 @@ static void write_compressed_frame(AlacEncodeContext *s) put_bits(&s->pbctx, 8, s->interlacing_leftweight); for (i = 0; i < s->avctx->channels; i++) { - calc_predictor_params(s, i); put_bits(&s->pbctx, 4, prediction_type); @@ -365,9 +357,8 @@ static void write_compressed_frame(AlacEncodeContext *s) put_bits(&s->pbctx, 3, s->rc.rice_modifier); put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); // predictor coeff. table - for (j = 0; j < s->lpc[i].lpc_order; j++) { + for (j = 0; j < s->lpc[i].lpc_order; j++) put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); - } } // apply lpc and entropy coding to audio samples @@ -398,11 +389,11 @@ static av_cold int alac_encode_close(AVCodecContext *avctx) static av_cold int alac_encode_init(AVCodecContext *avctx) { - AlacEncodeContext *s = avctx->priv_data; + AlacEncodeContext *s = avctx->priv_data; int ret; uint8_t *alac_extradata; - avctx->frame_size = DEFAULT_FRAME_SIZE; + avctx->frame_size = DEFAULT_FRAME_SIZE; if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); @@ -429,9 +420,11 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) s->rc.k_modifier = 14; s->rc.rice_modifier = 4; - s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels * DEFAULT_SAMPLE_SIZE >> 3); + s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels * + DEFAULT_SAMPLE_SIZE >> 3); - s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes + // FIXME: consider wasted_bytes + s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) { @@ -566,8 +559,8 @@ AVCodec ff_alac_encoder = { .init = alac_encode_init, .encode = alac_encode_frame, .close = alac_encode_close, - .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, - AV_SAMPLE_FMT_NONE }, - .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), + .capabilities = CODEC_CAP_SMALL_LAST_FRAME, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; -- cgit v1.2.3