From 7c278d2ae410a64bdd89f1777026b4b963c30a1a Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 9 Nov 2012 17:01:09 -0500 Subject: alacenc: support 24-bit encoding --- libavcodec/alacenc.c | 101 ++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 75 insertions(+), 26 deletions(-) (limited to 'libavcodec/alacenc.c') diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index 6b5c4f0069..4d6bf7bd53 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -27,7 +27,6 @@ #include "mathops.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 @@ -66,6 +65,7 @@ typedef struct AlacEncodeContext { int max_prediction_order; int max_coded_frame_size; int write_sample_size; + int extra_bits; int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; int32_t predictor_buf[DEFAULT_FRAME_SIZE]; int interlacing_shift; @@ -78,16 +78,26 @@ typedef struct AlacEncodeContext { } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples) +static void init_sample_buffers(AlacEncodeContext *s, + uint8_t * const *samples) { int ch, i; - - for (ch = 0; ch < s->avctx->channels; ch++) { - int32_t *bptr = s->sample_buf[ch]; - const int16_t *sptr = input_samples[ch]; - for (i = 0; i < s->frame_size; i++) - bptr[i] = sptr[i]; - } + 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 < s->avctx->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, @@ -128,7 +138,7 @@ static void write_frame_header(AlacEncodeContext *s) 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, encode_fs); // Sample count is in the header - put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field + 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 @@ -345,7 +355,8 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples) +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, + uint8_t * const *samples) { int i, j; int prediction_type = 0; @@ -356,9 +367,20 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples) if (s->verbatim) { write_frame_header(s); /* samples are channel-interleaved in verbatim mode */ - for (i = 0; i < s->frame_size; i++) - for (j = 0; j < s->avctx->channels; j++) - put_sbits(pb, 16, samples[j][i]); + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { + int shift = 32 - s->avctx->bits_per_raw_sample; + int32_t * const *samples_s32 = (int32_t * const *)samples; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < s->avctx->channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s32[j][i] >> shift); + } else { + int16_t * const *samples_s16 = (int16_t * const *)samples; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < s->avctx->channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s16[j][i]); + } } else { init_sample_buffers(s, samples); write_frame_header(s); @@ -381,6 +403,17 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples) put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); } + // 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 < s->avctx->channels; j++) { + put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask); + s->sample_buf[j][i] >>= s->extra_bits; + } + } + } + // apply lpc and entropy coding to audio samples for (i = 0; i < s->avctx->channels; i++) { @@ -433,6 +466,15 @@ static av_cold int alac_encode_init(AVCodecContext *avctx) 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 if (avctx->compression_level == FF_COMPRESSION_DEFAULT) s->compression_level = 2; @@ -447,10 +489,7 @@ 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); if (!avctx->extradata) { @@ -463,11 +502,11 @@ 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 @@ -536,13 +575,12 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, { AlacEncodeContext *s = avctx->priv_data; int out_bytes, max_frame_size, ret; - int16_t **samples = (int16_t **)frame->extended_data; s->frame_size = frame->nb_samples; 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; @@ -552,14 +590,24 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, } /* use verbatim mode for compression_level 0 */ - s->verbatim = !s->compression_level; + if (s->compression_level) { + s->verbatim = 0; + s->extra_bits = avctx->bits_per_raw_sample - 16; + } else { + s->verbatim = 1; + s->extra_bits = 0; + } + s->write_sample_size = avctx->bits_per_raw_sample - s->extra_bits + + avctx->channels - 1; - out_bytes = write_frame(s, avpkt, samples); + out_bytes = write_frame(s, avpkt, frame->extended_data); if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ s->verbatim = 1; - out_bytes = write_frame(s, avpkt, samples); + s->extra_bits = 0; + s->write_sample_size = avctx->bits_per_raw_sample + avctx->channels - 1; + out_bytes = write_frame(s, avpkt, frame->extended_data); } avpkt->size = out_bytes; @@ -576,7 +624,8 @@ AVCodec ff_alac_encoder = { .encode2 = alac_encode_frame, .close = alac_encode_close, .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, + .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)"), }; 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