X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Falacenc.c;h=401f26f66c34c40244a902a7d5fd1c6d73115706;hb=e61f39849c2e2b7f492c17b42058242ed2fa4d57;hp=9725be8185938d3261eed75a4fe8e7162852766f;hpb=64fe3eaeb351582787cbef75a2fe160253663363;p=ffmpeg diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 9725be81859..401f26f66c3 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -21,13 +21,12 @@ #include "avcodec.h" #include "put_bits.h" -#include "dsputil.h" +#include "internal.h" #include "lpc.h" #include "mathops.h" +#include "alac_data.h" #define DEFAULT_FRAME_SIZE 4096 -#define DEFAULT_SAMPLE_SIZE 16 -#define MAX_CHANNELS 8 #define ALAC_EXTRADATA_SIZE 36 #define ALAC_FRAME_HEADER_SIZE 55 #define ALAC_FRAME_FOOTER_SIZE 3 @@ -58,35 +57,46 @@ typedef struct AlacLPCContext { } AlacLPCContext; typedef struct AlacEncodeContext { + int frame_size; /**< current frame size */ + int verbatim; /**< current frame verbatim mode flag */ int compression_level; int min_prediction_order; int max_prediction_order; int max_coded_frame_size; int write_sample_size; - int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; + int extra_bits; + int32_t sample_buf[2][DEFAULT_FRAME_SIZE]; int32_t predictor_buf[DEFAULT_FRAME_SIZE]; int interlacing_shift; int interlacing_leftweight; PutBitContext pbctx; RiceContext rc; - AlacLPCContext lpc[MAX_CHANNELS]; + AlacLPCContext lpc[2]; LPCContext lpc_ctx; AVCodecContext *avctx; } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, - const int16_t *input_samples) +static void init_sample_buffers(AlacEncodeContext *s, int channels, + uint8_t const *samples[2]) { int ch, i; - - for (ch = 0; ch < s->avctx->channels; ch++) { - const int16_t *sptr = input_samples + ch; - for (i = 0; i < s->avctx->frame_size; i++) { - s->sample_buf[ch][i] = *sptr; - sptr += s->avctx->channels; - } - } + int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 - + s->avctx->bits_per_raw_sample; + +#define COPY_SAMPLES(type) do { \ + for (ch = 0; ch < channels; ch++) { \ + int32_t *bptr = s->sample_buf[ch]; \ + const type *sptr = (const type *)samples[ch]; \ + for (i = 0; i < s->frame_size; i++) \ + bptr[i] = sptr[i] >> shift; \ + } \ + } while (0) + + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) + COPY_SAMPLES(int32_t); + else + COPY_SAMPLES(int16_t); } static void encode_scalar(AlacEncodeContext *s, int x, @@ -117,14 +127,23 @@ static void encode_scalar(AlacEncodeContext *s, int x, } } -static void write_frame_header(AlacEncodeContext *s, int is_verbatim) +static void write_element_header(AlacEncodeContext *s, + enum AlacRawDataBlockType element, + int instance) { - 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 + int encode_fs = 0; + + if (s->frame_size < DEFAULT_FRAME_SIZE) + encode_fs = 1; + + put_bits(&s->pbctx, 3, element); // element type + put_bits(&s->pbctx, 4, instance); // element instance + put_bits(&s->pbctx, 12, 0); // unused header bits + put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header + put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit) + put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim + if (encode_fs) + put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame } static void calc_predictor_params(AlacEncodeContext *s, int ch) @@ -144,7 +163,7 @@ static void calc_predictor_params(AlacEncodeContext *s, int ch) s->lpc[ch].lpc_coeff[5] = -25; } else { opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch], - s->avctx->frame_size, + s->frame_size, s->min_prediction_order, s->max_prediction_order, ALAC_MAX_LPC_PRECISION, coefs, shift, @@ -167,8 +186,8 @@ 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 +203,8 @@ 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; } @@ -194,45 +212,40 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) static void alac_stereo_decorrelation(AlacEncodeContext *s) { int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; - int i, mode, n = s->avctx->frame_size; + int i, mode, n = s->frame_size; int32_t tmp; 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 +257,10 @@ 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->frame_size; i++) { + s->predictor_buf[i] = s->sample_buf[ch][i ] - + s->sample_buf[ch][i - 1]; + } return; } @@ -262,12 +277,12 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch) 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->frame_size; i++) { int sum = 1 << (lpc.lpc_quant - 1), res_val, j; 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 +291,20 @@ 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--; } } @@ -305,106 +319,189 @@ 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->frame_size;) { int x; 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) history = 0xFFFF; - if (history < 128 && i < s->avctx->frame_size) { + if (history < 128 && i < s->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->frame_size) { samples++; i++; block_size++; } encode_scalar(s, block_size, k, 16); - sign_modifier = (block_size <= 0xFFFF); - history = 0; } } } -static void write_compressed_frame(AlacEncodeContext *s) +static void write_element(AlacEncodeContext *s, + enum AlacRawDataBlockType element, int instance, + const uint8_t *samples0, const uint8_t *samples1) { - int i, j; + uint8_t const *samples[2] = { samples0, samples1 }; + int i, j, channels; int prediction_type = 0; + PutBitContext *pb = &s->pbctx; - 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); + channels = element == TYPE_CPE ? 2 : 1; + + if (s->verbatim) { + write_element_header(s, element, instance); + /* samples are channel-interleaved in verbatim mode */ + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { + int shift = 32 - s->avctx->bits_per_raw_sample; + int32_t const *samples_s32[2] = { (const int32_t *)samples0, + (const int32_t *)samples1 }; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s32[j][i] >> shift); + } else { + int16_t const *samples_s16[2] = { (const int16_t *)samples0, + (const int16_t *)samples1 }; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s16[j][i]); + } + } else { + s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits + + channels - 1; + + init_sample_buffers(s, channels, samples); + write_element_header(s, element, instance); + + if (channels == 2) + alac_stereo_decorrelation(s); + else + s->interlacing_shift = s->interlacing_leftweight = 0; + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); + + for (i = 0; i < channels; i++) { + calc_predictor_params(s, i); + + put_bits(pb, 4, prediction_type); + put_bits(pb, 4, s->lpc[i].lpc_quant); + + put_bits(pb, 3, s->rc.rice_modifier); + put_bits(pb, 5, s->lpc[i].lpc_order); + // predictor coeff. table + for (j = 0; j < s->lpc[i].lpc_order; j++) + put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); + } - for (i = 0; i < s->avctx->channels; i++) { + // write extra bits if needed + if (s->extra_bits) { + uint32_t mask = (1 << s->extra_bits) - 1; + for (i = 0; i < s->frame_size; i++) { + for (j = 0; j < channels; j++) { + put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask); + s->sample_buf[j][i] >>= s->extra_bits; + } + } + } - calc_predictor_params(s, i); + // apply lpc and entropy coding to audio samples + for (i = 0; i < channels; i++) { + alac_linear_predictor(s, i); - put_bits(&s->pbctx, 4, prediction_type); - put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); + // TODO: determine when this will actually help. for now it's not used. + if (prediction_type == 15) { + // 2nd pass 1st order filter + for (j = s->frame_size - 1; j > 0; j--) + s->predictor_buf[j] -= s->predictor_buf[j - 1]; + } + alac_entropy_coder(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++) { - put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, + uint8_t * const *samples) +{ + PutBitContext *pb = &s->pbctx; + const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1]; + const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1]; + int ch, element, sce, cpe; + + init_put_bits(pb, avpkt->data, avpkt->size); + + ch = element = sce = cpe = 0; + while (ch < s->avctx->channels) { + if (ch_elements[element] == TYPE_CPE) { + write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]], + samples[ch_map[ch + 1]]); + cpe++; + ch += 2; + } else { + write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL); + sce++; + ch++; } + element++; } - // apply lpc and entropy coding to audio samples + put_bits(pb, 3, TYPE_END); + flush_put_bits(pb); - for (i = 0; i < s->avctx->channels; i++) { - alac_linear_predictor(s, i); + return put_bits_count(pb) >> 3; +} - // TODO: determine when this will actually help. for now it's not used. - if (prediction_type == 15) { - // 2nd pass 1st order filter - for (j = s->avctx->frame_size - 1; j > 0; j--) - s->predictor_buf[j] -= s->predictor_buf[j - 1]; - } +static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) +{ + int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE); + return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8; +} - alac_entropy_coder(s); - } +static av_cold int alac_encode_close(AVCodecContext *avctx) +{ + AlacEncodeContext *s = avctx->priv_data; + ff_lpc_end(&s->lpc_ctx); + av_freep(&avctx->extradata); + avctx->extradata_size = 0; + av_freep(&avctx->coded_frame); + return 0; } 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 = av_mallocz(ALAC_EXTRADATA_SIZE+1); + uint8_t *alac_extradata; - 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"); - return -1; - } + avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE; - /* TODO: Correctly implement multi-channel ALAC. - It is similar to multi-channel AAC, in that it has a series of - single-channel (SCE), channel-pair (CPE), and LFE elements. */ - if (avctx->channels > 2) { - av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n"); - return AVERROR_PATCHWELCOME; + if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { + if (avctx->bits_per_raw_sample != 24) + av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n"); + avctx->bits_per_raw_sample = 24; + } else { + avctx->bits_per_raw_sample = 16; + s->extra_bits = 0; } // Set default compression level @@ -419,18 +516,26 @@ 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 = get_max_frame_size(avctx->frame_size, + avctx->channels, + avctx->bits_per_raw_sample); - s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes + avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); + if (!avctx->extradata) { + ret = AVERROR(ENOMEM); + goto error; + } + avctx->extradata_size = ALAC_EXTRADATA_SIZE; + alac_extradata = avctx->extradata; AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); AV_WB32(alac_extradata+12, avctx->frame_size); - AV_WB8 (alac_extradata+17, DEFAULT_SAMPLE_SIZE); + AV_WB8 (alac_extradata+17, avctx->bits_per_raw_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 * DEFAULT_SAMPLE_SIZE); // average bitrate + avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate AV_WB32(alac_extradata+32, avctx->sample_rate); // Set relevant extradata fields @@ -446,7 +551,8 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order); - return -1; + ret = AVERROR(EINVAL); + goto error; } s->min_prediction_order = avctx->min_prediction_order; @@ -458,7 +564,8 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) 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; + ret = AVERROR(EINVAL); + goto error; } s->max_prediction_order = avctx->max_prediction_order; @@ -468,88 +575,84 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n", s->min_prediction_order, s->max_prediction_order); - return -1; + ret = AVERROR(EINVAL); + goto error; } - avctx->extradata = alac_extradata; - avctx->extradata_size = ALAC_EXTRADATA_SIZE; - - avctx->coded_frame = avcodec_alloc_frame(); - avctx->coded_frame->key_frame = 1; + avctx->coded_frame = av_frame_alloc(); + if (!avctx->coded_frame) { + ret = AVERROR(ENOMEM); + goto error; + } s->avctx = avctx; - ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order, - FF_LPC_TYPE_LEVINSON); + if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, + s->max_prediction_order, + FF_LPC_TYPE_LEVINSON)) < 0) { + goto error; + } + + return 0; +error: + alac_encode_close(avctx); return ret; } -static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame, - int buf_size, void *data) +static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { AlacEncodeContext *s = avctx->priv_data; - PutBitContext *pb = &s->pbctx; - int i, out_bytes, verbatim_flag = 0; + int out_bytes, max_frame_size, ret; - if (buf_size < 2 * s->max_coded_frame_size) { - av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); - return -1; - } + s->frame_size = frame->nb_samples; -verbatim: - init_put_bits(pb, frame, buf_size); + if (frame->nb_samples < DEFAULT_FRAME_SIZE) + max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, + avctx->bits_per_raw_sample); + else + max_frame_size = s->max_coded_frame_size; - 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++) { - put_sbits(pb, 16, *samples++); - } + if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; + } + + /* use verbatim mode for compression_level 0 */ + if (s->compression_level) { + s->verbatim = 0; + s->extra_bits = avctx->bits_per_raw_sample - 16; } else { - init_sample_buffers(s, data); - write_frame_header(s, 0); - write_compressed_frame(s); + s->verbatim = 1; + s->extra_bits = 0; } - put_bits(pb, 3, 7); - flush_put_bits(pb); - out_bytes = put_bits_count(pb) >> 3; + out_bytes = write_frame(s, avpkt, frame->extended_data); - if (out_bytes > s->max_coded_frame_size) { + if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ - 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; - } - verbatim_flag = 1; - goto verbatim; + s->verbatim = 1; + s->extra_bits = 0; + out_bytes = write_frame(s, avpkt, frame->extended_data); } - return out_bytes; -} - -static av_cold int alac_encode_close(AVCodecContext *avctx) -{ - AlacEncodeContext *s = avctx->priv_data; - ff_lpc_end(&s->lpc_ctx); - av_freep(&avctx->extradata); - avctx->extradata_size = 0; - av_freep(&avctx->coded_frame); + avpkt->size = out_bytes; + *got_packet_ptr = 1; return 0; } AVCodec ff_alac_encoder = { .name = "alac", + .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), .type = AVMEDIA_TYPE_AUDIO, - .id = CODEC_ID_ALAC, + .id = AV_CODEC_ID_ALAC, .priv_data_size = sizeof(AlacEncodeContext), .init = alac_encode_init, - .encode = alac_encode_frame, + .encode2 = 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, + .channel_layouts = ff_alac_channel_layouts, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_NONE }, };