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

Merge commit '7c278d2ae410a64bdd89f1777026b4b963c30a1a'

Browse Source 
* commit '7c278d2ae410a64bdd89f1777026b4b963c30a1a':
  alacenc: support 24-bit encoding
  pcmdec: use planar sample format for pcm_s16le_planar
  vorbisdec: use float planar sample format

Conflicts:
	libavcodec/pcm.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-11-20 13:34:27 +01:00
commit 70c0f13a9a
3 changed files with 115 additions and 76 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

99
libavcodec/alacenc.c
View File

@ -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;
int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
s->avctx->bits_per_raw_sample;
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];
}
#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;
@ -550,14 +588,24 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
return ret;
/* 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;
@ -574,7 +622,8 @@ 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)"),
};

11
libavcodec/pcm.c
View File

@ -358,7 +358,16 @@ static int pcm_decode_frame(AVCodecContext *avctx, void *data,
DECODE_PLANAR(16, be16, src, samples, n, 0, 0);
break;
case AV_CODEC_ID_PCM_S16LE_PLANAR:
DECODE_PLANAR(16, le16, src, samples, n, 0, 0);
n /= avctx->channels;
for (c = 0; c < avctx->channels; c++) {
samples = s->frame.extended_data[c];
#if HAVE_BIGENDIAN
DECODE(16, le16, src, samples, n, 0, 0)
#else
memcpy(samples, src, n * 2);
#endif
src += n * 2;
}
break;
case AV_CODEC_ID_PCM_S24LE_PLANAR:
DECODE_PLANAR(32, le24, src, samples, n, 8, 0);

81
libavcodec/vorbisdec.c
View File

@ -153,9 +153,7 @@ typedef struct vorbis_context_s {
uint8_t mode_number; // mode number for the current packet
uint8_t previous_window;
float *channel_residues;
float *channel_floors;
float *saved;
float scale_bias; // for float->int conversion
} vorbis_context;
/* Helper functions */
@ -194,7 +192,6 @@ static void vorbis_free(vorbis_context *vc)
int i;
av_freep(&vc->channel_residues);
av_freep(&vc->channel_floors);
av_freep(&vc->saved);
for (i = 0; i < vc->residue_count; i++)
@ -953,12 +950,11 @@ static int vorbis_parse_id_hdr(vorbis_context *vc)
}
vc->channel_residues = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_residues));
vc->channel_floors = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_floors));
vc->saved = av_mallocz((vc->blocksize[1] / 4) * vc->audio_channels * sizeof(*vc->saved));
vc->previous_window = 0;
ff_mdct_init(&vc->mdct[0], bl0, 1, -vc->scale_bias);
ff_mdct_init(&vc->mdct[1], bl1, 1, -vc->scale_bias);
ff_mdct_init(&vc->mdct[0], bl0, 1, -1.0);
ff_mdct_init(&vc->mdct[1], bl1, 1, -1.0);
av_dlog(NULL, " vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
@ -990,13 +986,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avpriv_float_dsp_init(&vc->fdsp, avccontext->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&vc->fmt_conv, avccontext);
if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
vc->scale_bias = 1.0f;
} else {
avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
vc->scale_bias = 32768.0f;
}
avccontext->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
@ -1487,7 +1477,7 @@ void ff_vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
// Decode the audio packet using the functions above
static int vorbis_parse_audio_packet(vorbis_context *vc)
static int vorbis_parse_audio_packet(vorbis_context *vc, float **floor_ptr)
{
GetBitContext *gb = &vc->gb;
FFTContext *mdct;
@ -1498,7 +1488,6 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
uint8_t do_not_decode[255];
vorbis_mapping *mapping;
float *ch_res_ptr = vc->channel_residues;
float *ch_floor_ptr = vc->channel_floors;
uint8_t res_chan[255];
unsigned res_num = 0;
int retlen = 0;
@ -1530,7 +1519,8 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
}
memset(ch_res_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
memset(ch_floor_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
for (i = 0; i < vc->audio_channels; ++i)
memset(floor_ptr[i], 0, vlen * sizeof(floor_ptr[0][0])); //FIXME can this be removed ?
// Decode floor
@ -1543,14 +1533,13 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
floor = &vc->floors[mapping->submap_floor[0]];
}
ret = floor->decode(vc, &floor->data, ch_floor_ptr);
ret = floor->decode(vc, &floor->data, floor_ptr[i]);
if (ret < 0) {
av_log(vc->avccontext, AV_LOG_ERROR, "Invalid codebook in vorbis_floor_decode.\n");
return AVERROR_INVALIDDATA;
}
no_residue[i] = ret;
ch_floor_ptr += vlen;
}
// Nonzero vector propagate
@ -1614,10 +1603,9 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
mdct = &vc->mdct[blockflag];
for (j = vc->audio_channels-1;j >= 0; j--) {
ch_floor_ptr = vc->channel_floors + j * blocksize / 2;
ch_res_ptr = vc->channel_residues + res_chan[j] * blocksize / 2;
vc->fdsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
vc->fdsp.vector_fmul(floor_ptr[j], floor_ptr[j], ch_res_ptr, blocksize / 2);
mdct->imdct_half(mdct, ch_res_ptr, floor_ptr[j]);
}
// Overlap/add, save data for next overlapping
@ -1628,7 +1616,7 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
unsigned bs1 = vc->blocksize[1];
float *residue = vc->channel_residues + res_chan[j] * blocksize / 2;
float *saved = vc->saved + j * bs1 / 4;
float *ret = vc->channel_floors + j * retlen;
float *ret = floor_ptr[j];
float *buf = residue;
const float *win = vc->win[blockflag & previous_window];
@ -1657,14 +1645,31 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
int buf_size = avpkt->size;
vorbis_context *vc = avccontext->priv_data;
GetBitContext *gb = &vc->gb;
const float *channel_ptrs[255];
float *channel_ptrs[255];
int i, len, ret;
av_dlog(NULL, "packet length %d \n", buf_size);
/* get output buffer */
vc->frame.nb_samples = vc->blocksize[1] / 2;
if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (vc->audio_channels > 8) {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = (float *)vc->frame.extended_data[i];
} else {
for (i = 0; i < vc->audio_channels; i++) {
int ch = ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
channel_ptrs[ch] = (float *)vc->frame.extended_data[i];
}
}
init_get_bits(gb, buf, buf_size*8);
if ((len = vorbis_parse_audio_packet(vc)) <= 0)
if ((len = vorbis_parse_audio_packet(vc, channel_ptrs)) <= 0)
return len;
if (!vc->first_frame) {
@ -1676,30 +1681,7 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n",
get_bits_count(gb) / 8, get_bits_count(gb) % 8, len);
/* get output buffer */
vc->frame.nb_samples = len;
if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (vc->audio_channels > 8) {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = vc->channel_floors + i * len;
} else {
for (i = 0; i < vc->audio_channels; i++)
channel_ptrs[i] = vc->channel_floors +
len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
}
if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
len, vc->audio_channels);
else
vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
channel_ptrs, len,
vc->audio_channels);
*got_frame_ptr = 1;
*(AVFrame *)data = vc->frame;
@ -1740,7 +1722,6 @@ AVCodec ff_vorbis_decoder = {
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};