virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

flacenc: add 24-bit encoding

Browse Source
This commit is contained in:
Justin Ruggles 2012-10-27 00:26:02 -04:00
parent 799e232490
commit 13e1ee6c84
1 changed files with 62 additions and 24 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

86
libavcodec/flacenc.c
View File

@ -92,6 +92,7 @@ typedef struct FlacEncodeContext {
int channels;
int samplerate;
int sr_code[2];
int bps_code;
int max_blocksize;
int min_framesize;
int max_framesize;
@ -128,7 +129,7 @@ static void write_streaminfo(FlacEncodeContext *s, uint8_t *header)
put_bits(&pb, 24, s->max_framesize);
put_bits(&pb, 20, s->samplerate);
put_bits(&pb, 3, s->channels-1);
put_bits(&pb, 5, 15); /* bits per sample - 1 */
put_bits(&pb, 5, s->avctx->bits_per_raw_sample - 1);
/* write 36-bit sample count in 2 put_bits() calls */
put_bits(&pb, 24, (s->sample_count & 0xFFFFFF000LL) >> 12);
put_bits(&pb, 12, s->sample_count & 0x000000FFFLL);
@ -228,8 +229,18 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->avctx = avctx;
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16)
return -1;
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16:
avctx->bits_per_raw_sample = 16;
s->bps_code = 4;
break;
case AV_SAMPLE_FMT_S32:
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;
s->bps_code = 6;
break;
}
if (channels < 1 || channels > FLAC_MAX_CHANNELS)
return -1;
@ -359,7 +370,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
/* 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->channels,
s->avctx->bits_per_raw_sample);
/* initialize MD5 context */
s->md5ctx = av_md5_alloc();
@ -387,7 +399,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON);
ff_dsputil_init(&s->dsp, avctx);
ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, 16);
ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt,
avctx->bits_per_raw_sample);
dprint_compression_options(s);
@ -423,7 +436,7 @@ static void init_frame(FlacEncodeContext *s, int nb_samples)
for (ch = 0; ch < s->channels; ch++) {
frame->subframes[ch].wasted = 0;
frame->subframes[ch].obits = 16;
frame->subframes[ch].obits = s->avctx->bits_per_raw_sample;
}
frame->verbatim_only = 0;
@ -433,15 +446,25 @@ static void init_frame(FlacEncodeContext *s, int nb_samples)
/**
* Copy channel-interleaved input samples into separate subframes.
*/
static void copy_samples(FlacEncodeContext *s, const int16_t *samples)
static void copy_samples(FlacEncodeContext *s, const void *samples)
{
int i, j, ch;
FlacFrame *frame;
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;
frame = &s->frame;
for (i = 0, j = 0; i < frame->blocksize; i++)
for (ch = 0; ch < s->channels; ch++, j++)
frame->subframes[ch].samples[i] = samples[j];
#define COPY_SAMPLES(bits) do { \
const int ## bits ## _t *samples0 = samples; \
frame = &s->frame; \
for (i = 0, j = 0; i < frame->blocksize; i++) \
for (ch = 0; ch < s->channels; ch++, j++) \
frame->subframes[ch].samples[i] = samples0[j] >> shift; \
} while (0)
if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S16)
COPY_SAMPLES(16);
else
COPY_SAMPLES(32);
}
@ -1017,7 +1040,7 @@ static void write_frame_header(FlacEncodeContext *s)
else
put_bits(&s->pb, 4, frame->ch_mode + FLAC_MAX_CHANNELS - 1);
put_bits(&s->pb, 3, 4); /* bits-per-sample code */
put_bits(&s->pb, 3, s->bps_code);
put_bits(&s->pb, 1, 0);
write_utf8(&s->pb, s->frame_count);
@ -1119,23 +1142,38 @@ static int write_frame(FlacEncodeContext *s, AVPacket *avpkt)
}
static int update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
static int update_md5_sum(FlacEncodeContext *s, const void *samples)
{
const uint8_t *buf;
int buf_size = s->frame.blocksize * s->channels * 2;
int buf_size = s->frame.blocksize * s->channels *
((s->avctx->bits_per_raw_sample + 7) / 8);
if (HAVE_BIGENDIAN) {
if (s->avctx->bits_per_raw_sample > 16 || HAVE_BIGENDIAN) {
av_fast_malloc(&s->md5_buffer, &s->md5_buffer_size, buf_size);
if (!s->md5_buffer)
return AVERROR(ENOMEM);
}
buf = (const uint8_t *)samples;
if (s->avctx->bits_per_raw_sample <= 16) {
buf = (const uint8_t *)samples;
#if HAVE_BIGENDIAN
s->dsp.bswap16_buf((uint16_t *)s->md5_buffer,
(const uint16_t *)samples, buf_size / 2);
buf = s->md5_buffer;
s->dsp.bswap16_buf((uint16_t *)s->md5_buffer,
(const uint16_t *)samples, buf_size / 2);
buf = s->md5_buffer;
#endif
} else {
int i;
const int32_t *samples0 = samples;
uint8_t *tmp = s->md5_buffer;
for (i = 0; i < s->frame.blocksize * s->channels; i++) {
int32_t v = samples0[i] >> 8;
*tmp++ = (v ) & 0xFF;
*tmp++ = (v >> 8) & 0xFF;
*tmp++ = (v >> 16) & 0xFF;
}
buf = s->md5_buffer;
}
av_md5_update(s->md5ctx, buf, buf_size);
return 0;
@ -1146,7 +1184,6 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
FlacEncodeContext *s;
const int16_t *samples;
int frame_bytes, out_bytes, ret;
s = avctx->priv_data;
@ -1158,17 +1195,17 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
write_streaminfo(s, avctx->extradata);
return 0;
}
samples = (const int16_t *)frame->data[0];
/* change max_framesize for small final frame */
if (frame->nb_samples < s->frame.blocksize) {
s->max_framesize = ff_flac_get_max_frame_size(frame->nb_samples,
s->channels, 16);
s->channels,
avctx->bits_per_raw_sample);
}
init_frame(s, frame->nb_samples);
copy_samples(s, samples);
copy_samples(s, frame->data[0]);
channel_decorrelation(s);
@ -1196,7 +1233,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
s->frame_count++;
s->sample_count += frame->nb_samples;
if ((ret = update_md5_sum(s, samples)) < 0) {
if ((ret = update_md5_sum(s, frame->data[0])) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error updating MD5 checksum\n");
return ret;
}
@ -1273,6 +1310,7 @@ AVCodec ff_flac_encoder = {
.close = flac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
.priv_class = &flac_encoder_class,