-/**
+/*
* FLAC audio encoder
* Copyright (c) 2006 Justin Ruggles <justin.ruggles@gmail.com>
*
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
+#include "internal.h"
#include "lpc.h"
#include "flac.h"
#include "flacdata.h"
if (s->options.max_partition_order < 0)
s->options.max_partition_order = ((int[]){ 2, 2, 3, 3, 3, 8, 8, 8, 8, 8, 8, 8, 8})[level];
- /* set compression option overrides from AVCodecContext */
-#if FF_API_FLAC_GLOBAL_OPTS
- if (avctx->lpc_type > FF_LPC_TYPE_DEFAULT) {
- if (avctx->lpc_type > FF_LPC_TYPE_CHOLESKY) {
- av_log(avctx, AV_LOG_ERROR, "unknown lpc type: %d\n", avctx->lpc_type);
- return -1;
- }
- s->options.lpc_type = avctx->lpc_type;
- if (s->options.lpc_type == FF_LPC_TYPE_CHOLESKY) {
- if (avctx->lpc_passes < 0) {
- // default number of passes for Cholesky
- s->options.lpc_passes = 2;
- } else if (avctx->lpc_passes == 0) {
- av_log(avctx, AV_LOG_ERROR, "invalid number of lpc passes: %d\n",
- avctx->lpc_passes);
- return -1;
- } else {
- s->options.lpc_passes = avctx->lpc_passes;
- }
- }
- }
-#endif
-
if (s->options.lpc_type == FF_LPC_TYPE_NONE) {
s->options.min_prediction_order = 0;
} else if (avctx->min_prediction_order >= 0) {
return -1;
}
-#if FF_API_FLAC_GLOBAL_OPTS
- if (avctx->prediction_order_method >= 0) {
- if (avctx->prediction_order_method > ORDER_METHOD_LOG) {
- av_log(avctx, AV_LOG_ERROR, "invalid prediction order method: %d\n",
- avctx->prediction_order_method);
- return -1;
- }
- s->options.prediction_order_method = avctx->prediction_order_method;
- }
-
- if (avctx->min_partition_order >= 0) {
- if (avctx->min_partition_order > MAX_PARTITION_ORDER) {
- av_log(avctx, AV_LOG_ERROR, "invalid min partition order: %d\n",
- avctx->min_partition_order);
- return -1;
- }
- s->options.min_partition_order = avctx->min_partition_order;
- }
- if (avctx->max_partition_order >= 0) {
- if (avctx->max_partition_order > MAX_PARTITION_ORDER) {
- av_log(avctx, AV_LOG_ERROR, "invalid max partition order: %d\n",
- avctx->max_partition_order);
- return -1;
- }
- s->options.max_partition_order = avctx->max_partition_order;
- }
- if (s->options.max_partition_order < s->options.min_partition_order) {
- av_log(avctx, AV_LOG_ERROR, "invalid partition orders: min=%d max=%d\n",
- s->options.min_partition_order, s->options.max_partition_order);
- return -1;
- }
-#endif
-
if (avctx->frame_size > 0) {
if (avctx->frame_size < FLAC_MIN_BLOCKSIZE ||
avctx->frame_size > FLAC_MAX_BLOCKSIZE) {
}
s->max_blocksize = s->avctx->frame_size;
-#if FF_API_FLAC_GLOBAL_OPTS
- /* set LPC precision */
- if (avctx->lpc_coeff_precision > 0) {
- if (avctx->lpc_coeff_precision > MAX_LPC_PRECISION) {
- av_log(avctx, AV_LOG_ERROR, "invalid lpc coeff precision: %d\n",
- avctx->lpc_coeff_precision);
- return -1;
- }
- s->options.lpc_coeff_precision = avctx->lpc_coeff_precision;
- }
-#endif
-
/* set maximum encoded frame size in verbatim mode */
s->max_framesize = ff_flac_get_max_frame_size(s->avctx->frame_size,
s->channels, 16);
s->frame_count = 0;
s->min_framesize = s->max_framesize;
+#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
+#endif
if (channels == 3 &&
avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
}
-static void init_frame(FlacEncodeContext *s)
+static void init_frame(FlacEncodeContext *s, int nb_samples)
{
int i, ch;
FlacFrame *frame;
frame = &s->frame;
for (i = 0; i < 16; i++) {
- if (s->avctx->frame_size == ff_flac_blocksize_table[i]) {
+ if (nb_samples == ff_flac_blocksize_table[i]) {
frame->blocksize = ff_flac_blocksize_table[i];
frame->bs_code[0] = i;
frame->bs_code[1] = 0;
}
}
if (i == 16) {
- frame->blocksize = s->avctx->frame_size;
+ frame->blocksize = nb_samples;
if (frame->blocksize <= 256) {
frame->bs_code[0] = 6;
frame->bs_code[1] = frame->blocksize-1;
}
-static int write_frame(FlacEncodeContext *s, uint8_t *frame, int buf_size)
+static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
{
- init_put_bits(&s->pb, frame, buf_size);
+ init_put_bits(&s->pb, avpkt->data, avpkt->size);
write_frame_header(s);
write_subframes(s);
write_frame_footer(s);
}
-static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
- int buf_size, void *data)
+static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
FlacEncodeContext *s;
- const int16_t *samples = data;
- int frame_bytes, out_bytes;
+ const int16_t *samples;
+ int frame_bytes, out_bytes, ret;
s = avctx->priv_data;
/* when the last block is reached, update the header in extradata */
- if (!data) {
+ if (!frame) {
s->max_framesize = s->max_encoded_framesize;
av_md5_final(s->md5ctx, s->md5sum);
write_streaminfo(s, avctx->extradata);
return 0;
}
+ samples = (const int16_t *)frame->data[0];
/* change max_framesize for small final frame */
- if (avctx->frame_size < s->frame.blocksize) {
- s->max_framesize = ff_flac_get_max_frame_size(avctx->frame_size,
+ if (frame->nb_samples < s->frame.blocksize) {
+ s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples,
s->channels, 16);
}
- init_frame(s);
+ init_frame(s, frame->nb_samples);
copy_samples(s, samples);
frame_bytes = encode_frame(s);
}
- if (buf_size < frame_bytes) {
- av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
- return 0;
+ if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes))) {
+ return ret;
}
- out_bytes = write_frame(s, frame, buf_size);
+
+ out_bytes = write_frame(s, avpkt);
s->frame_count++;
- avctx->coded_frame->pts = s->sample_count;
- s->sample_count += avctx->frame_size;
+ s->sample_count += frame->nb_samples;
update_md5_sum(s, samples);
if (out_bytes > s->max_encoded_framesize)
s->max_encoded_framesize = out_bytes;
if (out_bytes < s->min_framesize)
s->min_framesize = out_bytes;
- return out_bytes;
+ avpkt->pts = frame->pts;
+ avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
+ avpkt->size = out_bytes;
+ *got_packet_ptr = 1;
+ return 0;
}
}
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
+#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
+#endif
return 0;
}
.id = CODEC_ID_FLAC,
.priv_data_size = sizeof(FlacEncodeContext),
.init = flac_encode_init,
- .encode = flac_encode_frame,
+ .encode2 = flac_encode_frame,
.close = flac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_LOSSLESS,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
projects / ffmpeg / blobdiff