diff options
author | Nathan Adil Maxson <nathan0n5ire@gmail.com> | 2011-12-01 09:37:33 +0400 |
---|---|---|
committer | Ronald S. Bultje <rsbultje@gmail.com> | 2011-12-02 08:14:21 +0400 |
commit | d0fd6fc20130ef514df294727bf21d01ccbf588c (patch) | |
tree | 71312103087b0979ec934fd6e30d961c36f2242d /libavcodec | |
parent | 04403ec2e405a3cfcfbdd45f1274be30c652e462 (diff) |
Cleaned up alacenc.c
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/alacenc.c | 101 |
1 files changed, 54 insertions, 47 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index fe03bb7dad..7e29a24c6f 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -75,20 +75,22 @@ typedef struct AlacEncodeContext { } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples) +static void init_sample_buffers(AlacEncodeContext *s, + const int16_t *input_samples) { int ch, i; - for(ch=0;ch<s->avctx->channels;ch++) { + for (ch = 0; ch < s->avctx->channels; ch++) { const int16_t *sptr = input_samples + ch; - for(i=0;i<s->avctx->frame_size;i++) { + for (i = 0; i < s->avctx->frame_size; i++) { s->sample_buf[ch][i] = *sptr; sptr += s->avctx->channels; } } } -static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) +static void encode_scalar(AlacEncodeContext *s, int x, + int k, int write_sample_size) { int divisor, q, r; @@ -97,17 +99,17 @@ static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_s q = x / divisor; r = x % divisor; - if(q > 8) { + if (q > 8) { // write escape code and sample value directly put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); put_bits(&s->pbctx, write_sample_size, x); } else { - if(q) + if (q) put_bits(&s->pbctx, q, (1<<q) - 1); put_bits(&s->pbctx, 1, 0); - if(k != 1) { - if(r > 0) + if (k != 1) { + if (r > 0) put_bits(&s->pbctx, k, r+1); else put_bits(&s->pbctx, k-1, 0); @@ -164,7 +166,7 @@ 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++) { + 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]; sum[2] += FFABS((lt + rt) >> 1); @@ -181,8 +183,8 @@ 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]) { + for (i = 1; i < 4; i++) { + if (score[i] < score[best]) { best = i; } } @@ -205,7 +207,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; case ALAC_CHMODE_LEFT_SIDE: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { right[i] = left[i] - right[i]; } s->interlacing_leftweight = 1; @@ -213,7 +215,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; case ALAC_CHMODE_RIGHT_SIDE: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { tmp = right[i]; right[i] = left[i] - right[i]; left[i] = tmp + (right[i] >> 31); @@ -223,7 +225,7 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s) break; default: - for(i=0; i<n; i++) { + for (i = 0; i < n; i++) { tmp = left[i]; left[i] = (tmp + right[i]) >> 1; right[i] = tmp - right[i]; @@ -239,10 +241,10 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) int i; AlacLPCContext lpc = s->lpc[ch]; - if(lpc.lpc_order == 31) { + if (lpc.lpc_order == 31) { s->predictor_buf[0] = s->sample_buf[ch][0]; - for(i=1; i<s->avctx->frame_size; i++) + 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; @@ -250,17 +252,17 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) // generalised linear predictor - if(lpc.lpc_order > 0) { + if (lpc.lpc_order > 0) { int32_t *samples = s->sample_buf[ch]; int32_t *residual = s->predictor_buf; // generate warm-up samples residual[0] = samples[0]; - for(i=1;i<=lpc.lpc_order;i++) + for (i = 1; i <= lpc.lpc_order; i++) residual[i] = samples[i] - samples[i-1]; // perform lpc on remaining samples - for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { + for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) { int sum = 1 << (lpc.lpc_quant - 1), res_val, j; for (j = 0; j < lpc.lpc_order; j++) { @@ -303,7 +305,7 @@ static void alac_entropy_coder(AlacEncodeContext *s) int sign_modifier = 0, i, k; int32_t *samples = s->predictor_buf; - for(i=0;i < s->avctx->frame_size;) { + for (i = 0; i < s->avctx->frame_size;) { int x; k = av_log2((history >> 9) + 3); @@ -320,15 +322,15 @@ static void alac_entropy_coder(AlacEncodeContext *s) - ((history * s->rc.history_mult) >> 9); sign_modifier = 0; - if(x > 0xFFFF) + if (x > 0xFFFF) history = 0xFFFF; - if((history < 128) && (i < s->avctx->frame_size)) { + if (history < 128 && i < s->avctx->frame_size) { unsigned int block_size = 0; k = 7 - av_log2(history) + ((history + 16) >> 6); - while((*samples == 0) && (i < s->avctx->frame_size)) { + while (*samples == 0 && i < s->avctx->frame_size) { samples++; i++; block_size++; @@ -347,12 +349,12 @@ static void write_compressed_frame(AlacEncodeContext *s) { int i, j; - if(s->avctx->channels == 2) + if (s->avctx->channels == 2) alac_stereo_decorrelation(s); put_bits(&s->pbctx, 8, s->interlacing_shift); put_bits(&s->pbctx, 8, s->interlacing_leftweight); - for(i=0;i<s->avctx->channels;i++) { + for (i = 0; i < s->avctx->channels; i++) { calc_predictor_params(s, i); @@ -362,14 +364,14 @@ 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 - for(i=0;i<s->avctx->channels;i++) { + for (i = 0; i < s->avctx->channels; i++) { alac_linear_predictor(s, i); alac_entropy_coder(s); } @@ -384,13 +386,13 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->frame_size = DEFAULT_FRAME_SIZE; avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE; - if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) { + if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); return -1; } // Set default compression level - if(avctx->compression_level == FF_COMPRESSION_DEFAULT) + if (avctx->compression_level == FF_COMPRESSION_DEFAULT) s->compression_level = 2; else s->compression_level = av_clip(avctx->compression_level, 0, 2); @@ -411,21 +413,23 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample); AV_WB8 (alac_extradata+21, avctx->channels); AV_WB32(alac_extradata+24, s->max_coded_frame_size); - AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate + AV_WB32(alac_extradata+28, + avctx->sample_rate * avctx->channels * avctx->bits_per_coded_sample); // average bitrate AV_WB32(alac_extradata+32, avctx->sample_rate); // Set relevant extradata fields - if(s->compression_level > 0) { + if (s->compression_level > 0) { AV_WB8(alac_extradata+18, s->rc.history_mult); AV_WB8(alac_extradata+19, s->rc.initial_history); AV_WB8(alac_extradata+20, s->rc.k_modifier); } s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; - if(avctx->min_prediction_order >= 0) { - if(avctx->min_prediction_order < MIN_LPC_ORDER || + if (avctx->min_prediction_order >= 0) { + if (avctx->min_prediction_order < MIN_LPC_ORDER || avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); + av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", + avctx->min_prediction_order); return -1; } @@ -433,18 +437,20 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) } s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; - if(avctx->max_prediction_order >= 0) { - if(avctx->max_prediction_order < MIN_LPC_ORDER || - avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order); + if (avctx->max_prediction_order >= 0) { + if (avctx->max_prediction_order < MIN_LPC_ORDER || + avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { + av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", + avctx->max_prediction_order); return -1; } s->max_prediction_order = avctx->max_prediction_order; } - if(s->max_prediction_order < s->min_prediction_order) { - av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", + if (s->max_prediction_order < s->min_prediction_order) { + av_log(avctx, AV_LOG_ERROR, + "invalid prediction orders: min=%d max=%d\n", s->min_prediction_order, s->max_prediction_order); return -1; } @@ -469,12 +475,12 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, PutBitContext *pb = &s->pbctx; int i, out_bytes, verbatim_flag = 0; - if(avctx->frame_size > DEFAULT_FRAME_SIZE) { + if (avctx->frame_size > DEFAULT_FRAME_SIZE) { av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n"); return -1; } - if(buf_size < 2*s->max_coded_frame_size) { + if (buf_size < 2 * s->max_coded_frame_size) { av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); return -1; } @@ -482,11 +488,11 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, verbatim: init_put_bits(pb, frame, buf_size); - if((s->compression_level == 0) || verbatim_flag) { + if (s->compression_level == 0 || verbatim_flag) { // Verbatim mode const int16_t *samples = data; write_frame_header(s, 1); - for(i=0; i<avctx->frame_size*avctx->channels; i++) { + for (i = 0; i < avctx->frame_size * avctx->channels; i++) { put_sbits(pb, 16, *samples++); } } else { @@ -499,9 +505,9 @@ verbatim: flush_put_bits(pb); out_bytes = put_bits_count(pb) >> 3; - if(out_bytes > s->max_coded_frame_size) { + if (out_bytes > s->max_coded_frame_size) { /* frame too large. use verbatim mode */ - if(verbatim_flag || (s->compression_level == 0)) { + if (verbatim_flag || s->compression_level == 0) { /* still too large. must be an error. */ av_log(avctx, AV_LOG_ERROR, "error encoding frame\n"); return -1; @@ -532,6 +538,7 @@ AVCodec ff_alac_encoder = { .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}, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; |