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Code Issues Releases Activity

Add avcodec_decode_audio4().

Browse Source 
Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
This commit is contained in:
Justin Ruggles 2011-09-06 12:17:45 -04:00
parent 560f773c7d
commit 0eea212943
64 changed files with 1588 additions and 764 deletions
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7
doc/APIchanges
View File

@ -13,6 +13,13 @@ libavutil: 2011-04-18
API changes, most recent first:
2011-xx-xx - xxxxxxx - lavc 53.25.0
Add nb_samples and extended_data fields to AVFrame.
Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4().
avcodec_decode_audio4() writes output samples to an AVFrame, which allows
audio decoders to use get_buffer().
2011-xx-xx - xxxxxxx - lavc 53.24.0
Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump.
Change AVPicture.data[4]/linesize[4] to [8] at next major bump.

38
libavcodec/8svx.c
View File

@ -32,6 +32,7 @@
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
uint8_t fib_acc[2];
const int8_t *table;
@ -83,13 +84,13 @@ static void raw_decode(uint8_t *dst, const int8_t *src, int src_size,
}
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
int buf_size;
uint8_t *out_data = data;
int out_data_size;
uint8_t *out_data;
int ret;
int is_compr = (avctx->codec_id != CODEC_ID_PCM_S8_PLANAR);
/* for the first packet, copy data to buffer */
@ -134,15 +135,18 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_si
/* decode next piece of data from the buffer */
buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
if (buf_size <= 0) {
*data_size = 0;
*got_frame_ptr = 0;
return avpkt->size;
}
out_data_size = buf_size * (is_compr + 1) * avctx->channels;
if (*data_size < out_data_size) {
av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n",
*data_size);
return AVERROR(EINVAL);
/* get output buffer */
esc->frame.nb_samples = buf_size * (is_compr + 1);
if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_data = esc->frame.data[0];
if (is_compr) {
delta_decode(out_data, &esc->data[0][esc->data_idx], buf_size,
&esc->fib_acc[0], esc->table, avctx->channels);
@ -158,7 +162,9 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_si
}
}
esc->data_idx += buf_size;
*data_size = out_data_size;
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
return avpkt->size;
}
@ -186,6 +192,10 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
return -1;
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
avcodec_get_frame_defaults(&esc->frame);
avctx->coded_frame = &esc->frame;
return 0;
}
@ -207,7 +217,7 @@ AVCodec ff_eightsvx_fib_decoder = {
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
@ -219,7 +229,7 @@ AVCodec ff_eightsvx_exp_decoder = {
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
@ -231,6 +241,6 @@ AVCodec ff_pcm_s8_planar_decoder = {
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
};

1
libavcodec/aac.h
View File

@ -251,6 +251,7 @@ typedef struct {
*/
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
MPEG4AudioConfig m4ac;

49
libavcodec/aacdec.c
View File

@ -646,6 +646,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
cbrt_tableinit();
avcodec_get_frame_defaults(&ac->frame);
avctx->coded_frame = &ac->frame;
return 0;
}
@ -2113,12 +2116,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
int *data_size, GetBitContext *gb)
int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id, data_size_tmp;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
@ -2222,24 +2225,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples;
}
data_size_tmp = samples * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
*data_size, data_size_tmp);
return -1;
}
*data_size = data_size_tmp;
if (samples) {
/* get output buffer */
ac->frame.nb_samples = samples;
if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return err;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
else
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
*(AVFrame *)data = ac->frame;
}
*got_frame_ptr = !!samples;
if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
@ -2248,7 +2253,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -2259,7 +2264,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
init_get_bits(&gb, buf, buf_size * 8);
if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@ -2481,8 +2486,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
}
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
AVPacket *avpkt)
static int latm_decode_frame(AVCodecContext *avctx, void *out,
int *got_frame_ptr, AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
@ -2504,7 +2509,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
if (!latmctx->initialized) {
if (!avctx->extradata) {
*out_size = 0;
*got_frame_ptr = 0;
return avpkt->size;
} else {
if ((err = decode_audio_specific_config(
@ -2522,7 +2527,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
return AVERROR_INVALIDDATA;
}
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
return err;
return muxlength;
@ -2552,7 +2557,7 @@ AVCodec ff_aac_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
@ -2573,6 +2578,6 @@ AVCodec ff_aac_latm_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};

32
libavcodec/ac3dec.c
View File

@ -208,6 +208,9 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
}
s->downmixed = 1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -1296,15 +1299,15 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
/**
* Decode a single AC-3 frame.
*/
static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
AVPacket *avpkt)
static int ac3_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
float *out_samples_flt = data;
int16_t *out_samples_s16 = data;
int blk, ch, err;
float *out_samples_flt;
int16_t *out_samples_s16;
int blk, ch, err, ret;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
@ -1321,7 +1324,6 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
init_get_bits(&s->gbc, buf, buf_size * 8);
/* parse the syncinfo */
*data_size = 0;
err = parse_frame_header(s);
if (err) {
@ -1343,6 +1345,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
/* TODO: add support for substreams and dependent frames */
if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) {
av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n");
*got_frame_ptr = 0;
return s->frame_size;
} else {
av_log(avctx, AV_LOG_ERROR, "invalid frame type\n");
@ -1400,6 +1403,15 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
if (s->bitstream_mode == 0x7 && s->channels > 1)
avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
/* get output buffer */
s->frame.nb_samples = s->num_blocks * 256;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_samples_flt = (float *)s->frame.data[0];
out_samples_s16 = (int16_t *)s->frame.data[0];
/* decode the audio blocks */
channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
for (ch = 0; ch < s->out_channels; ch++)
@ -1419,8 +1431,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
out_samples_s16 += 256 * s->out_channels;
}
}
*data_size = s->num_blocks * 256 * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return FFMIN(buf_size, s->frame_size);
}
@ -1458,6 +1472,7 @@ AVCodec ff_ac3_decoder = {
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
@ -1480,6 +1495,7 @@ AVCodec ff_eac3_decoder = {
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE

1
libavcodec/ac3dec.h
View File

@ -68,6 +68,7 @@
typedef struct {
AVClass *class; ///< class for AVOptions
AVCodecContext *avctx; ///< parent context
AVFrame frame; ///< AVFrame for decoded output
GetBitContext gbc; ///< bitstream reader
///@name Bit stream information

42
libavcodec/adpcm.c
View File

@ -84,6 +84,7 @@ static const int swf_index_tables[4][16] = {
/* end of tables */
typedef struct ADPCMDecodeContext {
AVFrame frame;
ADPCMChannelStatus status[6];
} ADPCMDecodeContext;
@ -124,6 +125,10 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
break;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
@ -501,9 +506,8 @@ static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
decode_top_nibble_next = 1; \
}
static int adpcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -514,7 +518,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
const uint8_t *src;
int st; /* stereo */
int count1, count2;
int nb_samples, coded_samples, out_bps, out_size;
int nb_samples, coded_samples, ret;
nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples);
if (nb_samples <= 0) {
@ -522,22 +526,22 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
return AVERROR_INVALIDDATA;
}
out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
out_size = nb_samples * avctx->channels * out_bps;
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = nb_samples;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (short *)c->frame.data[0];
/* use coded_samples when applicable */
/* it is always <= nb_samples, so the output buffer will be large enough */
if (coded_samples) {
if (coded_samples != nb_samples)
av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n");
nb_samples = coded_samples;
out_size = nb_samples * avctx->channels * out_bps;
c->frame.nb_samples = nb_samples = coded_samples;
}
samples = data;
src = buf;
st = avctx->channels == 2 ? 1 : 0;
@ -576,7 +580,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
cs->step_index = 88;
}
samples = (short*)data + channel;
samples = (short *)c->frame.data[0] + channel;
for (m = 0; m < 32; m++) {
*samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3);
@ -628,7 +632,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
for (i = 0; i < avctx->channels; i++) {
samples = (short*)data + i;
samples = (short *)c->frame.data[0] + i;
cs = &c->status[i];
for (n = nb_samples >> 1; n > 0; n--, src++) {
uint8_t v = *src;
@ -965,7 +969,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
}
}
out_size = count * 28 * avctx->channels * out_bps;
c->frame.nb_samples = count * 28;
src = src_end;
break;
}
@ -1144,7 +1148,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
prev[0][i] = (int16_t)bytestream_get_be16(&src);
for (ch = 0; ch <= st; ch++) {
samples = (unsigned short *) data + ch;
samples = (short *)c->frame.data[0] + ch;
/* Read in every sample for this channel. */
for (i = 0; i < nb_samples / 14; i++) {
@ -1177,7 +1181,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
default:
return -1;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return src - buf;
}
@ -1190,6 +1197,7 @@ AVCodec ff_ ## name_ ## _decoder = { \
.priv_data_size = sizeof(ADPCMDecodeContext), \
.init = adpcm_decode_init, \
.decode = adpcm_decode_frame, \
.capabilities = CODEC_CAP_DR1, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
}

1
libavcodec/adx.h
View File

@ -40,6 +40,7 @@ typedef struct {
} ADXChannelState;
typedef struct {
AVFrame frame;
int channels;
ADXChannelState prev[2];
int header_parsed;

41
libavcodec/adxdec.c
View File

@ -50,6 +50,10 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
c->channels = avctx->channels;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
@ -89,36 +93,42 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
return 0;
}
static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int adx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
int buf_size = avpkt->size;
ADXContext *c = avctx->priv_data;
int16_t *samples = data;
int16_t *samples;
const uint8_t *buf = avpkt->data;
int num_blocks, ch;
int num_blocks, ch, ret;
if (c->eof) {
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
/* 18 bytes of data are expanded into 32*2 bytes of audio,
so guard against buffer overflows */
/* calculate number of blocks in the packet */
num_blocks = buf_size / (BLOCK_SIZE * c->channels);
if (num_blocks > *data_size / (BLOCK_SAMPLES * c->channels)) {
buf_size = (*data_size / (BLOCK_SAMPLES * c->channels)) * BLOCK_SIZE;
num_blocks = buf_size / (BLOCK_SIZE * c->channels);
}
if (!buf_size || buf_size % (BLOCK_SIZE * avctx->channels)) {
/* if the packet is not an even multiple of BLOCK_SIZE, check for an EOF
packet */
if (!num_blocks || buf_size % (BLOCK_SIZE * avctx->channels)) {
if (buf_size >= 4 && (AV_RB16(buf) & 0x8000)) {
c->eof = 1;
*data_size = 0;
*got_frame_ptr = 0;
return avpkt->size;
}
return AVERROR_INVALIDDATA;
}
/* get output buffer */
c->frame.nb_samples = num_blocks * BLOCK_SAMPLES;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
while (num_blocks--) {
for (ch = 0; ch < c->channels; ch++) {
if (adx_decode(c, samples + ch, buf, ch)) {
@ -132,7 +142,9 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
samples += BLOCK_SAMPLES * c->channels;
}
*data_size = (uint8_t*)samples - (uint8_t*)data;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return buf - avpkt->data;
}
@ -143,5 +155,6 @@ AVCodec ff_adpcm_adx_decoder = {
.priv_data_size = sizeof(ADXContext),
.init = adx_decode_init,
.decode = adx_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};

45
libavcodec/alac.c
View File

@ -62,10 +62,10 @@
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
GetBitContext gb;
int numchannels;
int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
@ -351,9 +351,8 @@ static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
}
}
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
AVPacket *avpkt)
static int alac_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
@ -366,7 +365,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
int i, ch;
int i, ch, ret;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
@ -401,14 +400,17 @@ static int alac_decode_frame(AVCodecContext *avctx,
} else
outputsamples = alac->setinfo_max_samples_per_frame;
alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
return -1;
/* get output buffer */
if (outputsamples > INT32_MAX) {
av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples);
return AVERROR_INVALIDDATA;
}
alac->frame.nb_samples = outputsamples;
if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
@ -501,21 +503,23 @@ static int alac_decode_frame(AVCodecContext *avctx,
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
interleave_stereo_16(alac->outputsamples_buffer, outbuffer,
outputsamples);
interleave_stereo_16(alac->outputsamples_buffer,
(int16_t *)alac->frame.data[0], outputsamples);
} else {
int16_t *outbuffer = (int16_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++) {
((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
outbuffer[i] = alac->outputsamples_buffer[0][i];
}
}
break;
case 24:
if (channels == 2) {
interleave_stereo_24(alac->outputsamples_buffer, outbuffer,
outputsamples);
interleave_stereo_24(alac->outputsamples_buffer,
(int32_t *)alac->frame.data[0], outputsamples);
} else {
int32_t *outbuffer = (int32_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++)
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
@ -523,6 +527,9 @@ static int alac_decode_frame(AVCodecContext *avctx,
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
*got_frame_ptr = 1;
*(AVFrame *)data = alac->frame;
return input_buffer_size;
}
@ -637,6 +644,9 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
return ret;
}
avcodec_get_frame_defaults(&alac->frame);
avctx->coded_frame = &alac->frame;
return 0;
}
@ -648,5 +658,6 @@ AVCodec ff_alac_decoder = {
.init = alac_decode_init,
.close = alac_decode_close,
.decode = alac_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};

43
libavcodec/alsdec.c
View File

@ -191,6 +191,7 @@ typedef struct {
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
ALSSpecificConfig sconf;
GetBitContext gb;
DSPContext dsp;
@ -1415,15 +1416,14 @@ static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame)
/** Decode an ALS frame.
*/
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
ALSDecContext *ctx = avctx->priv_data;
ALSSpecificConfig *sconf = &ctx->sconf;
const uint8_t *buffer = avpkt->data;
int buffer_size = avpkt->size;
int invalid_frame, size;
int invalid_frame, ret;
unsigned int c, sample, ra_frame, bytes_read, shift;
init_get_bits(&ctx->gb, buffer, buffer_size * 8);
@ -1448,21 +1448,17 @@ static int decode_frame(AVCodecContext *avctx,
ctx->frame_id++;
// check for size of decoded data
size = ctx->cur_frame_length * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
return -1;
/* get output buffer */
ctx->frame.nb_samples = ctx->cur_frame_length;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*data_size = size;
// transform decoded frame into output format
#define INTERLEAVE_OUTPUT(bps) \
{ \
int##bps##_t *dest = (int##bps##_t*) data; \
int##bps##_t *dest = (int##bps##_t*)ctx->frame.data[0]; \
shift = bps - ctx->avctx->bits_per_raw_sample; \
for (sample = 0; sample < ctx->cur_frame_length; sample++) \
for (c = 0; c < avctx->channels; c++) \
@ -1480,7 +1476,7 @@ static int decode_frame(AVCodecContext *avctx,
int swap = HAVE_BIGENDIAN != sconf->msb_first;
if (ctx->avctx->bits_per_raw_sample == 24) {
int32_t *src = data;
int32_t *src = (int32_t *)ctx->frame.data[0];
for (sample = 0;
sample < ctx->cur_frame_length * avctx->channels;
@ -1501,22 +1497,25 @@ static int decode_frame(AVCodecContext *avctx,
if (swap) {
if (ctx->avctx->bits_per_raw_sample <= 16) {
int16_t *src = (int16_t*) data;
int16_t *src = (int16_t*) ctx->frame.data[0];
int16_t *dest = (int16_t*) ctx->crc_buffer;
for (sample = 0;
sample < ctx->cur_frame_length * avctx->channels;
sample++)
*dest++ = av_bswap16(src[sample]);
} else {
ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer, data,
ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer,
(uint32_t *)ctx->frame.data[0],
ctx->cur_frame_length * avctx->channels);
}
crc_source = ctx->crc_buffer;
} else {
crc_source = data;
crc_source = ctx->frame.data[0];
}
ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source, size);
ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source,
ctx->cur_frame_length * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt));
}
@ -1527,6 +1526,9 @@ static int decode_frame(AVCodecContext *avctx,
}
}
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->frame;
bytes_read = invalid_frame ? buffer_size :
(get_bits_count(&ctx->gb) + 7) >> 3;
@ -1724,6 +1726,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
dsputil_init(&ctx->dsp, avctx);
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
return 0;
}
@ -1747,7 +1752,7 @@ AVCodec ff_als_decoder = {
.close = decode_end,
.decode = decode_frame,
.flush = flush,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MPEG-4 Audio Lossless Coding (ALS)"),
};

25
libavcodec/amrnbdec.c
View File

@ -95,6 +95,7 @@
#define AMR_AGC_ALPHA 0.9
typedef struct AMRContext {
AVFrame avframe; ///< AVFrame for decoded samples
AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
enum Mode cur_frame_mode;
@ -167,6 +168,9 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
for (i = 0; i < 4; i++)
p->prediction_error[i] = MIN_ENERGY;
avcodec_get_frame_defaults(&p->avframe);
avctx->coded_frame = &p->avframe;
return 0;
}
@ -919,21 +923,29 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
/// @}
static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AMRContext *p = avctx->priv_data; // pointer to private data
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *buf_out = data; // pointer to the output data buffer
int i, subframe;
float *buf_out; // pointer to the output data buffer
int i, subframe, ret;
float fixed_gain_factor;
AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
float synth_fixed_gain; // the fixed gain that synthesis should use
const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
/* get output buffer */
p->avframe.nb_samples = AMR_BLOCK_SIZE;
if ((ret = avctx->get_buffer(avctx, &p->avframe)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
buf_out = (float *)p->avframe.data[0];
p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
if (p->cur_frame_mode == MODE_DTX) {
av_log_missing_feature(avctx, "dtx mode", 1);
@ -1028,8 +1040,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
0.84, 0.16, LP_FILTER_ORDER);
/* report how many samples we got */
*data_size = AMR_BLOCK_SIZE * sizeof(float);
*got_frame_ptr = 1;
*(AVFrame *)data = p->avframe;
/* return the amount of bytes consumed if everything was OK */
return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
@ -1043,6 +1055,7 @@ AVCodec ff_amrnb_decoder = {
.priv_data_size = sizeof(AMRContext),
.init = amrnb_decode_init,
.decode = amrnb_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
};

27
libavcodec/amrwbdec.c
View File

@ -41,6 +41,7 @@
#include "amrwbdata.h"
typedef struct {
AVFrame avframe; ///< AVFrame for decoded samples
AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
enum Mode fr_cur_mode; ///< mode index of current frame
uint8_t fr_quality; ///< frame quality index (FQI)
@ -102,6 +103,9 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
for (i = 0; i < 4; i++)
ctx->prediction_error[i] = MIN_ENERGY;
avcodec_get_frame_defaults(&ctx->avframe);
avctx->coded_frame = &ctx->avframe;
return 0;
}
@ -1062,15 +1066,15 @@ static void update_sub_state(AMRWBContext *ctx)
LP_ORDER_16k * sizeof(float));
}
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AMRWBContext *ctx = avctx->priv_data;
AMRWBFrame *cf = &ctx->frame;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int expected_fr_size, header_size;
float *buf_out = data;
float *buf_out;
float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
float fixed_gain_factor; // fixed gain correction factor (gamma)
float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
@ -1080,7 +1084,15 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
float hb_gain;
int sub, i;
int sub, i, ret;
/* get output buffer */
ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k;
if ((ret = avctx->get_buffer(avctx, &ctx->avframe)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
buf_out = (float *)ctx->avframe.data[0];
header_size = decode_mime_header(ctx, buf);
expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
@ -1088,7 +1100,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
if (buf_size < expected_fr_size) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
@ -1219,8 +1231,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
/* report how many samples we got */
*data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float);
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->avframe;
return expected_fr_size;
}
@ -1232,6 +1244,7 @@ AVCodec ff_amrwb_decoder = {
.priv_data_size = sizeof(AMRWBContext),
.init = amrwb_decode_init,
.decode = amrwb_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
};

37
libavcodec/apedec.c
View File

@ -129,6 +129,7 @@ typedef struct APEPredictor {
/** Decoder context */
typedef struct APEContext {
AVCodecContext *avctx;
AVFrame frame;
DSPContext dsp;
int channels;
int samples; ///< samples left to decode in current frame
@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
filter_alloc_fail:
ape_decode_close(avctx);
@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
}
}
static int ape_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int ape_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
int16_t *samples = data;
int i;
int blockstodecode, out_size;
int16_t *samples;
int i, ret;
int blockstodecode;
int bytes_used = 0;
/* this should never be negative, but bad things will happen if it is, so
@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx,
void *tmp_data;
if (!buf_size) {
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
if (buf_size < 8) {
@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx,
}
if (!s->data) {
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples);
out_size = blockstodecode * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small.\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = blockstodecode;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
s->error=0;
@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx,
s->samples -= blockstodecode;
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return bytes_used;
}
@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = {
.init = ape_decode_init,
.close = ape_decode_close,
.decode = ape_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
.flush = ape_flush,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};

26
libavcodec/atrac1.c
View File

@ -72,6 +72,7 @@ typedef struct {
* The atrac1 context, holds all needed parameters for decoding
*/
typedef struct {
AVFrame frame;
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer
@ -273,14 +274,14 @@ static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AT1Ctx *q = avctx->priv_data;
int ch, ret, out_size;
int ch, ret;
GetBitContext gb;
float* samples = data;
float *samples;
if (buf_size < 212 * q->channels) {
@ -288,12 +289,13 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
out_size = q->channels * AT1_SU_SAMPLES *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
q->frame.nb_samples = AT1_SU_SAMPLES;
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];
for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
@ -321,7 +323,9 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
AT1_SU_SAMPLES, 2);
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return avctx->block_align;
}
@ -389,6 +393,9 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
q->SUs[1].spectrum[0] = q->SUs[1].spec1;
q->SUs[1].spectrum[1] = q->SUs[1].spec2;
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
return 0;
}
@ -401,5 +408,6 @@ AVCodec ff_atrac1_decoder = {
.init = atrac1_decode_init,
.close = atrac1_decode_end,
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};

34
libavcodec/atrac3.c
View File

@ -86,6 +86,7 @@ typedef struct {
} channel_unit;
typedef struct {
AVFrame frame;
GetBitContext gb;
//@{
/** stream data */
@ -823,16 +824,16 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
* @param avctx pointer to the AVCodecContext
*/
static int atrac3_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
int result = 0, out_size;
int result;
const uint8_t* databuf;
float *samples_flt = data;
int16_t *samples_s16 = data;
float *samples_flt;
int16_t *samples_s16;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR,
@ -840,12 +841,14 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
return AVERROR_INVALIDDATA;
}
out_size = SAMPLES_PER_FRAME * q->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
q->frame.nb_samples = SAMPLES_PER_FRAME;
if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return result;
}
samples_flt = (float *)q->frame.data[0];
samples_s16 = (int16_t *)q->frame.data[0];
/* Check if we need to descramble and what buffer to pass on. */
if (q->scrambled_stream) {
@ -875,7 +878,9 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
(const float **)q->outSamples,
SAMPLES_PER_FRAME, q->channels);
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return avctx->block_align;
}
@ -1047,6 +1052,9 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
}
}
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
return 0;
}
@ -1060,6 +1068,6 @@ AVCodec ff_atrac3_decoder =
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};

145
libavcodec/avcodec.h
View File

@ -480,8 +480,10 @@ enum CodecID {
#define CH_LAYOUT_STEREO_DOWNMIX AV_CH_LAYOUT_STEREO_DOWNMIX
#endif
#if FF_API_OLD_DECODE_AUDIO
/* in bytes */
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
#endif
/**
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
@ -933,13 +935,24 @@ typedef struct AVFrame {
#define AV_NUM_DATA_POINTERS 8
#endif
/**
* pointer to the picture planes.
* pointer to the picture/channel planes.
* This might be different from the first allocated byte
* - encoding:
* - decoding:
* - encoding: Set by user
* - decoding: set by AVCodecContext.get_buffer()
*/
uint8_t *data[AV_NUM_DATA_POINTERS];
/**
* Size, in bytes, of the data for each picture/channel plane.
*
* For audio, only linesize[0] may be set. For planar audio, each channel
* plane must be the same size.
*
* - encoding: Set by user (video only)
* - decoding: set by AVCodecContext.get_buffer()
*/
int linesize[AV_NUM_DATA_POINTERS];
/**
* pointer to the first allocated byte of the picture. Can be used in get_buffer/release_buffer.
* This isn't used by libavcodec unless the default get/release_buffer() is used.
@ -993,7 +1006,7 @@ typedef struct AVFrame {
* buffer age (1->was last buffer and dint change, 2->..., ...).
* Set to INT_MAX if the buffer has not been used yet.
* - encoding: unused
* - decoding: MUST be set by get_buffer().
* - decoding: MUST be set by get_buffer() for video.
*/
int age;
@ -1190,6 +1203,33 @@ typedef struct AVFrame {
* - decoding: Set by libavcodec.
*/
void *thread_opaque;
/**
* number of audio samples (per channel) described by this frame
* - encoding: unused
* - decoding: Set by libavcodec
*/
int nb_samples;
/**
* pointers to the data planes/channels.
*
* For video, this should simply point to data[].
*
* For planar audio, each channel has a separate data pointer, and
* linesize[0] contains the size of each channel buffer.
* For packed audio, there is just one data pointer, and linesize[0]
* contains the total size of the buffer for all channels.
*
* Note: Both data and extended_data will always be set by get_buffer(),
* but for planar audio with more channels that can fit in data,
* extended_data must be used by the decoder in order to access all
* channels.
*
* encoding: unused
* decoding: set by AVCodecContext.get_buffer()
*/
uint8_t **extended_data;
} AVFrame;
struct AVCodecInternal;
@ -1545,15 +1585,56 @@ typedef struct AVCodecContext {
/**
* Called at the beginning of each frame to get a buffer for it.
* If pic.reference is set then the frame will be read later by libavcodec.
* avcodec_align_dimensions2() should be used to find the required width and
* height, as they normally need to be rounded up to the next multiple of 16.
*
* The function will set AVFrame.data[], AVFrame.linesize[].
* AVFrame.extended_data[] must also be set, but it should be the same as
* AVFrame.data[] except for planar audio with more channels than can fit
* in AVFrame.data[]. In that case, AVFrame.data[] shall still contain as
* many data pointers as it can hold.
*
* if CODEC_CAP_DR1 is not set then get_buffer() must call
* avcodec_default_get_buffer() instead of providing buffers allocated by
* some other means.
*
* AVFrame.data[] should be 32- or 16-byte-aligned unless the CPU doesn't
* need it. avcodec_default_get_buffer() aligns the output buffer properly,
* but if get_buffer() is overridden then alignment considerations should
* be taken into account.
*
* @see avcodec_default_get_buffer()
*
* Video:
*
* If pic.reference is set then the frame will be read later by libavcodec.
* avcodec_align_dimensions2() should be used to find the required width and
* height, as they normally need to be rounded up to the next multiple of 16.
*
* If frame multithreading is used and thread_safe_callbacks is set,
* it may be called from a different thread, but not from more than one at once.
* Does not need to be reentrant.
* it may be called from a different thread, but not from more than one at
* once. Does not need to be reentrant.
*
* @see release_buffer(), reget_buffer()
* @see avcodec_align_dimensions2()
*
* Audio:
*
* Decoders request a buffer of a particular size by setting
* AVFrame.nb_samples prior to calling get_buffer(). The decoder may,
* however, utilize only part of the buffer by setting AVFrame.nb_samples
* to a smaller value in the output frame.
*
* Decoders cannot use the buffer after returning from
* avcodec_decode_audio4(), so they will not call release_buffer(), as it
* is assumed to be released immediately upon return.
*
* As a convenience, av_samples_get_buffer_size() and
* av_samples_fill_arrays() in libavutil may be used by custom get_buffer()
* functions to find the required data size and to fill data pointers and
* linesize. In AVFrame.linesize, only linesize[0] may be set for audio
* since all planes must be the same size.
*
* @see av_samples_get_buffer_size(), av_samples_fill_arrays()
*
* - encoding: unused
* - decoding: Set by libavcodec, user can override.
*/
@ -3882,7 +3963,12 @@ int avcodec_open(AVCodecContext *avctx, AVCodec *codec);
*/
int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options);
#if FF_API_OLD_DECODE_AUDIO
/**
* Wrapper function which calls avcodec_decode_audio4.
*
* @deprecated Use avcodec_decode_audio4 instead.
*
* Decode the audio frame of size avpkt->size from avpkt->data into samples.
* Some decoders may support multiple frames in a single AVPacket, such
* decoders would then just decode the first frame. In this case,
@ -3917,6 +4003,8 @@ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options)
*
* @param avctx the codec context
* @param[out] samples the output buffer, sample type in avctx->sample_fmt
* If the sample format is planar, each channel plane will
* be the same size, with no padding between channels.
* @param[in,out] frame_size_ptr the output buffer size in bytes
* @param[in] avpkt The input AVPacket containing the input buffer.
* You can create such packet with av_init_packet() and by then setting
@ -3925,9 +4013,46 @@ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options)
* @return On error a negative value is returned, otherwise the number of bytes
* used or zero if no frame data was decompressed (used) from the input AVPacket.
*/
int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
AVPacket *avpkt);
#endif
/**
* Decode the audio frame of size avpkt->size from avpkt->data into frame.
*
* Some decoders may support multiple frames in a single AVPacket. Such
* decoders would then just decode the first frame. In this case,
* avcodec_decode_audio4 has to be called again with an AVPacket containing
* the remaining data in order to decode the second frame, etc...
* Even if no frames are returned, the packet needs to be fed to the decoder
* with remaining data until it is completely consumed or an error occurs.
*
* @warning The input buffer, avpkt->data must be FF_INPUT_BUFFER_PADDING_SIZE
* larger than the actual read bytes because some optimized bitstream
* readers read 32 or 64 bits at once and could read over the end.
*
* @note You might have to align the input buffer. The alignment requirements
* depend on the CPU and the decoder.
*
* @param avctx the codec context
* @param[out] frame The AVFrame in which to store decoded audio samples.
* Decoders request a buffer of a particular size by setting
* AVFrame.nb_samples prior to calling get_buffer(). The
* decoder may, however, only utilize part of the buffer by
* setting AVFrame.nb_samples to a smaller value in the
* output frame.
* @param[out] got_frame_ptr Zero if no frame could be decoded, otherwise it is
* non-zero.
* @param[in] avpkt The input AVPacket containing the input buffer.
* At least avpkt->data and avpkt->size should be set. Some
* decoders might also require additional fields to be set.
* @return A negative error code is returned if an error occurred during
* decoding, otherwise the number of bytes consumed from the input
* AVPacket is returned.
*/
int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt);
/**
* Decode the video frame of size avpkt->size from avpkt->data into picture.

34
libavcodec/binkaudio.c
View File

@ -45,6 +45,7 @@ static float quant_table[96];
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
AVFrame frame;
GetBitContext gb;
DSPContext dsp;
FmtConvertContext fmt_conv;
@ -147,6 +148,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
else
return -1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -293,6 +297,7 @@ static av_cold int decode_end(AVCodecContext *avctx)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_end(&s->trans.dct);
return 0;
}
@ -302,20 +307,19 @@ static void get_bits_align32(GetBitContext *s)
if (n) skip_bits(s, n);
}
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
int16_t *samples = data;
int16_t *samples;
GetBitContext *gb = &s->gb;
int out_size, consumed = 0;
int ret, consumed = 0;
if (!get_bits_left(gb)) {
uint8_t *buf;
/* handle end-of-stream */
if (!avpkt->size) {
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
if (avpkt->size < 4) {
@ -334,11 +338,13 @@ static int decode_frame(AVCodecContext *avctx,
skip_bits_long(gb, 32);
}
out_size = s->block_size * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = s->block_size / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) {
av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
@ -346,7 +352,9 @@ static int decode_frame(AVCodecContext *avctx,
}
get_bits_align32(gb);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return consumed;
}
@ -358,7 +366,7 @@ AVCodec ff_binkaudio_rdft_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
@ -370,6 +378,6 @@ AVCodec ff_binkaudio_dct_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
};

41
libavcodec/cook.c
View File

@ -122,6 +122,7 @@ typedef struct cook {
void (* saturate_output) (struct cook *q, int chan, float *out);
AVCodecContext* avctx;
AVFrame frame;
GetBitContext gb;
/* stream data */
int nb_channels;
@ -131,6 +132,7 @@ typedef struct cook {
int samples_per_channel;
/* states */
AVLFG random_state;
int discarded_packets;
/* transform data */
FFTContext mdct_ctx;
@ -896,6 +898,7 @@ mlt_compensate_output(COOKContext *q, float *decode_buffer,
float *out, int chan)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
if (out)
q->saturate_output(q, chan, out);
}
@ -953,24 +956,28 @@ static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
* @param avctx pointer to the AVCodecContext
*/
static int cook_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
static int cook_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
int i, out_size;
float *samples = NULL;
int i, ret;
int offset = 0;
int chidx = 0;
if (buf_size < avctx->block_align)
return buf_size;
out_size = q->nb_channels * q->samples_per_channel *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
if (q->discarded_packets >= 2) {
q->frame.nb_samples = q->samples_per_channel;
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];
}
/* estimate subpacket sizes */
@ -990,15 +997,21 @@ static int cook_decode_frame(AVCodecContext *avctx,
q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
q->subpacket[i].ch_idx = chidx;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
decode_subpacket(q, &q->subpacket[i], buf + offset, data);
decode_subpacket(q, &q->subpacket[i], buf + offset, samples);
offset += q->subpacket[i].size;
chidx += q->subpacket[i].num_channels;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
}
*data_size = out_size;
/* Discard the first two frames: no valid audio. */
if (avctx->frame_number < 2) *data_size = 0;
if (q->discarded_packets < 2) {
q->discarded_packets++;
*got_frame_ptr = 0;
return avctx->block_align;
}
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return avctx->block_align;
}
@ -1246,6 +1259,9 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
else
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
#ifdef DEBUG
dump_cook_context(q);
#endif
@ -1262,5 +1278,6 @@ AVCodec ff_cook_decoder =
.init = cook_decode_init,
.close = cook_decode_close,
.decode = cook_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("COOK"),
};

32
libavcodec/dca.c
View File

@ -261,6 +261,7 @@ static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int id
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
@ -1635,9 +1636,8 @@ static void dca_exss_parse_header(DCAContext *s)
* Main frame decoding function
* FIXME add arguments
*/
static int dca_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int dca_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -1645,9 +1645,8 @@ static int dca_decode_frame(AVCodecContext * avctx,
int lfe_samples;
int num_core_channels = 0;
int i, ret;
float *samples_flt = data;
int16_t *samples_s16 = data;
int out_size;
float *samples_flt;
int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
@ -1839,11 +1838,14 @@ static int dca_decode_frame(AVCodecContext * avctx,
return AVERROR_PATCHWELCOME;
}
out_size = 256 / 8 * s->sample_blocks * channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size)
return AVERROR(EINVAL);
*data_size = out_size;
/* get output buffer */
s->frame.nb_samples = 256 * (s->sample_blocks / 8);
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples_flt = (float *)s->frame.data[0];
samples_s16 = (int16_t *)s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
@ -1877,6 +1879,9 @@ static int dca_decode_frame(AVCodecContext * avctx,
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -1919,6 +1924,9 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
avctx->channels = avctx->request_channels;
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -1947,7 +1955,7 @@ AVCodec ff_dca_decoder = {
.decode = dca_decode_frame,
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF,
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},

28
libavcodec/dpcm.c
View File

@ -42,6 +42,7 @@
#include "bytestream.h"
typedef struct DPCMContext {
AVFrame frame;
int channels;
int16_t roq_square_array[256];
int sample[2]; ///< previous sample (for SOL_DPCM)
@ -162,22 +163,25 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx)
else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
const uint8_t *buf_end = buf + buf_size;
DPCMContext *s = avctx->priv_data;
int out = 0;
int out = 0, ret;
int predictor[2];
int ch = 0;
int stereo = s->channels - 1;
int16_t *output_samples = data;
int16_t *output_samples;
/* calculate output size */
switch(avctx->codec->id) {
@ -197,15 +201,18 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
out = buf_size;
break;
}
out *= av_get_bytes_per_sample(avctx->sample_fmt);
if (out <= 0) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
if (*data_size < out) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = out / s->channels;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
output_samples = (int16_t *)s->frame.data[0];
switch(avctx->codec->id) {
@ -307,7 +314,9 @@ static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
break;
}
*data_size = out;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -319,6 +328,7 @@ AVCodec ff_ ## name_ ## _decoder = { \
.priv_data_size = sizeof(DPCMContext), \
.init = dpcm_decode_init, \
.decode = dpcm_decode_frame, \
.capabilities = CODEC_CAP_DR1, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
}

28
libavcodec/dsicinav.c
View File

@ -44,6 +44,7 @@ typedef struct CinVideoContext {
} CinVideoContext;
typedef struct CinAudioContext {
AVFrame frame;
int initial_decode_frame;
int delta;
} CinAudioContext;
@ -317,25 +318,28 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx)
cin->delta = 0;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&cin->frame);
avctx->coded_frame = &cin->frame;
return 0;
}
static int cinaudio_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int cinaudio_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
CinAudioContext *cin = avctx->priv_data;
const uint8_t *buf_end = buf + avpkt->size;
int16_t *samples = data;
int delta, out_size;
int16_t *samples;
int delta, ret;
out_size = (avpkt->size - cin->initial_decode_frame) *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
cin->frame.nb_samples = avpkt->size - cin->initial_decode_frame;
if ((ret = avctx->get_buffer(avctx, &cin->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)cin->frame.data[0];
delta = cin->delta;
if (cin->initial_decode_frame) {
@ -351,7 +355,8 @@ static int cinaudio_decode_frame(AVCodecContext *avctx,
}
cin->delta = delta;
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = cin->frame;
return avpkt->size;
}
@ -376,5 +381,6 @@ AVCodec ff_dsicinaudio_decoder = {
.priv_data_size = sizeof(CinAudioContext),
.init = cinaudio_decode_init,
.decode = cinaudio_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Delphine Software International CIN audio"),
};

37
libavcodec/flacdec.c
View File

@ -49,6 +49,7 @@ typedef struct FLACContext {
FLACSTREAMINFO
AVCodecContext *avctx; ///< parent AVCodecContext
AVFrame frame;
GetBitContext gb; ///< GetBitContext initialized to start at the current frame
int blocksize; ///< number of samples in the current frame
@ -116,6 +117,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
allocate_buffers(s);
s->got_streaminfo = 1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -542,20 +546,18 @@ static int decode_frame(FLACContext *s)
return 0;
}
static int flac_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int flac_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
int i, j = 0, bytes_read = 0;
int16_t *samples_16 = data;
int32_t *samples_32 = data;
int alloc_data_size= *data_size;
int output_size;
int16_t *samples_16;
int32_t *samples_32;
int ret;
*data_size=0;
*got_frame_ptr = 0;
if (s->max_framesize == 0) {
s->max_framesize =
@ -586,15 +588,14 @@ static int flac_decode_frame(AVCodecContext *avctx,
}
bytes_read = (get_bits_count(&s->gb)+7)/8;
/* check if allocated data size is large enough for output */
output_size = s->blocksize * s->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (output_size > alloc_data_size) {
av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
"allocated data size\n");
return -1;
/* get output buffer */
s->frame.nb_samples = s->blocksize;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*data_size = output_size;
samples_16 = (int16_t *)s->frame.data[0];
samples_32 = (int32_t *)s->frame.data[0];
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
@ -639,6 +640,9 @@ static int flac_decode_frame(AVCodecContext *avctx,
buf_size - bytes_read, buf_size);
}
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return bytes_read;
}
@ -662,5 +666,6 @@ AVCodec ff_flac_decoder = {
.init = flac_decode_init,
.close = flac_decode_close,
.decode = flac_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};

2
libavcodec/g722.h
View File

@ -26,10 +26,12 @@
#define AVCODEC_G722_H
#include <stdint.h>
#include "avcodec.h"
#define PREV_SAMPLES_BUF_SIZE 1024
typedef struct {
AVFrame frame;
int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
int prev_samples_pos; ///< the number of values in prev_samples

25
libavcodec/g722dec.c
View File

@ -66,6 +66,9 @@ static av_cold int g722_decode_init(AVCodecContext * avctx)
c->band[1].scale_factor = 2;
c->prev_samples_pos = 22;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
@ -81,20 +84,22 @@ static const int16_t *low_inv_quants[3] = { ff_g722_low_inv_quant6,
ff_g722_low_inv_quant4 };
static int g722_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
G722Context *c = avctx->priv_data;
int16_t *out_buf = data;
int j, out_len;
int16_t *out_buf;
int j, ret;
const int skip = 8 - avctx->bits_per_coded_sample;
const int16_t *quantizer_table = low_inv_quants[skip];
GetBitContext gb;
out_len = avpkt->size * 2 * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_len) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = avpkt->size * 2;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_buf = (int16_t *)c->frame.data[0];
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
@ -128,7 +133,10 @@ static int g722_decode_frame(AVCodecContext *avctx, void *data,
c->prev_samples_pos = 22;
}
}
*data_size = out_len;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return avpkt->size;
}
@ -139,5 +147,6 @@ AVCodec ff_adpcm_g722_decoder = {
.priv_data_size = sizeof(G722Context),
.init = g722_decode_init,
.decode = g722_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
};

29
libavcodec/g726.c
View File

@ -75,6 +75,7 @@ typedef struct G726Tables {
typedef struct G726Context {
AVClass *class;
AVFrame frame;
G726Tables tbls; /**< static tables needed for computation */
Float11 sr[2]; /**< prev. reconstructed samples */
@ -427,26 +428,31 @@ static av_cold int g726_decode_init(AVCodecContext *avctx)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
static int g726_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int g726_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
G726Context *c = avctx->priv_data;
int16_t *samples = data;
int16_t *samples;
GetBitContext gb;
int out_samples, out_size;
int out_samples, ret;
out_samples = buf_size * 8 / c->code_size;
out_size = out_samples * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = out_samples;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
init_get_bits(&gb, buf, buf_size * 8);
@ -456,7 +462,9 @@ static int g726_decode_frame(AVCodecContext *avctx,
if (get_bits_left(&gb) > 0)
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return buf_size;
}
@ -474,6 +482,7 @@ AVCodec ff_adpcm_g726_decoder = {
.init = g726_decode_init,
.decode = g726_decode_frame,
.flush = g726_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif

32
libavcodec/gsmdec.c
View File

@ -32,6 +32,8 @@
static av_cold int gsm_init(AVCodecContext *avctx)
{
GSMContext *s = avctx->priv_data;
avctx->channels = 1;
if (!avctx->sample_rate)
avctx->sample_rate = 8000;
@ -47,30 +49,35 @@ static av_cold int gsm_init(AVCodecContext *avctx)
avctx->block_align = GSM_MS_BLOCK_SIZE;
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int gsm_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
GSMContext *s = avctx->priv_data;
int res;
GetBitContext gb;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples = data;
int frame_bytes = avctx->frame_size *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < frame_bytes) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
int16_t *samples;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
s->frame.nb_samples = avctx->frame_size;
if ((res = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return res;
}
samples = (int16_t *)s->frame.data[0];
switch (avctx->codec_id) {
case CODEC_ID_GSM:
init_get_bits(&gb, buf, buf_size * 8);
@ -85,7 +92,10 @@ static int gsm_decode_frame(AVCodecContext *avctx, void *data,
if (res < 0)
return res;
}
*data_size = frame_bytes;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return avctx->block_align;
}
@ -103,6 +113,7 @@ AVCodec ff_gsm_decoder = {
.init = gsm_init,
.decode = gsm_decode_frame,
.flush = gsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("GSM"),
};
@ -114,5 +125,6 @@ AVCodec ff_gsm_ms_decoder = {
.init = gsm_init,
.decode = gsm_decode_frame,
.flush = gsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("GSM Microsoft variant"),
};

2
libavcodec/gsmdec_data.h
View File

@ -23,6 +23,7 @@
#define AVCODEC_GSMDEC_DATA
#include <stdint.h>
#include "avcodec.h"
// input and output sizes in byte
#define GSM_BLOCK_SIZE 33
@ -30,6 +31,7 @@
#define GSM_FRAME_SIZE 160
typedef struct {
AVFrame frame;
// Contains first 120 elements from the previous frame
// (used by long_term_synth according to the "lag"),
// then in the following 160 elements the current

29
libavcodec/imc.c
View File

@ -51,6 +51,8 @@
#define COEFFS 256
typedef struct {
AVFrame frame;
float old_floor[BANDS];
float flcoeffs1[BANDS];
float flcoeffs2[BANDS];
@ -168,6 +170,10 @@ static av_cold int imc_decode_init(AVCodecContext * avctx)
dsputil_init(&q->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&q->frame);
avctx->coded_frame = &q->frame;
return 0;
}
@ -649,9 +655,8 @@ static int imc_get_coeffs (IMCContext* q) {
return 0;
}
static int imc_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int imc_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -659,7 +664,7 @@ static int imc_decode_frame(AVCodecContext * avctx,
IMCContext *q = avctx->priv_data;
int stream_format_code;
int imc_hdr, i, j, out_size, ret;
int imc_hdr, i, j, ret;
int flag;
int bits, summer;
int counter, bitscount;
@ -670,15 +675,16 @@ static int imc_decode_frame(AVCodecContext * avctx,
return AVERROR_INVALIDDATA;
}
out_size = COEFFS * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
q->frame.nb_samples = COEFFS;
if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
q->out_samples = (float *)q->frame.data[0];
q->dsp.bswap16_buf(buf16, (const uint16_t*)buf, IMC_BLOCK_SIZE / 2);
q->out_samples = data;
init_get_bits(&q->gb, (const uint8_t*)buf16, IMC_BLOCK_SIZE * 8);
/* Check the frame header */
@ -823,7 +829,8 @@ static int imc_decode_frame(AVCodecContext * avctx,
imc_imdct256(q);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = q->frame;
return IMC_BLOCK_SIZE;
}
@ -834,6 +841,7 @@ static av_cold int imc_decode_close(AVCodecContext * avctx)
IMCContext *q = avctx->priv_data;
ff_fft_end(&q->fft);
return 0;
}
@ -846,5 +854,6 @@ AVCodec ff_imc_decoder = {
.init = imc_decode_init,
.close = imc_decode_close,
.decode = imc_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("IMC (Intel Music Coder)"),
};

9
libavcodec/internal.h
View File

@ -31,12 +31,15 @@
typedef struct InternalBuffer {
int last_pic_num;
uint8_t *base[4];
uint8_t *data[4];
int linesize[4];
uint8_t *base[AV_NUM_DATA_POINTERS];
uint8_t *data[AV_NUM_DATA_POINTERS];
int linesize[AV_NUM_DATA_POINTERS];
int width;
int height;
enum PixelFormat pix_fmt;
uint8_t **extended_data;
int audio_data_size;
int nb_channels;
} InternalBuffer;
typedef struct AVCodecInternal {

58
libavcodec/libgsm.c
View File

@ -124,7 +124,14 @@ AVCodec ff_libgsm_ms_encoder = {
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};
typedef struct LibGSMDecodeContext {
AVFrame frame;
struct gsm_state *state;
} LibGSMDecodeContext;
static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
LibGSMDecodeContext *s = avctx->priv_data;
if (avctx->channels > 1) {
av_log(avctx, AV_LOG_ERROR, "Mono required for GSM, got %d channels\n",
avctx->channels);
@ -139,7 +146,7 @@ static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->priv_data = gsm_create();
s->state = gsm_create();
switch(avctx->codec_id) {
case CODEC_ID_GSM:
@ -154,59 +161,72 @@ static av_cold int libgsm_decode_init(AVCodecContext *avctx) {
}
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static av_cold int libgsm_decode_close(AVCodecContext *avctx) {
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
LibGSMDecodeContext *s = avctx->priv_data;
gsm_destroy(s->state);
s->state = NULL;
return 0;
}
static int libgsm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
static int libgsm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
int i, ret;
struct gsm_state *s = avctx->priv_data;
LibGSMDecodeContext *s = avctx->priv_data;
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples = data;
int out_size = avctx->frame_size * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
int16_t *samples;
if (buf_size < avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
/* get output buffer */
s->frame.nb_samples = avctx->frame_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
for (i = 0; i < avctx->frame_size / GSM_FRAME_SIZE; i++) {
if ((ret = gsm_decode(s, buf, samples)) < 0)
if ((ret = gsm_decode(s->state, buf, samples)) < 0)
return -1;
buf += GSM_BLOCK_SIZE;
samples += GSM_FRAME_SIZE;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return avctx->block_align;
}
static void libgsm_flush(AVCodecContext *avctx) {
gsm_destroy(avctx->priv_data);
avctx->priv_data = gsm_create();
LibGSMDecodeContext *s = avctx->priv_data;
gsm_destroy(s->state);
s->state = gsm_create();
}
AVCodec ff_libgsm_decoder = {
.name = "libgsm",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM,
.priv_data_size = sizeof(LibGSMDecodeContext),
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
@ -214,9 +234,11 @@ AVCodec ff_libgsm_ms_decoder = {
.name = "libgsm_ms",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_GSM_MS,
.priv_data_size = sizeof(LibGSMDecodeContext),
.init = libgsm_decode_init,
.close = libgsm_decode_close,
.decode = libgsm_decode_frame,
.flush = libgsm_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};

52
libavcodec/libopencore-amr.c
View File

@ -79,6 +79,7 @@ static int get_bitrate_mode(int bitrate, void *log_ctx)
typedef struct AMRContext {
AVClass *av_class;
AVFrame frame;
void *dec_state;
void *enc_state;
int enc_bitrate;
@ -112,6 +113,9 @@ static av_cold int amr_nb_decode_init(AVCodecContext *avctx)
return AVERROR(ENOSYS);
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -120,26 +124,28 @@ static av_cold int amr_nb_decode_close(AVCodecContext *avctx)
AMRContext *s = avctx->priv_data;
Decoder_Interface_exit(s->dec_state);
return 0;
}
static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AMRContext *s = avctx->priv_data;
static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
enum Mode dec_mode;
int packet_size, out_size;
int packet_size, ret;
av_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
buf, buf_size, avctx->frame_number);
out_size = 160 * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = 160;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
dec_mode = (buf[0] >> 3) & 0x000F;
@ -154,8 +160,10 @@ static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
av_dlog(avctx, "packet_size=%d buf= 0x%X %X %X %X\n",
packet_size, buf[0], buf[1], buf[2], buf[3]);
/* call decoder */
Decoder_Interface_Decode(s->dec_state, buf, data, 0);
*data_size = out_size;
Decoder_Interface_Decode(s->dec_state, buf, (short *)s->frame.data[0], 0);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return packet_size;
}
@ -168,6 +176,7 @@ AVCodec ff_libopencore_amrnb_decoder = {
.init = amr_nb_decode_init,
.close = amr_nb_decode_close,
.decode = amr_nb_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"),
};
@ -251,6 +260,7 @@ AVCodec ff_libopencore_amrnb_encoder = {
#include <opencore-amrwb/if_rom.h>
typedef struct AMRWBContext {
AVFrame frame;
void *state;
} AMRWBContext;
@ -267,23 +277,27 @@ static av_cold int amr_wb_decode_init(AVCodecContext *avctx)
return AVERROR(ENOSYS);
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AMRWBContext *s = avctx->priv_data;
int mode;
int packet_size, out_size;
int mode, ret;
int packet_size;
static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
out_size = 320 * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = 320;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
mode = (buf[0] >> 3) & 0x000F;
@ -295,8 +309,11 @@ static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
D_IF_decode(s->state, buf, data, _good_frame);
*data_size = out_size;
D_IF_decode(s->state, buf, (short *)s->frame.data[0], _good_frame);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return packet_size;
}
@ -316,6 +333,7 @@ AVCodec ff_libopencore_amrwb_decoder = {
.init = amr_wb_decode_init,
.close = amr_wb_decode_close,
.decode = amr_wb_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Wide-Band"),
};

34
libavcodec/libspeexdec.c
View File

@ -25,6 +25,7 @@
#include "avcodec.h"
typedef struct {
AVFrame frame;
SpeexBits bits;
SpeexStereoState stereo;
void *dec_state;
@ -89,26 +90,29 @@ static av_cold int libspeex_decode_init(AVCodecContext *avctx)
s->stereo = (SpeexStereoState)SPEEX_STEREO_STATE_INIT;
speex_decoder_ctl(s->dec_state, SPEEX_SET_HANDLER, &callback);
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int libspeex_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int libspeex_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
LibSpeexContext *s = avctx->priv_data;
int16_t *output = data;
int out_size, ret, consumed = 0;
int16_t *output;
int ret, consumed = 0;
/* check output buffer size */
out_size = s->frame_size * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = s->frame_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
output = (int16_t *)s->frame.data[0];
/* if there is not enough data left for the smallest possible frame,
reset the libspeex buffer using the current packet, otherwise ignore
@ -116,7 +120,7 @@ static int libspeex_decode_frame(AVCodecContext *avctx,
if (speex_bits_remaining(&s->bits) < 43) {
/* check for flush packet */
if (!buf || !buf_size) {
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
/* set new buffer */
@ -133,7 +137,9 @@ static int libspeex_decode_frame(AVCodecContext *avctx,
if (avctx->channels == 2)
speex_decode_stereo_int(output, s->frame_size, &s->stereo);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return consumed;
}
@ -163,6 +169,6 @@ AVCodec ff_libspeex_decoder = {
.close = libspeex_decode_close,
.decode = libspeex_decode_frame,
.flush = libspeex_decode_flush,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
};

33
libavcodec/mace.c
View File

@ -153,6 +153,7 @@ typedef struct ChannelData {
} ChannelData;
typedef struct MACEContext {
AVFrame frame;
ChannelData chd[2];
} MACEContext;
@ -228,30 +229,35 @@ static void chomp6(ChannelData *chd, int16_t *output, uint8_t val,
static av_cold int mace_decode_init(AVCodecContext * avctx)
{
MACEContext *ctx = avctx->priv_data;
if (avctx->channels > 2)
return -1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
return 0;
}
static int mace_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int mace_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples = data;
int16_t *samples;
MACEContext *ctx = avctx->priv_data;
int i, j, k, l;
int out_size;
int i, j, k, l, ret;
int is_mace3 = (avctx->codec_id == CODEC_ID_MACE3);
out_size = 3 * (buf_size << (1 - is_mace3)) *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
ctx->frame.nb_samples = 3 * (buf_size << (1 - is_mace3)) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)ctx->frame.data[0];
for(i = 0; i < avctx->channels; i++) {
int16_t *output = samples + i;
@ -277,7 +283,8 @@ static int mace_decode_frame(AVCodecContext *avctx,
}
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->frame;
return buf_size;
}
@ -289,6 +296,7 @@ AVCodec ff_mace3_decoder = {
.priv_data_size = sizeof(MACEContext),
.init = mace_decode_init,
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 3:1"),
};
@ -299,6 +307,7 @@ AVCodec ff_mace6_decoder = {
.priv_data_size = sizeof(MACEContext),
.init = mace_decode_init,
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 6:1"),
};

39
libavcodec/mlpdec.c
View File

@ -120,6 +120,7 @@ typedef struct SubStream {
typedef struct MLPDecodeContext {
AVCodecContext *avctx;
AVFrame frame;
//! Current access unit being read has a major sync.
int is_major_sync_unit;
@ -239,6 +240,9 @@ static av_cold int mlp_decode_init(AVCodecContext *avctx)
m->substream[substr].lossless_check_data = 0xffffffff;
dsputil_init(&m->dsp, avctx);
avcodec_get_frame_defaults(&m->frame);
avctx->coded_frame = &m->frame;
return 0;
}
@ -905,13 +909,14 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
/** Write the audio data into the output buffer. */
static int output_data(MLPDecodeContext *m, unsigned int substr,
uint8_t *data, unsigned int *data_size)
void *data, int *got_frame_ptr)
{
AVCodecContext *avctx = m->avctx;
SubStream *s = &m->substream[substr];
unsigned int i, out_ch = 0;
int out_size;
int32_t *data_32 = (int32_t*) data;
int16_t *data_16 = (int16_t*) data;
int32_t *data_32;
int16_t *data_16;
int ret;
int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
if (m->avctx->channels != s->max_matrix_channel + 1) {
@ -919,11 +924,14 @@ static int output_data(MLPDecodeContext *m, unsigned int substr,
return AVERROR_INVALIDDATA;
}
out_size = s->blockpos * m->avctx->channels *
av_get_bytes_per_sample(m->avctx->sample_fmt);
if (*data_size < out_size)
return AVERROR(EINVAL);
/* get output buffer */
m->frame.nb_samples = s->blockpos;
if ((ret = avctx->get_buffer(avctx, &m->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
data_32 = (int32_t *)m->frame.data[0];
data_16 = (int16_t *)m->frame.data[0];
for (i = 0; i < s->blockpos; i++) {
for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
@ -936,7 +944,8 @@ static int output_data(MLPDecodeContext *m, unsigned int substr,
}
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = m->frame;
return 0;
}
@ -945,8 +954,8 @@ static int output_data(MLPDecodeContext *m, unsigned int substr,
* @return negative on error, 0 if not enough data is present in the input stream,
* otherwise the number of bytes consumed. */
static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
AVPacket *avpkt)
static int read_access_unit(AVCodecContext *avctx, void* data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -982,7 +991,7 @@ static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
if (!m->params_valid) {
av_log(m->avctx, AV_LOG_WARNING,
"Stream parameters not seen; skipping frame.\n");
*data_size = 0;
*got_frame_ptr = 0;
return length;
}
@ -1127,7 +1136,7 @@ next_substr:
rematrix_channels(m, m->max_decoded_substream);
if ((ret = output_data(m, m->max_decoded_substream, data, data_size)) < 0)
if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
return ret;
return length;
@ -1148,6 +1157,7 @@ AVCodec ff_mlp_decoder = {
.priv_data_size = sizeof(MLPDecodeContext),
.init = mlp_decode_init,
.decode = read_access_unit,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
};
@ -1159,6 +1169,7 @@ AVCodec ff_truehd_decoder = {
.priv_data_size = sizeof(MLPDecodeContext),
.init = mlp_decode_init,
.decode = read_access_unit,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
};
#endif /* CONFIG_TRUEHD_DECODER */

1
libavcodec/mpc.h
View File

@ -50,6 +50,7 @@ typedef struct {
}Band;
typedef struct {
AVFrame frame;
DSPContext dsp;
MPADSPContext mpadsp;
GetBitContext gb;

29
libavcodec/mpc7.c
View File

@ -136,6 +136,10 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx)
}
}
vlc_initialized = 1;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
@ -192,9 +196,8 @@ static int get_scale_idx(GetBitContext *gb, int ref)
return ref + t;
}
static int mpc7_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -204,7 +207,7 @@ static int mpc7_decode_frame(AVCodecContext * avctx,
int i, ch;
int mb = -1;
Band *bands = c->bands;
int off, out_size;
int off, ret;
int bits_used, bits_avail;
memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
@ -213,10 +216,11 @@ static int mpc7_decode_frame(AVCodecContext * avctx,
return AVERROR(EINVAL);
}
out_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4;
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = buf[1] ? c->lastframelen : MPC_FRAME_SIZE;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE);
@ -276,7 +280,7 @@ static int mpc7_decode_frame(AVCodecContext * avctx,
for(ch = 0; ch < 2; ch++)
idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
ff_mpc_dequantize_and_synth(c, mb, data, 2);
ff_mpc_dequantize_and_synth(c, mb, c->frame.data[0], 2);
av_free(bits);
@ -288,10 +292,12 @@ static int mpc7_decode_frame(AVCodecContext * avctx,
}
if(c->frames_to_skip){
c->frames_to_skip--;
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return buf_size;
}
@ -312,5 +318,6 @@ AVCodec ff_mpc7_decoder = {
.init = mpc7_decode_init,
.decode = mpc7_decode_frame,
.flush = mpc7_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
};

27
libavcodec/mpc8.c
View File

@ -228,12 +228,15 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx)
&mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
}
vlc_initialized = 1;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
static int mpc8_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int mpc8_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -241,14 +244,15 @@ static int mpc8_decode_frame(AVCodecContext * avctx,
GetBitContext gb2, *gb = &gb2;
int i, j, k, ch, cnt, res, t;
Band *bands = c->bands;
int off, out_size;
int off;
int maxband, keyframe;
int last[2];
out_size = MPC_FRAME_SIZE * 2 * avctx->channels;
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = MPC_FRAME_SIZE;
if ((res = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return res;
}
keyframe = c->cur_frame == 0;
@ -401,14 +405,16 @@ static int mpc8_decode_frame(AVCodecContext * avctx,
}
}
ff_mpc_dequantize_and_synth(c, maxband, data, avctx->channels);
ff_mpc_dequantize_and_synth(c, maxband, c->frame.data[0], avctx->channels);
c->cur_frame++;
c->last_bits_used = get_bits_count(gb);
if(c->cur_frame >= c->frames)
c->cur_frame = 0;
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return c->cur_frame ? c->last_bits_used >> 3 : buf_size;
}
@ -420,5 +426,6 @@ AVCodec ff_mpc8_decoder = {
.priv_data_size = sizeof(MPCContext),
.init = mpc8_decode_init,
.decode = mpc8_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
};

86
libavcodec/mpegaudiodec.c
View File

@ -79,6 +79,7 @@ typedef struct MPADecodeContext {
int err_recognition;
AVCodecContext* avctx;
MPADSPContext mpadsp;
AVFrame frame;
} MPADecodeContext;
#if CONFIG_FLOAT
@ -474,6 +475,10 @@ static av_cold int decode_init(AVCodecContext * avctx)
if (avctx->codec_id == CODEC_ID_MP3ADU)
s->adu_mode = 1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -1695,7 +1700,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
const uint8_t *buf, int buf_size)
{
int i, nb_frames, ch;
int i, nb_frames, ch, ret;
OUT_INT *samples_ptr;
init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
@ -1743,8 +1748,16 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
assert(i <= buf_size - HEADER_SIZE && i >= 0);
memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
s->last_buf_size += i;
}
break;
/* get output buffer */
if (!samples) {
s->frame.nb_samples = s->avctx->frame_size;
if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (OUT_INT *)s->frame.data[0];
}
/* apply the synthesis filter */
@ -1764,7 +1777,7 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}
static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
@ -1772,7 +1785,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int out_size;
OUT_INT *out_samples = data;
if (buf_size < HEADER_SIZE)
return AVERROR_INVALIDDATA;
@ -1795,10 +1807,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
return AVERROR(EINVAL);
*data_size = 0;
if (s->frame_size <= 0 || s->frame_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
@ -1807,9 +1815,10 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
buf_size= s->frame_size;
}
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
out_size = mp_decode_frame(s, NULL, buf, buf_size);
if (out_size >= 0) {
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
avctx->sample_rate = s->sample_rate;
//FIXME maybe move the other codec info stuff from above here too
} else {
@ -1818,6 +1827,7 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
If there is more data in the packet, just consume the bad frame
instead of returning an error, which would discard the whole
packet. */
*got_frame_ptr = 0;
if (buf_size == avpkt->size)
return out_size;
}
@ -1833,15 +1843,14 @@ static void flush(AVCodecContext *avctx)
}
#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int decode_frame_adu(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int len, out_size;
OUT_INT *out_samples = data;
len = buf_size;
@ -1871,9 +1880,6 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
return AVERROR(EINVAL);
s->frame_size = len;
#if FF_API_PARSE_FRAME
@ -1881,9 +1887,11 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
out_size = buf_size;
else
#endif
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
out_size = mp_decode_frame(s, NULL, buf, buf_size);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
*data_size = out_size;
return buf_size;
}
#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
@ -1894,6 +1902,7 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
* Context for MP3On4 decoder
*/
typedef struct MP3On4DecodeContext {
AVFrame *frame;
int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
int syncword; ///< syncword patch
const uint8_t *coff; ///< channel offsets in output buffer
@ -1984,6 +1993,7 @@ static int decode_init_mp3on4(AVCodecContext * avctx)
// Put decoder context in place to make init_decode() happy
avctx->priv_data = s->mp3decctx[0];
decode_init(avctx);
s->frame = avctx->coded_frame;
// Restore mp3on4 context pointer
avctx->priv_data = s;
s->mp3decctx[0]->adu_mode = 1; // Set adu mode
@ -2028,9 +2038,8 @@ static void flush_mp3on4(AVCodecContext *avctx)
}
static int decode_frame_mp3on4(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -2038,14 +2047,17 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
MPADecodeContext *m;
int fsize, len = buf_size, out_size = 0;
uint32_t header;
OUT_INT *out_samples = data;
OUT_INT *out_samples;
OUT_INT *outptr, *bp;
int fr, j, n, ch;
int fr, j, n, ch, ret;
if (*data_size < MPA_FRAME_SIZE * avctx->channels * sizeof(OUT_INT)) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame->nb_samples = MPA_FRAME_SIZE;
if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_samples = (OUT_INT *)s->frame->data[0];
// Discard too short frames
if (buf_size < HEADER_SIZE)
@ -2104,7 +2116,10 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
/* update codec info */
avctx->sample_rate = s->mp3decctx[0]->sample_rate;
*data_size = out_size;
s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
*got_frame_ptr = 1;
*(AVFrame *)data = *s->frame;
return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
@ -2119,7 +2134,9 @@ AVCodec ff_mp1_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
@ -2134,7 +2151,9 @@ AVCodec ff_mp2_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
@ -2149,7 +2168,9 @@ AVCodec ff_mp3_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
@ -2164,7 +2185,9 @@ AVCodec ff_mp3adu_decoder = {
.init = decode_init,
.decode = decode_frame_adu,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
@ -2179,6 +2202,7 @@ AVCodec ff_mp3on4_decoder = {
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
};

17
libavcodec/mpegaudiodec_float.c
View File

@ -31,7 +31,9 @@ AVCodec ff_mp1float_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
@ -46,7 +48,9 @@ AVCodec ff_mp2float_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
@ -61,7 +65,9 @@ AVCodec ff_mp3float_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
@ -76,7 +82,9 @@ AVCodec ff_mp3adufloat_decoder = {
.init = decode_init,
.decode = decode_frame_adu,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
@ -91,6 +99,7 @@ AVCodec ff_mp3on4float_decoder = {
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
};

37
libavcodec/nellymoserdec.c
View File

@ -47,6 +47,7 @@
typedef struct NellyMoserDecodeContext {
AVCodecContext* avctx;
AVFrame frame;
float *float_buf;
DECLARE_ALIGNED(16, float, state)[NELLY_BUF_LEN];
AVLFG random_state;
@ -142,29 +143,28 @@ static av_cold int decode_init(AVCodecContext * avctx) {
ff_init_ff_sine_windows(7);
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int decode_tag(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt) {
static int decode_tag(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
NellyMoserDecodeContext *s = avctx->priv_data;
int blocks, i, block_size;
int16_t *samples_s16 = data;
float *samples_flt = data;
int blocks, i, ret;
int16_t *samples_s16;
float *samples_flt;
block_size = NELLY_SAMPLES * av_get_bytes_per_sample(avctx->sample_fmt);
blocks = buf_size / NELLY_BLOCK_LEN;
if (blocks <= 0) {
av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
return AVERROR_INVALIDDATA;
}
if (*data_size < blocks * block_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
if (buf_size % NELLY_BLOCK_LEN) {
av_log(avctx, AV_LOG_WARNING, "Leftover bytes: %d.\n",
buf_size % NELLY_BLOCK_LEN);
@ -177,6 +177,15 @@ static int decode_tag(AVCodecContext * avctx,
* 44100 Hz - 8
*/
/* get output buffer */
s->frame.nb_samples = NELLY_SAMPLES * blocks;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples_s16 = (int16_t *)s->frame.data[0];
samples_flt = (float *)s->frame.data[0];
for (i=0 ; i<blocks ; i++) {
if (avctx->sample_fmt == SAMPLE_FMT_FLT) {
nelly_decode_block(s, buf, samples_flt);
@ -188,7 +197,9 @@ static int decode_tag(AVCodecContext * avctx,
}
buf += NELLY_BLOCK_LEN;
}
*data_size = blocks * block_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -198,6 +209,7 @@ static av_cold int decode_end(AVCodecContext * avctx) {
av_freep(&s->float_buf);
ff_mdct_end(&s->imdct_ctx);
return 0;
}
@ -209,6 +221,7 @@ AVCodec ff_nellymoser_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_tag,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,

42
libavcodec/pcm.c
View File

@ -192,6 +192,7 @@ static int pcm_encode_frame(AVCodecContext *avctx,
}
typedef struct PCMDecode {
AVFrame frame;
short table[256];
} PCMDecode;
@ -223,6 +224,9 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx)
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
avctx->bits_per_raw_sample = av_get_bits_per_sample(avctx->codec->id);
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -243,22 +247,20 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx)
dst += size / 8; \
}
static int pcm_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int pcm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *src = avpkt->data;
int buf_size = avpkt->size;
PCMDecode *s = avctx->priv_data;
int sample_size, c, n, out_size;
int sample_size, c, n, ret, samples_per_block;
uint8_t *samples;
int32_t *dst_int32_t;
samples = data;
sample_size = av_get_bits_per_sample(avctx->codec_id)/8;
/* av_get_bits_per_sample returns 0 for CODEC_ID_PCM_DVD */
samples_per_block = 1;
if (CODEC_ID_PCM_DVD == avctx->codec_id) {
if (avctx->bits_per_coded_sample != 20 &&
avctx->bits_per_coded_sample != 24) {
@ -266,10 +268,13 @@ static int pcm_decode_frame(AVCodecContext *avctx,
return AVERROR(EINVAL);
}
/* 2 samples are interleaved per block in PCM_DVD */
samples_per_block = 2;
sample_size = avctx->bits_per_coded_sample * 2 / 8;
} else if (avctx->codec_id == CODEC_ID_PCM_LXF)
} else if (avctx->codec_id == CODEC_ID_PCM_LXF) {
/* we process 40-bit blocks per channel for LXF */
samples_per_block = 2;
sample_size = 5;
}
if (sample_size == 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
@ -288,14 +293,13 @@ static int pcm_decode_frame(AVCodecContext *avctx,
n = buf_size/sample_size;
out_size = n * av_get_bytes_per_sample(avctx->sample_fmt);
if (avctx->codec_id == CODEC_ID_PCM_DVD ||
avctx->codec_id == CODEC_ID_PCM_LXF)
out_size *= 2;
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = n * samples_per_block / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = s->frame.data[0];
switch(avctx->codec->id) {
case CODEC_ID_PCM_U32LE:
@ -401,7 +405,7 @@ static int pcm_decode_frame(AVCodecContext *avctx,
case CODEC_ID_PCM_DVD:
{
const uint8_t *src8;
dst_int32_t = data;
dst_int32_t = (int32_t *)s->frame.data[0];
n /= avctx->channels;
switch (avctx->bits_per_coded_sample) {
case 20:
@ -433,7 +437,7 @@ static int pcm_decode_frame(AVCodecContext *avctx,
{
int i;
const uint8_t *src8;
dst_int32_t = data;
dst_int32_t = (int32_t *)s->frame.data[0];
n /= avctx->channels;
//unpack and de-planerize
for (i = 0; i < n; i++) {
@ -454,7 +458,10 @@ static int pcm_decode_frame(AVCodecContext *avctx,
default:
return -1;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -483,6 +490,7 @@ AVCodec ff_ ## name_ ## _decoder = { \
.priv_data_size = sizeof(PCMDecode), \
.init = pcm_decode_init, \
.decode = pcm_decode_frame, \
.capabilities = CODEC_CAP_DR1, \
.sample_fmts = (const enum AVSampleFormat[]){sample_fmt_,AV_SAMPLE_FMT_NONE}, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
}

26
libavcodec/qcelpdec.c
View File

@ -56,6 +56,7 @@ typedef enum
typedef struct
{
AVFrame avframe;
GetBitContext gb;
qcelp_packet_rate bitrate;
QCELPFrame frame; /**< unpacked data frame */
@ -97,6 +98,9 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx)
for(i=0; i<10; i++)
q->prev_lspf[i] = (i+1)/11.;
avcodec_get_frame_defaults(&q->avframe);
avctx->coded_frame = &q->avframe;
return 0;
}
@ -682,23 +686,25 @@ static void postfilter(QCELPContext *q, float *samples, float *lpc)
160, 0.9375, &q->postfilter_agc_mem);
}
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QCELPContext *q = avctx->priv_data;
float *outbuffer = data;
int i, out_size;
float *outbuffer;
int i, ret;
float quantized_lspf[10], lpc[10];
float gain[16];
float *formant_mem;
out_size = 160 * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
q->avframe.nb_samples = 160;
if ((ret = avctx->get_buffer(avctx, &q->avframe)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
outbuffer = (float *)q->avframe.data[0];
if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
@ -783,7 +789,8 @@ erasure:
memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
q->prev_bitrate = q->bitrate;
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = q->avframe;
return buf_size;
}
@ -795,6 +802,7 @@ AVCodec ff_qcelp_decoder =
.id = CODEC_ID_QCELP,
.init = qcelp_decode_init,
.decode = qcelp_decode_frame,
.capabilities = CODEC_CAP_DR1,
.priv_data_size = sizeof(QCELPContext),
.long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
};

32
libavcodec/qdm2.c
View File

@ -130,6 +130,8 @@ typedef struct {
* QDM2 decoder context
*/
typedef struct {
AVFrame frame;
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
@ -1875,6 +1877,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
// dump_context(s);
return 0;
}
@ -1952,30 +1957,27 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
}
static int qdm2_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
int16_t *out = data;
int i, out_size;
int16_t *out;
int i, ret;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
out_size = 16 * s->channels * s->frame_size *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = 16 * s->frame_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
buf_size, buf, s->checksum_size, data, *data_size);
out = (int16_t *)s->frame.data[0];
for (i = 0; i < 16; i++) {
if (qdm2_decode(s, buf, out) < 0)
@ -1983,7 +1985,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx,
out += s->channels * s->frame_size;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return s->checksum_size;
}
@ -1997,5 +2000,6 @@ AVCodec ff_qdm2_decoder =
.init = qdm2_decode_init,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};

1
libavcodec/ra144.h
View File

@ -34,6 +34,7 @@
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
LPCContext lpc_ctx;
unsigned int old_energy; ///< previous frame energy

31
libavcodec/ra144dec.c
View File

@ -38,6 +38,10 @@ static av_cold int ra144_decode_init(AVCodecContext * avctx)
ractx->lpc_coef[1] = ractx->lpc_tables[1];
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&ractx->frame);
avctx->coded_frame = &ractx->frame;
return 0;
}
@ -54,8 +58,8 @@ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
}
/** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
int *data_size, AVPacket *avpkt)
static int ra144_decode_frame(AVCodecContext * avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@ -64,23 +68,25 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
uint16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
int i, j;
int out_size;
int16_t *data = vdata;
int ret;
int16_t *samples;
unsigned int energy;
RA144Context *ractx = avctx->priv_data;
GetBitContext gb;
out_size = NBLOCKS * BLOCKSIZE * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
ractx->frame.nb_samples = NBLOCKS * BLOCKSIZE;
if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)ractx->frame.data[0];
if(buf_size < FRAMESIZE) {
av_log(avctx, AV_LOG_ERROR,
"Frame too small (%d bytes). Truncated file?\n", buf_size);
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
init_get_bits(&gb, buf, FRAMESIZE * 8);
@ -106,7 +112,7 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
for (j=0; j < BLOCKSIZE; j++)
*data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
*samples++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2);
}
ractx->old_energy = energy;
@ -114,7 +120,9 @@ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = ractx->frame;
return FRAMESIZE;
}
@ -125,5 +133,6 @@ AVCodec ff_ra_144_decoder = {
.priv_data_size = sizeof(RA144Context),
.init = ra144_decode_init,
.decode = ra144_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};

27
libavcodec/ra288.c
View File

@ -36,6 +36,7 @@
#define RA288_BLOCKS_PER_FRAME 32
typedef struct {
AVFrame frame;
DSPContext dsp;
DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A)
DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB)
@ -62,6 +63,10 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx)
RA288Context *ractx = avctx->priv_data;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
dsputil_init(&ractx->dsp, avctx);
avcodec_get_frame_defaults(&ractx->frame);
avctx->coded_frame = &ractx->frame;
return 0;
}
@ -165,12 +170,12 @@ static void backward_filter(RA288Context *ractx,
}
static int ra288_decode_frame(AVCodecContext * avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *out = data;
int i, out_size;
float *out;
int i, ret;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
@ -181,12 +186,13 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data,
return AVERROR_INVALIDDATA;
}
out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out = (float *)ractx->frame.data[0];
init_get_bits(&gb, buf, avctx->block_align * 8);
@ -208,7 +214,9 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data,
}
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = ractx->frame;
return avctx->block_align;
}
@ -219,5 +227,6 @@ AVCodec ff_ra_288_decoder = {
.priv_data_size = sizeof(RA288Context),
.init = ra288_decode_init,
.decode = ra288_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
};

46
libavcodec/s302m.c
View File

@ -25,6 +25,10 @@
#define AES3_HEADER_LEN 4
typedef struct S302MDecodeContext {
AVFrame frame;
} S302MDecodeContext;
static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
int buf_size)
{
@ -73,10 +77,12 @@ static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
}
static int s302m_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
S302MDecodeContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int block_size, ret;
int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
if (frame_size < 0)
@ -85,11 +91,18 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
buf_size -= AES3_HEADER_LEN;
buf += AES3_HEADER_LEN;
if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
return -1;
/* get output buffer */
block_size = (avctx->bits_per_coded_sample + 4) / 4;
s->frame.nb_samples = 2 * (buf_size / block_size) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
buf_size = (s->frame.nb_samples * avctx->channels / 2) * block_size;
if (avctx->bits_per_coded_sample == 24) {
uint32_t *o = data;
uint32_t *o = (uint32_t *)s->frame.data[0];
for (; buf_size > 6; buf_size -= 7) {
*o++ = (av_reverse[buf[2]] << 24) |
(av_reverse[buf[1]] << 16) |
@ -100,9 +113,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
(av_reverse[buf[3] & 0x0f] << 4);
buf += 7;
}
*data_size = (uint8_t*) o - (uint8_t*) data;
} else if (avctx->bits_per_coded_sample == 20) {
uint32_t *o = data;
uint32_t *o = (uint32_t *)s->frame.data[0];
for (; buf_size > 5; buf_size -= 6) {
*o++ = (av_reverse[buf[2] & 0xf0] << 28) |
(av_reverse[buf[1]] << 20) |
@ -112,9 +124,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
(av_reverse[buf[3]] << 12);
buf += 6;
}
*data_size = (uint8_t*) o - (uint8_t*) data;
} else {
uint16_t *o = data;
uint16_t *o = (uint16_t *)s->frame.data[0];
for (; buf_size > 4; buf_size -= 5) {
*o++ = (av_reverse[buf[1]] << 8) |
av_reverse[buf[0]];
@ -123,10 +134,22 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
(av_reverse[buf[2]] >> 4);
buf += 5;
}
*data_size = (uint8_t*) o - (uint8_t*) data;
}
return buf - avpkt->data;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return avpkt->size;
}
static int s302m_decode_init(AVCodecContext *avctx)
{
S302MDecodeContext *s = avctx->priv_data;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -134,6 +157,9 @@ AVCodec ff_s302m_decoder = {
.name = "s302m",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_S302M,
.priv_data_size = sizeof(S302MDecodeContext),
.init = s302m_decode_init,
.decode = s302m_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
};

45
libavcodec/shorten.c
View File

@ -79,6 +79,7 @@ static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
typedef struct ShortenContext {
AVCodecContext *avctx;
AVFrame frame;
GetBitContext gb;
int min_framesize, max_framesize;
@ -112,6 +113,9 @@ static av_cold int shorten_decode_init(AVCodecContext * avctx)
s->avctx = avctx;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -394,15 +398,13 @@ static int read_header(ShortenContext *s)
return 0;
}
static int shorten_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int shorten_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ShortenContext *s = avctx->priv_data;
int i, input_buf_size = 0;
int16_t *samples = data;
int ret;
/* allocate internal bitstream buffer */
@ -436,7 +438,7 @@ static int shorten_decode_frame(AVCodecContext *avctx,
/* do not decode until buffer has at least max_framesize bytes or
the end of the file has been reached */
if (buf_size < s->max_framesize && avpkt->data) {
*data_size = 0;
*got_frame_ptr = 0;
return input_buf_size;
}
}
@ -448,13 +450,13 @@ static int shorten_decode_frame(AVCodecContext *avctx,
if (!s->got_header) {
if ((ret = read_header(s)) < 0)
return ret;
*data_size = 0;
*got_frame_ptr = 0;
goto finish_frame;
}
/* if quit command was read previously, don't decode anything */
if (s->got_quit_command) {
*data_size = 0;
*got_frame_ptr = 0;
return avpkt->size;
}
@ -464,7 +466,7 @@ static int shorten_decode_frame(AVCodecContext *avctx,
int len;
if (get_bits_left(&s->gb) < 3+FNSIZE) {
*data_size = 0;
*got_frame_ptr = 0;
break;
}
@ -472,7 +474,7 @@ static int shorten_decode_frame(AVCodecContext *avctx,
if (cmd > FN_VERBATIM) {
av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
*data_size = 0;
*got_frame_ptr = 0;
break;
}
@ -507,7 +509,7 @@ static int shorten_decode_frame(AVCodecContext *avctx,
break;
}
if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
*data_size = 0;
*got_frame_ptr = 0;
break;
}
} else {
@ -571,19 +573,23 @@ static int shorten_decode_frame(AVCodecContext *avctx,
/* if this is the last channel in the block, output the samples */
s->cur_chan++;
if (s->cur_chan == s->channels) {
int out_size = s->blocksize * s->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame.nb_samples = s->blocksize;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
interleave_buffer(samples, s->channels, s->blocksize, s->decoded);
*data_size = out_size;
/* interleave output */
interleave_buffer((int16_t *)s->frame.data[0], s->channels,
s->blocksize, s->decoded);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
}
}
}
if (s->cur_chan < s->channels)
*data_size = 0;
*got_frame_ptr = 0;
finish_frame:
s->bitindex = get_bits_count(&s->gb) - 8*((get_bits_count(&s->gb))/8);
@ -614,6 +620,7 @@ static av_cold int shorten_decode_close(AVCodecContext *avctx)
}
av_freep(&s->bitstream);
av_freep(&s->coeffs);
return 0;
}
@ -625,6 +632,6 @@ AVCodec ff_shorten_decoder = {
.init = shorten_decode_init,
.close = shorten_decode_close,
.decode = shorten_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("Shorten"),
};

34
libavcodec/sipr.c
View File

@ -507,20 +507,23 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx)
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
return 0;
}
static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
int *data_size, AVPacket *avpkt)
static int sipr_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
SiprContext *ctx = avctx->priv_data;
const uint8_t *buf=avpkt->data;
SiprParameters parm;
const SiprModeParam *mode_par = &modes[ctx->mode];
GetBitContext gb;
float *data = datap;
float *samples;
int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
int i, out_size;
int i, ret;
ctx->avctx = avctx;
if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
@ -530,27 +533,27 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
return -1;
}
out_size = mode_par->frames_per_packet * subframe_size *
mode_par->subframe_count *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR,
"Error processing packet: output buffer (%d) too small\n",
*data_size);
return -1;
/* get output buffer */
ctx->frame.nb_samples = mode_par->frames_per_packet * subframe_size *
mode_par->subframe_count;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)ctx->frame.data[0];
init_get_bits(&gb, buf, mode_par->bits_per_frame);
for (i = 0; i < mode_par->frames_per_packet; i++) {
decode_parameters(&parm, &gb, mode_par);
ctx->decode_frame(ctx, &parm, data);
ctx->decode_frame(ctx, &parm, samples);
data += subframe_size * mode_par->subframe_count;
samples += subframe_size * mode_par->subframe_count;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->frame;
return mode_par->bits_per_frame >> 3;
}
@ -562,5 +565,6 @@ AVCodec ff_sipr_decoder = {
.priv_data_size = sizeof(SiprContext),
.init = sipr_decoder_init,
.decode = sipr_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
};

41
libavcodec/smacker.c
View File

@ -558,31 +558,43 @@ static av_cold int decode_end(AVCodecContext *avctx)
}
typedef struct SmackerAudioContext {
AVFrame frame;
} SmackerAudioContext;
static av_cold int smka_decode_init(AVCodecContext *avctx)
{
SmackerAudioContext *s = avctx->priv_data;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avctx->sample_fmt = avctx->bits_per_coded_sample == 8 ? AV_SAMPLE_FMT_U8 : AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
/**
* Decode Smacker audio data
*/
static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
static int smka_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
SmackerAudioContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
GetBitContext gb;
HuffContext h[4];
VLC vlc[4];
int16_t *samples = data;
uint8_t *samples8 = data;
int16_t *samples;
uint8_t *samples8;
int val;
int i, res;
int i, res, ret;
int unp_size;
int bits, stereo;
int pred[2] = {0, 0};
@ -598,15 +610,11 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
if(!get_bits1(&gb)){
av_log(avctx, AV_LOG_INFO, "Sound: no data\n");
*data_size = 0;
*got_frame_ptr = 0;
return 1;
}
stereo = get_bits1(&gb);
bits = get_bits1(&gb);
if (unp_size & 0xC0000000 || unp_size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n");
return -1;
}
if (stereo ^ (avctx->channels != 1)) {
av_log(avctx, AV_LOG_ERROR, "channels mismatch\n");
return AVERROR(EINVAL);
@ -616,6 +624,15 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
return AVERROR(EINVAL);
}
/* get output buffer */
s->frame.nb_samples = unp_size / (avctx->channels * (bits + 1));
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
samples8 = s->frame.data[0];
memset(vlc, 0, sizeof(VLC) * 4);
memset(h, 0, sizeof(HuffContext) * 4);
// Initialize
@ -705,7 +722,9 @@ static int smka_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
av_free(h[i].values);
}
*data_size = unp_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -725,8 +744,10 @@ AVCodec ff_smackaud_decoder = {
.name = "smackaud",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_SMACKAUDIO,
.priv_data_size = sizeof(SmackerAudioContext),
.init = smka_decode_init,
.decode = smka_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Smacker audio"),
};

29
libavcodec/truespeech.c
View File

@ -34,6 +34,7 @@
* TrueSpeech decoder context
*/
typedef struct {
AVFrame frame;
DSPContext dsp;
/* input data */
uint8_t buffer[32];
@ -69,6 +70,9 @@ static av_cold int truespeech_decode_init(AVCodecContext * avctx)
dsputil_init(&c->dsp, avctx);
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
@ -299,17 +303,16 @@ static void truespeech_save_prevvec(TSContext *c)
c->prevfilt[i] = c->cvector[i];
}
static int truespeech_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
TSContext *c = avctx->priv_data;
int i, j;
short *samples = data;
int iterations, out_size;
int16_t *samples;
int iterations, ret;
iterations = buf_size / 32;
@ -319,13 +322,15 @@ static int truespeech_decode_frame(AVCodecContext *avctx,
return -1;
}
out_size = iterations * 240 * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
c->frame.nb_samples = iterations * 240;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
memset(samples, 0, out_size);
memset(samples, 0, iterations * 240 * sizeof(*samples));
for(j = 0; j < iterations; j++) {
truespeech_read_frame(c, buf);
@ -345,7 +350,8 @@ static int truespeech_decode_frame(AVCodecContext *avctx,
truespeech_save_prevvec(c);
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return buf_size;
}
@ -357,5 +363,6 @@ AVCodec ff_truespeech_decoder = {
.priv_data_size = sizeof(TSContext),
.init = truespeech_decode_init,
.decode = truespeech_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
};

28
libavcodec/tta.c
View File

@ -56,6 +56,7 @@ typedef struct TTAChannel {
typedef struct TTAContext {
AVCodecContext *avctx;
AVFrame frame;
GetBitContext gb;
int format, channels, bps, data_length;
@ -276,17 +277,19 @@ static av_cold int tta_decode_init(AVCodecContext * avctx)
return -1;
}
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int tta_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int tta_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
TTAContext *s = avctx->priv_data;
int i, out_size;
int i, ret;
int cur_chan = 0, framelen = s->frame_length;
int32_t *p;
@ -297,10 +300,11 @@ static int tta_decode_frame(AVCodecContext *avctx,
if (!s->total_frames && s->last_frame_length)
framelen = s->last_frame_length;
out_size = framelen * s->channels * av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer size is too small.\n");
return -1;
/* get output buffer */
s->frame.nb_samples = framelen;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
// decode directly to output buffer for 24-bit sample format
@ -396,19 +400,20 @@ static int tta_decode_frame(AVCodecContext *avctx,
// convert to output buffer
if (s->bps == 2) {
int16_t *samples = data;
int16_t *samples = (int16_t *)s->frame.data[0];
for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++)
*samples++ = *p;
} else {
// shift samples for 24-bit sample format
int32_t *samples = data;
int32_t *samples = (int32_t *)s->frame.data[0];
for (i = 0; i < framelen * s->channels; i++)
*samples++ <<= 8;
// reset decode buffer
s->decode_buffer = NULL;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -430,5 +435,6 @@ AVCodec ff_tta_decoder = {
.init = tta_decode_init,
.close = tta_decode_close,
.decode = tta_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("True Audio (TTA)"),
};

36
libavcodec/twinvq.c
View File

@ -174,6 +174,7 @@ static const ModeTab mode_44_48 = {
typedef struct TwinContext {
AVCodecContext *avctx;
AVFrame frame;
DSPContext dsp;
FFTContext mdct_ctx[3];
@ -195,6 +196,7 @@ typedef struct TwinContext {
float *curr_frame; ///< non-interleaved output
float *prev_frame; ///< non-interleaved previous frame
int last_block_pos[2];
int discarded_packets;
float *cos_tabs[3];
@ -676,6 +678,9 @@ static void imdct_output(TwinContext *tctx, enum FrameType ftype, int wtype,
i);
}
if (!out)
return;
size2 = tctx->last_block_pos[0];
size1 = mtab->size - size2;
if (tctx->avctx->channels == 2) {
@ -811,16 +816,16 @@ static void read_and_decode_spectrum(TwinContext *tctx, GetBitContext *gb,
}
static int twin_decode_frame(AVCodecContext * avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
TwinContext *tctx = avctx->priv_data;
GetBitContext gb;
const ModeTab *mtab = tctx->mtab;
float *out = data;
float *out = NULL;
enum FrameType ftype;
int window_type, out_size;
int window_type, ret;
static const enum FrameType wtype_to_ftype_table[] = {
FT_LONG, FT_LONG, FT_SHORT, FT_LONG,
FT_MEDIUM, FT_LONG, FT_LONG, FT_MEDIUM, FT_MEDIUM
@ -832,11 +837,14 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data,
return AVERROR(EINVAL);
}
out_size = mtab->size * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
if (tctx->discarded_packets >= 2) {
tctx->frame.nb_samples = mtab->size;
if ((ret = avctx->get_buffer(avctx, &tctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out = (float *)tctx->frame.data[0];
}
init_get_bits(&gb, buf, buf_size * 8);
@ -856,12 +864,14 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data,
FFSWAP(float*, tctx->curr_frame, tctx->prev_frame);
if (tctx->avctx->frame_number < 2) {
*data_size=0;
if (tctx->discarded_packets < 2) {
tctx->discarded_packets++;
*got_frame_ptr = 0;
return buf_size;
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = tctx->frame;;
return buf_size;
}
@ -1153,6 +1163,9 @@ static av_cold int twin_decode_init(AVCodecContext *avctx)
memset_float(tctx->bark_hist[0][0], 0.1, FF_ARRAY_ELEMS(tctx->bark_hist));
avcodec_get_frame_defaults(&tctx->frame);
avctx->coded_frame = &tctx->frame;
return 0;
}
@ -1164,5 +1177,6 @@ AVCodec ff_twinvq_decoder = {
.init = twin_decode_init,
.close = twin_decode_close,
.decode = twin_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("VQF TwinVQ"),
};

228
libavcodec/utils.c
View File

@ -222,9 +222,8 @@ void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height,
if(s->codec_id == CODEC_ID_SVQ1 || s->codec_id == CODEC_ID_VP5 ||
s->codec_id == CODEC_ID_VP6 || s->codec_id == CODEC_ID_VP6F ||
s->codec_id == CODEC_ID_VP6A) {
linesize_align[0] =
linesize_align[1] =
linesize_align[2] = 16;
for (i = 0; i < AV_NUM_DATA_POINTERS; i++)
linesize_align[i] = 16;
}
#endif
}
@ -241,7 +240,108 @@ void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height){
*width=FFALIGN(*width, align);
}
int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame)
{
AVCodecInternal *avci = avctx->internal;
InternalBuffer *buf;
int buf_size, ret, i, needs_extended_data;
buf_size = av_samples_get_buffer_size(NULL, avctx->channels,
frame->nb_samples, avctx->sample_fmt,
32);
if (buf_size < 0)
return AVERROR(EINVAL);
needs_extended_data = av_sample_fmt_is_planar(avctx->sample_fmt) &&
avctx->channels > AV_NUM_DATA_POINTERS;
/* allocate InternalBuffer if needed */
if (!avci->buffer) {
avci->buffer = av_mallocz(sizeof(InternalBuffer));
if (!avci->buffer)
return AVERROR(ENOMEM);
}
buf = avci->buffer;
/* if there is a previously-used internal buffer, check its size and
channel count to see if we can reuse it */
if (buf->extended_data) {
/* if current buffer is too small, free it */
if (buf->extended_data[0] && buf_size > buf->audio_data_size) {
av_free(buf->extended_data[0]);
if (buf->extended_data != buf->data)
av_free(&buf->extended_data);
buf->extended_data = NULL;
buf->data[0] = NULL;
}
/* if number of channels has changed, reset and/or free extended data
pointers but leave data buffer in buf->data[0] for reuse */
if (buf->nb_channels != avctx->channels) {
if (buf->extended_data != buf->data)
av_free(buf->extended_data);
buf->extended_data = NULL;
}
}
/* if there is no previous buffer or the previous buffer cannot be used
as-is, allocate a new buffer and/or rearrange the channel pointers */
if (!buf->extended_data) {
/* if the channel pointers will fit, just set extended_data to data,
otherwise allocate the extended_data channel pointers */
if (needs_extended_data) {
buf->extended_data = av_mallocz(avctx->channels *
sizeof(*buf->extended_data));
if (!buf->extended_data)
return AVERROR(ENOMEM);
} else {
buf->extended_data = buf->data;
}
/* if there is a previous buffer and it is large enough, reuse it and
just fill-in new channel pointers and linesize, otherwise allocate
a new buffer */
if (buf->extended_data[0]) {
ret = av_samples_fill_arrays(buf->extended_data, &buf->linesize[0],
buf->extended_data[0], avctx->channels,
frame->nb_samples, avctx->sample_fmt,
32);
} else {
ret = av_samples_alloc(buf->extended_data, &buf->linesize[0],
avctx->channels, frame->nb_samples,
avctx->sample_fmt, 32);
}
if (ret)
return ret;
/* if data was not used for extended_data, we need to copy as many of
the extended_data channel pointers as will fit */
if (needs_extended_data) {
for (i = 0; i < AV_NUM_DATA_POINTERS; i++)
buf->data[i] = buf->extended_data[i];
}
buf->audio_data_size = buf_size;
buf->nb_channels = avctx->channels;
}
/* copy InternalBuffer info to the AVFrame */
frame->type = FF_BUFFER_TYPE_INTERNAL;
frame->extended_data = buf->extended_data;
frame->linesize[0] = buf->linesize[0];
memcpy(frame->data, buf->data, sizeof(frame->data));
if (avctx->pkt) frame->pkt_pts = avctx->pkt->pts;
else frame->pkt_pts = AV_NOPTS_VALUE;
frame->reordered_opaque = avctx->reordered_opaque;
if (avctx->debug & FF_DEBUG_BUFFERS)
av_log(avctx, AV_LOG_DEBUG, "default_get_buffer called on frame %p, "
"internal audio buffer used\n", frame);
return 0;
}
static int video_get_buffer(AVCodecContext *s, AVFrame *pic)
{
int i;
int w= s->width;
int h= s->height;
@ -362,6 +462,7 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
pic->data[i]= buf->data[i];
pic->linesize[i]= buf->linesize[i];
}
pic->extended_data = pic->data;
avci->buffer_count++;
if(s->pkt) pic->pkt_pts= s->pkt->pts;
@ -375,11 +476,25 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
return 0;
}
int avcodec_default_get_buffer(AVCodecContext *avctx, AVFrame *frame)
{
switch (avctx->codec_type) {
case AVMEDIA_TYPE_VIDEO:
return video_get_buffer(avctx, frame);
case AVMEDIA_TYPE_AUDIO:
return audio_get_buffer(avctx, frame);
default:
return -1;
}
}
void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic){
int i;
InternalBuffer *buf, *last;
AVCodecInternal *avci = s->internal;
assert(s->codec_type == AVMEDIA_TYPE_VIDEO);
assert(pic->type==FF_BUFFER_TYPE_INTERNAL);
assert(avci->buffer_count);
@ -412,6 +527,8 @@ int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame *pic){
AVFrame temp_pic;
int i;
assert(s->codec_type == AVMEDIA_TYPE_VIDEO);
/* If no picture return a new buffer */
if(pic->data[0] == NULL) {
/* We will copy from buffer, so must be readable */
@ -761,11 +878,59 @@ int attribute_align_arg avcodec_decode_video2(AVCodecContext *avctx, AVFrame *pi
return ret;
}
#if FF_API_OLD_DECODE_AUDIO
int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
AVPacket *avpkt)
{
int ret;
AVFrame frame;
int ret, got_frame = 0;
if (avctx->get_buffer != avcodec_default_get_buffer) {
av_log(avctx, AV_LOG_ERROR, "A custom get_buffer() cannot be used with "
"avcodec_decode_audio3()\n");
return AVERROR(EINVAL);
}
ret = avcodec_decode_audio4(avctx, &frame, &got_frame, avpkt);
if (ret >= 0 && got_frame) {
int ch, plane_size;
int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
int data_size = av_samples_get_buffer_size(&plane_size, avctx->channels,
frame.nb_samples,
avctx->sample_fmt, 1);
if (*frame_size_ptr < data_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small for "
"the current frame (%d < %d)\n", *frame_size_ptr, data_size);
return AVERROR(EINVAL);
}
memcpy(samples, frame.extended_data[0], plane_size);
if (planar && avctx->channels > 1) {
uint8_t *out = ((uint8_t *)samples) + plane_size;
for (ch = 1; ch < avctx->channels; ch++) {
memcpy(out, frame.extended_data[ch], plane_size);
out += plane_size;
}
}
*frame_size_ptr = data_size;
} else {
*frame_size_ptr = 0;
}
return ret;
}
#endif
int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx,
AVFrame *frame,
int *got_frame_ptr,
AVPacket *avpkt)
{
int ret = 0;
*got_frame_ptr = 0;
avctx->pkt = avpkt;
@ -775,22 +940,11 @@ int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *sa
}
if ((avctx->codec->capabilities & CODEC_CAP_DELAY) || avpkt->size) {
//FIXME remove the check below _after_ ensuring that all audio check that the available space is enough
if(*frame_size_ptr < AVCODEC_MAX_AUDIO_FRAME_SIZE){
av_log(avctx, AV_LOG_ERROR, "buffer smaller than AVCODEC_MAX_AUDIO_FRAME_SIZE\n");
return -1;
}
if(*frame_size_ptr < FF_MIN_BUFFER_SIZE ||
*frame_size_ptr < avctx->channels * avctx->frame_size * sizeof(int16_t)){
av_log(avctx, AV_LOG_ERROR, "buffer %d too small\n", *frame_size_ptr);
return -1;
}
ret = avctx->codec->decode(avctx, samples, frame_size_ptr, avpkt);
ret = avctx->codec->decode(avctx, frame, got_frame_ptr, avpkt);
if (ret >= 0 && *got_frame_ptr) {
avctx->frame_number++;
}else{
ret= 0;
*frame_size_ptr=0;
frame->pkt_dts = avpkt->dts;
}
}
return ret;
}
@ -1115,7 +1269,8 @@ void avcodec_flush_buffers(AVCodecContext *avctx)
avctx->codec->flush(avctx);
}
void avcodec_default_free_buffers(AVCodecContext *s){
static void video_free_buffers(AVCodecContext *s)
{
AVCodecInternal *avci = s->internal;
int i, j;
@ -1137,6 +1292,37 @@ void avcodec_default_free_buffers(AVCodecContext *s){
avci->buffer_count=0;
}
static void audio_free_buffers(AVCodecContext *avctx)
{
AVCodecInternal *avci = avctx->internal;
InternalBuffer *buf;
if (!avci->buffer)
return;
buf = avci->buffer;
if (buf->extended_data) {
av_free(buf->extended_data[0]);
if (buf->extended_data != buf->data)
av_free(buf->extended_data);
}
av_freep(&avci->buffer);
}
void avcodec_default_free_buffers(AVCodecContext *avctx)
{
switch (avctx->codec_type) {
case AVMEDIA_TYPE_VIDEO:
video_free_buffers(avctx);
break;
case AVMEDIA_TYPE_AUDIO:
audio_free_buffers(avctx);
break;
default:
break;
}
}
#if FF_API_OLD_FF_PICT_TYPES
char av_get_pict_type_char(int pict_type){
return av_get_picture_type_char(pict_type);

5
libavcodec/version.h
View File

@ -21,7 +21,7 @@
#define AVCODEC_VERSION_H
#define LIBAVCODEC_VERSION_MAJOR 53
#define LIBAVCODEC_VERSION_MINOR 24
#define LIBAVCODEC_VERSION_MINOR 25
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
@ -113,5 +113,8 @@
#ifndef FF_API_DATA_POINTERS
#define FF_API_DATA_POINTERS (LIBAVCODEC_VERSION_MAJOR < 54)
#endif
#ifndef FF_API_OLD_DECODE_AUDIO
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 54)
#endif
#endif /* AVCODEC_VERSION_H */

35
libavcodec/vmdav.c
View File

@ -473,6 +473,7 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
#define BLOCK_TYPE_SILENCE 3
typedef struct VmdAudioContext {
AVFrame frame;
int out_bps;
int chunk_size;
} VmdAudioContext;
@ -514,6 +515,9 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
"block align = %d, sample rate = %d\n",
avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
@ -551,22 +555,21 @@ static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
}
}
static int vmdaudio_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
const uint8_t *buf_end;
int buf_size = avpkt->size;
VmdAudioContext *s = avctx->priv_data;
int block_type, silent_chunks, audio_chunks;
int nb_samples, out_size;
uint8_t *output_samples_u8 = data;
int16_t *output_samples_s16 = data;
int ret;
uint8_t *output_samples_u8;
int16_t *output_samples_s16;
if (buf_size < 16) {
av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
@ -597,10 +600,15 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
/* ensure output buffer is large enough */
audio_chunks = buf_size / s->chunk_size;
nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels;
out_size = nb_samples * avctx->channels * s->out_bps;
if (*data_size < out_size)
return -1;
/* get output buffer */
s->frame.nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
output_samples_u8 = s->frame.data[0];
output_samples_s16 = (int16_t *)s->frame.data[0];
/* decode silent chunks */
if (silent_chunks > 0) {
@ -630,7 +638,9 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
}
}
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return avpkt->size;
}
@ -658,5 +668,6 @@ AVCodec ff_vmdaudio_decoder = {
.priv_data_size = sizeof(VmdAudioContext),
.init = vmdaudio_decode_init,
.decode = vmdaudio_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
};

33
libavcodec/vorbisdec.c
View File

@ -121,6 +121,7 @@ typedef struct {
typedef struct vorbis_context_s {
AVCodecContext *avccontext;
AVFrame frame;
GetBitContext gb;
DSPContext dsp;
FmtConvertContext fmt_conv;
@ -1033,6 +1034,9 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avccontext->sample_rate = vc->audio_samplerate;
avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
avcodec_get_frame_defaults(&vc->frame);
avccontext->coded_frame = &vc->frame;
return 0;
}
@ -1605,16 +1609,15 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
// Return the decoded audio packet through the standard api
static int vorbis_decode_frame(AVCodecContext *avccontext,
void *data, int *data_size,
AVPacket *avpkt)
static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
vorbis_context *vc = avccontext->priv_data;
GetBitContext *gb = &(vc->gb);
const float *channel_ptrs[255];
int i, len, out_size;
int i, len, ret;
av_dlog(NULL, "packet length %d \n", buf_size);
@ -1625,18 +1628,18 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
if (!vc->first_frame) {
vc->first_frame = 1;
*data_size = 0;
*got_frame_ptr = 0;
return buf_size;
}
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);
out_size = len * vc->audio_channels *
av_get_bytes_per_sample(avccontext->sample_fmt);
if (*data_size < out_size) {
av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* 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) {
@ -1649,12 +1652,15 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
}
if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
len, vc->audio_channels);
else
vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
channel_ptrs, len,
vc->audio_channels);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = vc->frame;
return buf_size;
}
@ -1678,6 +1684,7 @@ AVCodec ff_vorbis_decoder = {
.init = vorbis_decode_init,
.close = vorbis_decode_close,
.decode = vorbis_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
.sample_fmts = (const enum AVSampleFormat[]) {

112
libavcodec/wavpack.c
View File

@ -115,8 +115,6 @@ typedef struct WavpackFrameContext {
int float_shift;
int float_max_exp;
WvChannel ch[2];
int samples_left;
int max_samples;
int pos;
SavedContext sc, extra_sc;
} WavpackFrameContext;
@ -125,6 +123,7 @@ typedef struct WavpackFrameContext {
typedef struct WavpackContext {
AVCodecContext *avctx;
AVFrame frame;
WavpackFrameContext *fdec[WV_MAX_FRAME_DECODERS];
int fdec_num;
@ -133,7 +132,6 @@ typedef struct WavpackContext {
int mkv_mode;
int block;
int samples;
int samples_left;
int ch_offset;
} WavpackContext;
@ -485,7 +483,6 @@ static float wv_get_value_float(WavpackFrameContext *s, uint32_t *crc, int S)
static void wv_reset_saved_context(WavpackFrameContext *s)
{
s->pos = 0;
s->samples_left = 0;
s->sc.crc = s->extra_sc.crc = 0xFFFFFFFF;
}
@ -502,7 +499,6 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo
float *dstfl = dst;
const int channel_pad = s->avctx->channels - 2;
if(s->samples_left == s->samples)
s->one = s->zero = s->zeroes = 0;
do{
L = wv_get_value(s, gb, 0, &last);
@ -594,13 +590,8 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo
dst16 += channel_pad;
}
count++;
}while(!last && count < s->max_samples);
} while (!last && count < s->samples);
if (last)
s->samples_left = 0;
else
s->samples_left -= count;
if(!s->samples_left){
wv_reset_saved_context(s);
if(crc != s->CRC){
av_log(s->avctx, AV_LOG_ERROR, "CRC error\n");
@ -610,15 +601,7 @@ static inline int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, vo
av_log(s->avctx, AV_LOG_ERROR, "Extra bits CRC error\n");
return -1;
}
}else{
s->pos = pos;
s->sc.crc = crc;
s->sc.bits_used = get_bits_count(&s->gb);
if(s->got_extra_bits){
s->extra_sc.crc = crc_extra_bits;
s->extra_sc.bits_used = get_bits_count(&s->gb_extra_bits);
}
}
return count * 2;
}
@ -635,7 +618,6 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void
float *dstfl = dst;
const int channel_stride = s->avctx->channels;
if(s->samples_left == s->samples)
s->one = s->zero = s->zeroes = 0;
do{
T = wv_get_value(s, gb, 0, &last);
@ -675,13 +657,8 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void
dst16 += channel_stride;
}
count++;
}while(!last && count < s->max_samples);
} while (!last && count < s->samples);
if (last)
s->samples_left = 0;
else
s->samples_left -= count;
if(!s->samples_left){
wv_reset_saved_context(s);
if(crc != s->CRC){
av_log(s->avctx, AV_LOG_ERROR, "CRC error\n");
@ -691,15 +668,7 @@ static inline int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void
av_log(s->avctx, AV_LOG_ERROR, "Extra bits CRC error\n");
return -1;
}
}else{
s->pos = pos;
s->sc.crc = crc;
s->sc.bits_used = get_bits_count(&s->gb);
if(s->got_extra_bits){
s->extra_sc.crc = crc_extra_bits;
s->extra_sc.bits_used = get_bits_count(&s->gb_extra_bits);
}
}
return count;
}
@ -743,6 +712,9 @@ static av_cold int wavpack_decode_init(AVCodecContext *avctx)
s->fdec_num = 0;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -759,7 +731,7 @@ static av_cold int wavpack_decode_end(AVCodecContext *avctx)
}
static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
void *data, int *data_size,
void *data, int *got_frame_ptr,
const uint8_t *buf, int buf_size)
{
WavpackContext *wc = avctx->priv_data;
@ -774,7 +746,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
int bpp, chan, chmask;
if (buf_size == 0){
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
@ -789,18 +761,16 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
return -1;
}
if(!s->samples_left){
memset(s->decorr, 0, MAX_TERMS * sizeof(Decorr));
memset(s->ch, 0, sizeof(s->ch));
s->extra_bits = 0;
s->and = s->or = s->shift = 0;
s->got_extra_bits = 0;
}
if(!wc->mkv_mode){
s->samples = AV_RL32(buf); buf += 4;
if(!s->samples){
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
}else{
@ -829,13 +799,6 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
wc->ch_offset += 1 + s->stereo;
s->max_samples = *data_size / (bpp * avctx->channels);
s->max_samples = FFMIN(s->max_samples, s->samples);
if(s->samples_left > 0){
s->max_samples = FFMIN(s->max_samples, s->samples_left);
buf = buf_end;
}
// parse metadata blocks
while(buf < buf_end){
id = *buf++;
@ -1064,7 +1027,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
}
if(id & WP_IDF_ODD) buf++;
}
if(!s->samples_left){
if(!got_terms){
av_log(avctx, AV_LOG_ERROR, "No block with decorrelation terms\n");
return -1;
@ -1101,16 +1064,6 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
s->got_extra_bits = 0;
}
}
s->samples_left = s->samples;
}else{
init_get_bits(&s->gb, orig_buf + s->sc.offset, s->sc.size);
skip_bits_long(&s->gb, s->sc.bits_used);
if(s->got_extra_bits){
init_get_bits(&s->gb_extra_bits, orig_buf + s->extra_sc.offset,
s->extra_sc.size);
skip_bits_long(&s->gb_extra_bits, s->extra_sc.bits_used);
}
}
if(s->stereo_in){
if(avctx->sample_fmt == AV_SAMPLE_FMT_S16)
@ -1167,7 +1120,7 @@ static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
}
}
wc->samples_left = s->samples_left;
*got_frame_ptr = 1;
return samplecount * bpp;
}
@ -1181,23 +1134,40 @@ static void wavpack_decode_flush(AVCodecContext *avctx)
wv_reset_saved_context(s->fdec[i]);
}
static int wavpack_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int wavpack_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
WavpackContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int frame_size;
int frame_size, ret;
int samplecount = 0;
s->block = 0;
s->samples_left = 0;
s->ch_offset = 0;
/* determine number of samples */
if(s->mkv_mode){
s->samples = AV_RL32(buf); buf += 4;
} else {
if (s->multichannel)
s->samples = AV_RL32(buf + 4);
else
s->samples = AV_RL32(buf);
}
if (s->samples <= 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of samples: %d\n",
s->samples);
return AVERROR(EINVAL);
}
/* get output buffer */
s->frame.nb_samples = s->samples;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
while(buf_size > 0){
if(!s->multichannel){
frame_size = buf_size;
@ -1216,17 +1186,19 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
wavpack_decode_flush(avctx);
return -1;
}
if((samplecount = wavpack_decode_block(avctx, s->block, data,
data_size, buf, frame_size)) < 0) {
if((samplecount = wavpack_decode_block(avctx, s->block, s->frame.data[0],
got_frame_ptr, buf, frame_size)) < 0) {
wavpack_decode_flush(avctx);
return -1;
}
s->block++;
buf += frame_size; buf_size -= frame_size;
}
*data_size = samplecount * avctx->channels;
return s->samples_left > 0 ? 0 : avpkt->size;
if (*got_frame_ptr)
*(AVFrame *)data = s->frame;
return avpkt->size;
}
AVCodec ff_wavpack_decoder = {
@ -1238,6 +1210,6 @@ AVCodec ff_wavpack_decoder = {
.close = wavpack_decode_end,
.decode = wavpack_decode_frame,
.flush = wavpack_decode_flush,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("WavPack"),
};

1
libavcodec/wma.h
View File

@ -65,6 +65,7 @@ typedef struct CoefVLCTable {
typedef struct WMACodecContext {
AVCodecContext* avctx;
AVFrame frame;
GetBitContext gb;
PutBitContext pb;
int sample_rate;

30
libavcodec/wmadec.c
View File

@ -124,6 +124,10 @@ static int wma_decode_init(AVCodecContext * avctx)
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -797,14 +801,13 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
return 0;
}
static int wma_decode_superframe(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int wma_decode_superframe(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
WMACodecContext *s = avctx->priv_data;
int nb_frames, bit_offset, i, pos, len, out_size;
int nb_frames, bit_offset, i, pos, len, ret;
uint8_t *q;
int16_t *samples;
@ -818,8 +821,6 @@ static int wma_decode_superframe(AVCodecContext *avctx,
return 0;
buf_size = s->block_align;
samples = data;
init_get_bits(&s->gb, buf, buf_size*8);
if (s->use_bit_reservoir) {
@ -830,12 +831,13 @@ static int wma_decode_superframe(AVCodecContext *avctx,
nb_frames = 1;
}
out_size = nb_frames * s->frame_len * s->nb_channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(s->avctx, AV_LOG_ERROR, "Insufficient output space\n");
goto fail;
/* get output buffer */
s->frame.nb_samples = nb_frames * s->frame_len;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)s->frame.data[0];
if (s->use_bit_reservoir) {
bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
@ -903,7 +905,9 @@ static int wma_decode_superframe(AVCodecContext *avctx,
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d outbytes:%d eaten:%d\n", s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len, (int8_t *)samples - (int8_t *)data, s->block_align);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return s->block_align;
fail:
/* when error, we reset the bit reservoir */
@ -928,6 +932,7 @@ AVCodec ff_wmav1_decoder = {
.close = ff_wma_end,
.decode = wma_decode_superframe,
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
};
@ -940,5 +945,6 @@ AVCodec ff_wmav2_decoder = {
.close = ff_wma_end,
.decode = wma_decode_superframe,
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};

63
libavcodec/wmaprodec.c
View File

@ -167,6 +167,7 @@ typedef struct {
typedef struct WMAProDecodeCtx {
/* generic decoder variables */
AVCodecContext* avctx; ///< codec context for av_log
AVFrame frame; ///< AVFrame for decoded output
DSPContext dsp; ///< accelerated DSP functions
FmtConvertContext fmt_conv;
uint8_t frame_data[MAX_FRAMESIZE +
@ -209,8 +210,6 @@ typedef struct WMAProDecodeCtx {
uint32_t frame_num; ///< current frame number (not used for decoding)
GetBitContext gb; ///< bitstream reader context
int buf_bit_size; ///< buffer size in bits
float* samples; ///< current samplebuffer pointer
float* samples_end; ///< maximum samplebuffer pointer
uint8_t drc_gain; ///< gain for the DRC tool
int8_t skip_frame; ///< skip output step
int8_t parsed_all_subframes; ///< all subframes decoded?
@ -453,6 +452,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
dump_context(s);
avctx->channel_layout = channel_mask;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@ -1279,22 +1282,15 @@ static int decode_subframe(WMAProDecodeCtx *s)
*@return 0 if the trailer bit indicates that this is the last frame,
* 1 if there are additional frames
*/
static int decode_frame(WMAProDecodeCtx *s)
static int decode_frame(WMAProDecodeCtx *s, int *got_frame_ptr)
{
AVCodecContext *avctx = s->avctx;
GetBitContext* gb = &s->gb;
int more_frames = 0;
int len = 0;
int i;
int i, ret;
const float *out_ptr[WMAPRO_MAX_CHANNELS];
/** check for potential output buffer overflow */
if (s->num_channels * s->samples_per_frame > s->samples_end - s->samples) {
/** return an error if no frame could be decoded at all */
av_log(s->avctx, AV_LOG_ERROR,
"not enough space for the output samples\n");
s->packet_loss = 1;
return 0;
}
float *samples;
/** get frame length */
if (s->len_prefix)
@ -1360,10 +1356,19 @@ static int decode_frame(WMAProDecodeCtx *s)
}
}
/* get output buffer */
s->frame.nb_samples = s->samples_per_frame;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
s->packet_loss = 1;
return 0;
}
samples = (float *)s->frame.data[0];
/** interleave samples and write them to the output buffer */
for (i = 0; i < s->num_channels; i++)
out_ptr[i] = s->channel[i].out;
s->fmt_conv.float_interleave(s->samples, out_ptr, s->samples_per_frame,
s->fmt_conv.float_interleave(samples, out_ptr, s->samples_per_frame,
s->num_channels);
for (i = 0; i < s->num_channels; i++) {
@ -1375,8 +1380,10 @@ static int decode_frame(WMAProDecodeCtx *s)
if (s->skip_frame) {
s->skip_frame = 0;
} else
s->samples += s->num_channels * s->samples_per_frame;
*got_frame_ptr = 0;
} else {
*got_frame_ptr = 1;
}
if (s->len_prefix) {
if (len != (get_bits_count(gb) - s->frame_offset) + 2) {
@ -1473,8 +1480,8 @@ static void save_bits(WMAProDecodeCtx *s, GetBitContext* gb, int len,
*@param avpkt input packet
*@return number of bytes that were read from the input buffer
*/
static int decode_packet(AVCodecContext *avctx,
void *data, int *data_size, AVPacket* avpkt)
static int decode_packet(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket* avpkt)
{
WMAProDecodeCtx *s = avctx->priv_data;
GetBitContext* gb = &s->pgb;
@ -1483,9 +1490,7 @@ static int decode_packet(AVCodecContext *avctx,
int num_bits_prev_frame;
int packet_sequence_number;
s->samples = data;
s->samples_end = (float*)((int8_t*)data + *data_size);
*data_size = 0;
*got_frame_ptr = 0;
if (s->packet_done || s->packet_loss) {
s->packet_done = 0;
@ -1532,7 +1537,7 @@ static int decode_packet(AVCodecContext *avctx,
/** decode the cross packet frame if it is valid */
if (!s->packet_loss)
decode_frame(s);
decode_frame(s, got_frame_ptr);
} else if (s->num_saved_bits - s->frame_offset) {
av_dlog(avctx, "ignoring %x previously saved bits\n",
s->num_saved_bits - s->frame_offset);
@ -1555,7 +1560,7 @@ static int decode_packet(AVCodecContext *avctx,
(frame_size = show_bits(gb, s->log2_frame_size)) &&
frame_size <= remaining_bits(s, gb)) {
save_bits(s, gb, frame_size, 0);
s->packet_done = !decode_frame(s);
s->packet_done = !decode_frame(s, got_frame_ptr);
} else if (!s->len_prefix
&& s->num_saved_bits > get_bits_count(&s->gb)) {
/** when the frames do not have a length prefix, we don't know
@ -1565,7 +1570,7 @@ static int decode_packet(AVCodecContext *avctx,
therefore we save the incoming packet first, then we append
the "previous frame" data from the next packet so that
we get a buffer that only contains full frames */
s->packet_done = !decode_frame(s);
s->packet_done = !decode_frame(s, got_frame_ptr);
} else
s->packet_done = 1;
}
@ -1577,10 +1582,14 @@ static int decode_packet(AVCodecContext *avctx,
save_bits(s, gb, remaining_bits(s, gb), 0);
}
*data_size = (int8_t *)s->samples - (int8_t *)data;
s->packet_offset = get_bits_count(gb) & 7;
if (s->packet_loss)
return AVERROR_INVALIDDATA;
return (s->packet_loss) ? AVERROR_INVALIDDATA : get_bits_count(gb) >> 3;
if (*got_frame_ptr)
*(AVFrame *)data = s->frame;
return get_bits_count(gb) >> 3;
}
/**
@ -1611,7 +1620,7 @@ AVCodec ff_wmapro_decoder = {
.init = decode_init,
.close = decode_end,
.decode = decode_packet,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.flush= flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"),
};

46
libavcodec/wmavoice.c
View File

@ -131,6 +131,7 @@ typedef struct {
* @name Global values specified in the stream header / extradata or used all over.
* @{
*/
AVFrame frame;
GetBitContext gb; ///< packet bitreader. During decoder init,
///< it contains the extradata from the
///< demuxer. During decoding, it contains
@ -438,6 +439,9 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avcodec_get_frame_defaults(&s->frame);
ctx->coded_frame = &s->frame;
return 0;
}
@ -1725,17 +1729,17 @@ static int check_bits_for_superframe(GetBitContext *orig_gb,
* @return 0 on success, <0 on error or 1 if there was not enough data to
* fully parse the superframe
*/
static int synth_superframe(AVCodecContext *ctx,
float *samples, int *data_size)
static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr)
{
WMAVoiceContext *s = ctx->priv_data;
GetBitContext *gb = &s->gb, s_gb;
int n, res, out_size, n_samples = 480;
int n, res, n_samples = 480;
double lsps[MAX_FRAMES][MAX_LSPS];
const double *mean_lsf = s->lsps == 16 ?
wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
float synth[MAX_LSPS + MAX_SFRAMESIZE];
float *samples;
memcpy(synth, s->synth_history,
s->lsps * sizeof(*synth));
@ -1749,7 +1753,7 @@ static int synth_superframe(AVCodecContext *ctx,
}
if ((res = check_bits_for_superframe(gb, s)) == 1) {
*data_size = 0;
*got_frame_ptr = 0;
return 1;
}
@ -1792,13 +1796,14 @@ static int synth_superframe(AVCodecContext *ctx,
stabilize_lsps(lsps[n], s->lsps);
}
out_size = n_samples * av_get_bytes_per_sample(ctx->sample_fmt);
if (*data_size < out_size) {
av_log(ctx, AV_LOG_ERROR,
"Output buffer too small (%d given - %d needed)\n",
*data_size, out_size);
return -1;
/* get output buffer */
s->frame.nb_samples = 480;
if ((res = ctx->get_buffer(ctx, &s->frame)) < 0) {
av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
return res;
}
s->frame.nb_samples = n_samples;
samples = (float *)s->frame.data[0];
/* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
for (n = 0; n < 3; n++) {
@ -1820,7 +1825,7 @@ static int synth_superframe(AVCodecContext *ctx,
lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
&excitation[s->history_nsamples + n * MAX_FRAMESIZE],
&synth[s->lsps + n * MAX_FRAMESIZE]))) {
*data_size = 0;
*got_frame_ptr = 0;
return res;
}
}
@ -1833,8 +1838,7 @@ static int synth_superframe(AVCodecContext *ctx,
skip_bits(gb, 10 * (res + 1));
}
/* Specify nr. of output samples */
*data_size = out_size;
*got_frame_ptr = 1;
/* Update history */
memcpy(s->prev_lsps, lsps[2],
@ -1922,7 +1926,7 @@ static void copy_bits(PutBitContext *pb,
* For more information about frames, see #synth_superframe().
*/
static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
WMAVoiceContext *s = ctx->priv_data;
GetBitContext *gb = &s->gb;
@ -1935,7 +1939,7 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
* capping the packet size at ctx->block_align. */
for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
if (!size) {
*data_size = 0;
*got_frame_ptr = 0;
return 0;
}
init_get_bits(&s->gb, avpkt->data, size << 3);
@ -1956,10 +1960,11 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
flush_put_bits(&s->pb);
s->sframe_cache_size += s->spillover_nbits;
if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
*data_size > 0) {
if ((res = synth_superframe(ctx, got_frame_ptr)) == 0 &&
*got_frame_ptr) {
cnt += s->spillover_nbits;
s->skip_bits_next = cnt & 7;
*(AVFrame *)data = s->frame;
return cnt >> 3;
} else
skip_bits_long (gb, s->spillover_nbits - cnt +
@ -1974,11 +1979,12 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
s->sframe_cache_size = 0;
s->skip_bits_next = 0;
pos = get_bits_left(gb);
if ((res = synth_superframe(ctx, data, data_size)) < 0) {
if ((res = synth_superframe(ctx, got_frame_ptr)) < 0) {
return res;
} else if (*data_size > 0) {
} else if (*got_frame_ptr) {
int cnt = get_bits_count(gb);
s->skip_bits_next = cnt & 7;
*(AVFrame *)data = s->frame;
return cnt >> 3;
} else if ((s->sframe_cache_size = pos) > 0) {
/* rewind bit reader to start of last (incomplete) superframe... */
@ -2046,7 +2052,7 @@ AVCodec ff_wmavoice_decoder = {
.init = wmavoice_decode_init,
.close = wmavoice_decode_end,
.decode = wmavoice_decode_packet,
.capabilities = CODEC_CAP_SUBFRAMES,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.flush = wmavoice_flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
};

38
libavcodec/ws-snd1.c
View File

@ -37,26 +37,37 @@ static const int8_t ws_adpcm_4bit[] = {
0, 1, 2, 3, 4, 5, 6, 8
};
typedef struct WSSndContext {
AVFrame frame;
} WSSndContext;
static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
{
WSSndContext *s = avctx->priv_data;
if (avctx->channels != 1) {
av_log_ask_for_sample(avctx, "unsupported number of channels\n");
return AVERROR(EINVAL);
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
WSSndContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int in_size, out_size;
int in_size, out_size, ret;
int sample = 128;
uint8_t *samples = data;
uint8_t *samples;
uint8_t *samples_end;
if (!buf_size)
@ -71,19 +82,24 @@ static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
in_size = AV_RL16(&buf[2]);
buf += 4;
if (out_size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n");
return -1;
}
if (in_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n");
return -1;
}
/* get output buffer */
s->frame.nb_samples = out_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = s->frame.data[0];
samples_end = samples + out_size;
if (in_size == out_size) {
memcpy(samples, buf, out_size);
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -159,7 +175,9 @@ static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
}
}
*data_size = samples - (uint8_t *)data;
s->frame.nb_samples = samples - s->frame.data[0];
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
@ -168,7 +186,9 @@ AVCodec ff_ws_snd1_decoder = {
.name = "ws_snd1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WESTWOOD_SND1,
.priv_data_size = sizeof(WSSndContext),
.init = ws_snd_decode_init,
.decode = ws_snd_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"),
};