X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Falacenc.c;h=8a94f81bb426be4c52e1a6f0655bc9ebe8be0d91;hb=9f1245eb9620a70feaa00ba745c6c7a56a839556;hp=d2a24b145c8b3ed40ad005c80d01284ac3c307ae;hpb=b590f3a7bf9103ac7a7a61c48568676201d6824b;p=ffmpeg diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index d2a24b145c8..8a94f81bb42 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 @@ -59,35 +58,45 @@ typedef struct 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->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, @@ -118,18 +127,21 @@ 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) { int encode_fs = 0; if (s->frame_size < DEFAULT_FRAME_SIZE) encode_fs = 1; - 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, 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, 0); // FIXME: Wasted bytes field - put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim + 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 } @@ -345,43 +357,118 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -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++) { - calc_predictor_params(s, 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; + } + } + } - put_bits(&s->pbctx, 4, prediction_type); - put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); + // apply lpc and entropy coding to audio samples + for (i = 0; i < channels; i++) { + alac_linear_predictor(s, i); - 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]); + // 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); + } } +} - // apply lpc and entropy coding to audio samples - - for (i = 0; i < s->avctx->channels; i++) { - alac_linear_predictor(s, i); - - // 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]; +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++; } - - alac_entropy_coder(s); + element++; } + + put_bits(pb, 3, TYPE_END); + flush_put_bits(pb); + + return put_bits_count(pb) >> 3; } static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) @@ -396,7 +483,6 @@ static av_cold int alac_encode_close(AVCodecContext *avctx) ff_lpc_end(&s->lpc_ctx); av_freep(&avctx->extradata); avctx->extradata_size = 0; - av_freep(&avctx->coded_frame); return 0; } @@ -408,17 +494,13 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) avctx->frame_size = s->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; - } - - /* 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 @@ -435,12 +517,9 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) s->max_coded_frame_size = get_max_frame_size(avctx->frame_size, avctx->channels, - DEFAULT_SAMPLE_SIZE); - - // FIXME: consider wasted_bytes - s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; + avctx->bits_per_raw_sample); - avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); + avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + AV_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) { ret = AVERROR(ENOMEM); goto error; @@ -451,11 +530,11 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) 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 @@ -499,12 +578,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) goto error; } - avctx->coded_frame = avcodec_alloc_frame(); - if (!avctx->coded_frame) { - ret = AVERROR(ENOMEM); - goto error; - } - s->avctx = avctx; if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, @@ -519,71 +592,60 @@ error: 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 max_frame_size; + int out_bytes, max_frame_size, ret; - s->frame_size = avctx->frame_size; + s->frame_size = frame->nb_samples; - if (avctx->frame_size < DEFAULT_FRAME_SIZE) + if (frame->nb_samples < DEFAULT_FRAME_SIZE) max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, - DEFAULT_SAMPLE_SIZE); + avctx->bits_per_raw_sample); else max_frame_size = s->max_coded_frame_size; - if (buf_size < 2 * max_frame_size) { - av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n"); - return AVERROR(EINVAL); + if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; } -verbatim: - init_put_bits(pb, frame, buf_size); - - if (s->compression_level == 0 || verbatim_flag) { - // Verbatim mode - const int16_t *samples = data; - write_frame_header(s, 1); - for (i = 0; i < s->frame_size * avctx->channels; i++) { - put_sbits(pb, 16, *samples++); - } + /* 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 > 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 AVERROR_BUG; - } - verbatim_flag = 1; - goto verbatim; + s->verbatim = 1; + s->extra_bits = 0; + out_bytes = write_frame(s, avpkt, frame->extended_data); } - return out_bytes; + 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, + .capabilities = AV_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 }, - .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), };