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Replace deprecated symbols SAMPLE_FMT_* with AV_SAMPLE_FMT_*, and enum

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SampleFormat with AVSampleFormat.

Originally committed as revision 25730 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Stefano Sabatini
2010-11-12 11:04:40 +00:00
parent 09f47fa44e
commit 5d6e4c160a
80 changed files with 234 additions and 234 deletions
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12
ffmpeg.c
View File

@@ -148,7 +148,7 @@ static int frame_width = 0;
static int frame_height = 0;
static float frame_aspect_ratio = 0;
static enum PixelFormat frame_pix_fmt = PIX_FMT_NONE;
static enum SampleFormat audio_sample_fmt = SAMPLE_FMT_NONE;
static enum AVSampleFormat audio_sample_fmt = AV_SAMPLE_FMT_NONE;
static int max_frames[4] = {INT_MAX, INT_MAX, INT_MAX, INT_MAX};
static AVRational frame_rate;
static float video_qscale = 0;
@@ -597,7 +597,7 @@ static void *grow_array(void *array, int elem_size, int *size, int new_size)
static void choose_sample_fmt(AVStream *st, AVCodec *codec)
{
if(codec && codec->sample_fmts){
const enum SampleFormat *p= codec->sample_fmts;
const enum AVSampleFormat *p= codec->sample_fmts;
for(; *p!=-1; p++){
if(*p == st->codec->sample_fmt)
break;
@@ -809,7 +809,7 @@ need_realloc:
ost->audio_resample = 1;
if (ost->audio_resample && !ost->resample) {
if (dec->sample_fmt != SAMPLE_FMT_S16)
if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate,
@@ -823,7 +823,7 @@ need_realloc:
}
}
#define MAKE_SFMT_PAIR(a,b) ((a)+SAMPLE_FMT_NB*(b))
#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (ost->reformat_ctx)
@@ -2175,7 +2175,7 @@ static int transcode(AVFormatContext **output_files,
ost->fifo= av_fifo_alloc(1024);
if(!ost->fifo)
goto fail;
ost->reformat_pair = MAKE_SFMT_PAIR(SAMPLE_FMT_NONE,SAMPLE_FMT_NONE);
ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
icodec->request_channels = codec->channels;
ist->decoding_needed = 1;
@@ -2851,7 +2851,7 @@ static void opt_audio_sample_fmt(const char *arg)
if (strcmp(arg, "list"))
audio_sample_fmt = av_get_sample_fmt(arg);
else {
list_fmts(av_get_sample_fmt_string, SAMPLE_FMT_NB);
list_fmts(av_get_sample_fmt_string, AV_SAMPLE_FMT_NB);
ffmpeg_exit(0);
}
}

8
ffplay.c
View File

@@ -163,7 +163,7 @@ typedef struct VideoState {
int audio_buf_index; /* in bytes */
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
enum SampleFormat audio_src_fmt;
enum AVSampleFormat audio_src_fmt;
AVAudioConvert *reformat_ctx;
int show_audio; /* if true, display audio samples */
@@ -2095,12 +2095,12 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
if (dec->sample_fmt != is->audio_src_fmt) {
if (is->reformat_ctx)
av_audio_convert_free(is->reformat_ctx);
is->reformat_ctx= av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
is->reformat_ctx= av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
dec->sample_fmt, 1, NULL, 0);
if (!is->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
av_get_sample_fmt_name(dec->sample_fmt),
av_get_sample_fmt_name(SAMPLE_FMT_S16));
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16));
break;
}
is->audio_src_fmt= dec->sample_fmt;
@@ -2268,7 +2268,7 @@ static int stream_component_open(VideoState *is, int stream_index)
return -1;
}
is->audio_hw_buf_size = spec.size;
is->audio_src_fmt= SAMPLE_FMT_S16;
is->audio_src_fmt= AV_SAMPLE_FMT_S16;
}
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;

2
libavcodec/8svx.c
View File

@@ -88,7 +88,7 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
default:
return -1;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

10
libavcodec/aacdec.c
View File

@@ -545,7 +545,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
return -1;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
@@ -2369,8 +2369,8 @@ AVCodec aac_decoder = {
aac_decode_close,
aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum SampleFormat[]) {
SAMPLE_FMT_S16,SAMPLE_FMT_NONE
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};
@@ -2389,8 +2389,8 @@ AVCodec aac_latm_decoder = {
.close = aac_decode_close,
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum SampleFormat[]) {
SAMPLE_FMT_S16,SAMPLE_FMT_NONE
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
},
.channel_layouts = aac_channel_layout,
};

2
libavcodec/aacenc.c
View File

@@ -645,6 +645,6 @@ AVCodec aac_encoder = {
aac_encode_frame,
aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};

2
libavcodec/ac3dec.c
View File

@@ -219,7 +219,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
return AVERROR(ENOMEM);
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/ac3enc.c
View File

@@ -1400,7 +1400,7 @@ AVCodec ac3_encoder = {
AC3_encode_frame,
AC3_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.channel_layouts = (const int64_t[]){
CH_LAYOUT_MONO,

4
libavcodec/adpcm.c
View File

@@ -737,7 +737,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
default:
break;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
@@ -1678,7 +1678,7 @@ AVCodec name ## _encoder = { \
adpcm_encode_frame, \
adpcm_encode_close, \
NULL, \
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, \
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
};
#else

2
libavcodec/adxdec.c
View File

@@ -34,7 +34,7 @@
static av_cold int adx_decode_init(AVCodecContext *avctx)
{
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/adxenc.c
View File

@@ -192,6 +192,6 @@ AVCodec adpcm_adx_encoder = {
adx_encode_frame,
adx_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};

4
libavcodec/alac.c
View File

@@ -505,10 +505,10 @@ static int alac_decode_frame(AVCodecContext *avctx,
outputsamples = alac->setinfo_max_samples_per_frame;
switch (alac->setinfo_sample_size) {
case 16: avctx->sample_fmt = SAMPLE_FMT_S16;
case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
alac->bytespersample = channels << 1;
break;
case 24: avctx->sample_fmt = SAMPLE_FMT_S32;
case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
alac->bytespersample = channels << 2;
break;
default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",

4
libavcodec/alacenc.c
View File

@@ -383,7 +383,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
avctx->frame_size = DEFAULT_FRAME_SIZE;
avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
if(avctx->sample_fmt != SAMPLE_FMT_S16) {
if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
@@ -528,6 +528,6 @@ AVCodec alac_encoder = {
alac_encode_frame,
alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};

4
libavcodec/alsdec.c
View File

@@ -1573,11 +1573,11 @@ static av_cold int decode_init(AVCodecContext *avctx)
ff_bgmc_init(avctx, &ctx->bgmc_lut, &ctx->bgmc_lut_status);
if (sconf->floating) {
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->bits_per_raw_sample = 32;
} else {
avctx->sample_fmt = sconf->resolution > 1
? SAMPLE_FMT_S32 : SAMPLE_FMT_S16;
? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16;
avctx->bits_per_raw_sample = (sconf->resolution + 1) * 8;
}

4
libavcodec/amrnbdec.c
View File

@@ -154,7 +154,7 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
AMRContext *p = avctx->priv_data;
int i;
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
// p->excitation always points to the same position in p->excitation_buf
p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
@@ -1044,5 +1044,5 @@ AVCodec amrnb_decoder = {
.init = amrnb_decode_init,
.decode = amrnb_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
.sample_fmts = (enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
};

2
libavcodec/apedec.c
View File

@@ -198,7 +198,7 @@ static av_cold int ape_decode_init(AVCodecContext * avctx)
}
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
return 0;
}

2
libavcodec/atrac1.c
View File

@@ -326,7 +326,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
AT1Ctx *q = avctx->priv_data;
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
q->channels = avctx->channels;

2
libavcodec/atrac3.c
View File

@@ -1014,7 +1014,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
return AVERROR(ENOMEM);
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

60
libavcodec/audioconvert.c
View File

@@ -37,7 +37,7 @@ const char *avcodec_get_sample_fmt_name(int sample_fmt)
return av_get_sample_fmt_name(sample_fmt);
}
enum SampleFormat avcodec_get_sample_fmt(const char* name)
enum AVSampleFormat avcodec_get_sample_fmt(const char* name)
{
return av_get_sample_fmt(name);
}
@@ -152,8 +152,8 @@ struct AVAudioConvert {
int fmt_pair;
};
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
enum AVSampleFormat in_fmt, int in_channels,
const float *matrix, int flags)
{
AVAudioConvert *ctx;
@@ -164,7 +164,7 @@ AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channe
return NULL;
ctx->in_channels = in_channels;
ctx->out_channels = out_channels;
ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
return ctx;
}
@@ -191,7 +191,7 @@ int av_audio_convert(AVAudioConvert *ctx,
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
@@ -200,31 +200,31 @@ if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?
CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi)
CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
return 0;

6
libavcodec/audioconvert.h
View File

@@ -49,7 +49,7 @@ const char *avcodec_get_sample_fmt_name(int sample_fmt);
* @deprecated Use av_get_sample_fmt() instead.
*/
attribute_deprecated
enum SampleFormat avcodec_get_sample_fmt(const char* name);
enum AVSampleFormat avcodec_get_sample_fmt(const char* name);
#endif
/**
@@ -94,8 +94,8 @@ typedef struct AVAudioConvert AVAudioConvert;
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
AVAudioConvert *av_audio_convert_alloc(enum AVSampleFormat out_fmt, int out_channels,
enum AVSampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**

12
libavcodec/avcodec.h
View File

@@ -1231,7 +1231,7 @@ typedef struct AVCodecContext {
* - encoding: Set by user.
* - decoding: Set by libavcodec.
*/
enum SampleFormat sample_fmt; ///< sample format
enum AVSampleFormat sample_fmt; ///< sample format
/* The following data should not be initialized. */
/**
@@ -2555,7 +2555,7 @@ typedef struct AVCodecContext {
/**
* Bits per sample/pixel of internal libavcodec pixel/sample format.
* This field is applicable only when sample_fmt is SAMPLE_FMT_S32.
* This field is applicable only when sample_fmt is AV_SAMPLE_FMT_S32.
* - encoding: set by user.
* - decoding: set by libavcodec.
*/
@@ -2796,7 +2796,7 @@ typedef struct AVCodec {
*/
const char *long_name;
const int *supported_samplerates; ///< array of supported audio samplerates, or NULL if unknown, array is terminated by 0
const enum SampleFormat *sample_fmts; ///< array of supported sample formats, or NULL if unknown, array is terminated by -1
const enum AVSampleFormat *sample_fmts; ///< array of supported sample formats, or NULL if unknown, array is terminated by -1
const int64_t *channel_layouts; ///< array of support channel layouts, or NULL if unknown. array is terminated by 0
uint8_t max_lowres; ///< maximum value for lowres supported by the decoder
AVClass *priv_class; ///< AVClass for the private context
@@ -3060,8 +3060,8 @@ attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, i
*/
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum SampleFormat sample_fmt_out,
enum SampleFormat sample_fmt_in,
enum AVSampleFormat sample_fmt_out,
enum AVSampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff);
@@ -3744,7 +3744,7 @@ int av_get_bits_per_sample(enum CodecID codec_id);
* @deprecated Use av_get_bits_per_sample_fmt() instead.
*/
attribute_deprecated
int av_get_bits_per_sample_format(enum SampleFormat sample_fmt);
int av_get_bits_per_sample_format(enum AVSampleFormat sample_fmt);
#endif
/* frame parsing */

2
libavcodec/binkaudio.c
View File

@@ -119,7 +119,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
s->bands[s->num_bands] = s->frame_len / 2;
s->first = 1;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
for (i = 0; i < s->channels; i++)
s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;

2
libavcodec/cook.c
View File

@@ -1270,7 +1270,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
return -1;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (channel_mask)
avctx->channel_layout = channel_mask;
else

2
libavcodec/dca.c
View File

@@ -1464,7 +1464,7 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
s->add_bias = 385.0f;

2
libavcodec/dpcm.c
View File

@@ -155,7 +155,7 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx)
break;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/dsicinav.c
View File

@@ -307,7 +307,7 @@ static av_cold int cinaudio_decode_init(AVCodecContext *avctx)
cin->avctx = avctx;
cin->initial_decode_frame = 1;
cin->delta = 0;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

10
libavcodec/flacdec.c
View File

@@ -113,7 +113,7 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* for now, the raw FLAC header is allowed to be passed to the decoder as
frame data instead of extradata. */
@@ -126,9 +126,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
/* initialize based on the demuxer-supplied streamdata header */
ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
if (s->bps > 16)
avctx->sample_fmt = SAMPLE_FMT_S32;
avctx->sample_fmt = AV_SAMPLE_FMT_S32;
else
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
allocate_buffers(s);
s->got_streaminfo = 1;
@@ -603,11 +603,11 @@ static int decode_frame(FLACContext *s)
s->bps = s->avctx->bits_per_raw_sample = fi.bps;
if (s->bps > 16) {
s->avctx->sample_fmt = SAMPLE_FMT_S32;
s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
s->sample_shift = 32 - s->bps;
s->is32 = 1;
} else {
s->avctx->sample_fmt = SAMPLE_FMT_S16;
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->sample_shift = 16 - s->bps;
s->is32 = 0;
}

4
libavcodec/flacenc.c
View File

@@ -219,7 +219,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx);
if (avctx->sample_fmt != SAMPLE_FMT_S16)
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16)
return -1;
if (channels < 1 || channels > FLAC_MAX_CHANNELS)
@@ -1335,6 +1335,6 @@ AVCodec flac_encoder = {
flac_encode_close,
NULL,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};

4
libavcodec/g722.c
View File

@@ -193,7 +193,7 @@ static av_cold int g722_init(AVCodecContext * avctx)
av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n");
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
switch (avctx->bits_per_coded_sample) {
case 8:
@@ -379,7 +379,7 @@ AVCodec adpcm_g722_encoder = {
.init = g722_init,
.encode = g722_encode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
};
#endif

4
libavcodec/g726.c
View File

@@ -332,7 +332,7 @@ static av_cold int g726_init(AVCodecContext * avctx)
avctx->coded_frame->key_frame = 1;
if (avctx->codec->decode)
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
@@ -401,7 +401,7 @@ AVCodec adpcm_g726_encoder = {
g726_close,
NULL,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif

2
libavcodec/gsmdec.c
View File

@@ -35,7 +35,7 @@ static av_cold int gsm_init(AVCodecContext *avctx)
avctx->channels = 1;
if (!avctx->sample_rate)
avctx->sample_rate = 8000;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
switch (avctx->codec_id) {
case CODEC_ID_GSM:

2
libavcodec/imc.c
View File

@@ -156,7 +156,7 @@ static av_cold int imc_decode_init(AVCodecContext * avctx)
ff_fft_init(&q->fft, 7, 1);
dsputil_init(&q->dsp, avctx);
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
return 0;
}

2
libavcodec/libfaac.c
View File

@@ -153,6 +153,6 @@ AVCodec libfaac_encoder = {
Faac_encode_init,
Faac_encode_frame,
Faac_encode_close,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Codec)"),
};

6
libavcodec/libgsm.c
View File

@@ -49,7 +49,7 @@ static av_cold int libgsm_init(AVCodecContext *avctx) {
if(!avctx->sample_rate)
avctx->sample_rate= 8000;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}else{
if (avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
@@ -120,7 +120,7 @@ AVCodec libgsm_encoder = {
libgsm_init,
libgsm_encode_frame,
libgsm_close,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
};
@@ -132,7 +132,7 @@ AVCodec libgsm_ms_encoder = {
libgsm_init,
libgsm_encode_frame,
libgsm_close,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
};

2
libavcodec/libmp3lame.c
View File

@@ -222,7 +222,7 @@ AVCodec libmp3lame_encoder = {
MP3lame_encode_frame,
MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.supported_samplerates= sSampleRates,
.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
};

4
libavcodec/libopencore-amr.c
View File

@@ -32,7 +32,7 @@ static void amr_decode_fix_avctx(AVCodecContext *avctx)
avctx->channels = 1;
avctx->frame_size = 160 * is_amr_wb;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
#if CONFIG_LIBOPENCORE_AMRNB
@@ -222,7 +222,7 @@ AVCodec libopencore_amrnb_encoder = {
amr_nb_encode_frame,
amr_nb_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("OpenCORE Adaptive Multi-Rate (AMR) Narrow-Band"),
};

2
libavcodec/libspeexdec.c
View File

@@ -49,7 +49,7 @@ static av_cold int libspeex_decode_init(AVCodecContext *avctx)
if (avctx->extradata_size >= 80)
s->header = speex_packet_to_header(avctx->extradata, avctx->extradata_size);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (s->header) {
avctx->sample_rate = s->header->rate;
avctx->channels = s->header->nb_channels;

2
libavcodec/libvorbis.c
View File

@@ -252,7 +252,7 @@ AVCodec libvorbis_encoder = {
oggvorbis_encode_frame,
oggvorbis_encode_close,
.capabilities= CODEC_CAP_DELAY,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name= NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
.priv_class= &class,
} ;

2
libavcodec/mace.c
View File

@@ -230,7 +230,7 @@ static av_cold int mace_decode_init(AVCodecContext * avctx)
{
if (avctx->channels > 2)
return -1;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

4
libavcodec/mlp_parser.c
View File

@@ -255,9 +255,9 @@ static int mlp_parse(AVCodecParserContext *s,
avctx->bits_per_raw_sample = mh.group1_bits;
if (avctx->bits_per_raw_sample > 16)
avctx->sample_fmt = SAMPLE_FMT_S32;
avctx->sample_fmt = AV_SAMPLE_FMT_S32;
else
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->sample_rate = mh.group1_samplerate;
avctx->frame_size = mh.access_unit_size;

6
libavcodec/mlpdec.c
View File

@@ -318,9 +318,9 @@ static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
m->avctx->bits_per_raw_sample = mh.group1_bits;
if (mh.group1_bits > 16)
m->avctx->sample_fmt = SAMPLE_FMT_S32;
m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
else
m->avctx->sample_fmt = SAMPLE_FMT_S16;
m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
m->params_valid = 1;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
@@ -931,7 +931,7 @@ static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
static int output_data(MLPDecodeContext *m, unsigned int substr,
uint8_t *data, unsigned int *data_size)
{
if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
if (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32)
return output_data_internal(m, substr, data, data_size, 1);
else
return output_data_internal(m, substr, data, data_size, 0);

2
libavcodec/mpc7.c
View File

@@ -85,7 +85,7 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx)
c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
c->frames_to_skip = 0;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
if(vlc_initialized) return 0;

2
libavcodec/mpc8.c
View File

@@ -129,7 +129,7 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx)
c->MSS = get_bits1(&gb);
c->frames = 1 << (get_bits(&gb, 3) * 2);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
if(vlc_initialized) return 0;

6
libavcodec/mpegaudio.h
View File

@@ -72,19 +72,19 @@
#if CONFIG_FLOAT
typedef float OUT_INT;
#define OUT_FMT SAMPLE_FMT_FLT
#define OUT_FMT AV_SAMPLE_FMT_FLT
#elif CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT
typedef int32_t OUT_INT;
#define OUT_MAX INT32_MAX
#define OUT_MIN INT32_MIN
#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31)
#define OUT_FMT SAMPLE_FMT_S32
#define OUT_FMT AV_SAMPLE_FMT_S32
#else
typedef int16_t OUT_INT;
#define OUT_MAX INT16_MAX
#define OUT_MIN INT16_MIN
#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
#define OUT_FMT SAMPLE_FMT_S16
#define OUT_FMT AV_SAMPLE_FMT_S16
#endif
#if CONFIG_FLOAT

2
libavcodec/mpegaudioenc.c
View File

@@ -792,7 +792,7 @@ AVCodec mp2_encoder = {
MPA_encode_frame,
MPA_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0},
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};

2
libavcodec/nellymoserdec.c
View File

@@ -147,7 +147,7 @@ static av_cold int decode_init(AVCodecContext * avctx) {
if (!ff_sine_128[127])
ff_init_ff_sine_windows(7);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = CH_LAYOUT_MONO;
return 0;
}

2
libavcodec/nellymoserenc.c
View File

@@ -392,5 +392,5 @@ AVCodec nellymoser_encoder = {
.close = encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("Nellymoser Asao"),
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
};

2
libavcodec/options.c
View File

@@ -461,7 +461,7 @@ void avcodec_get_context_defaults2(AVCodecContext *s, enum AVMediaType codec_typ
s->execute2= avcodec_default_execute2;
s->sample_aspect_ratio= (AVRational){0,1};
s->pix_fmt= PIX_FMT_NONE;
s->sample_fmt= SAMPLE_FMT_NONE;
s->sample_fmt= AV_SAMPLE_FMT_NONE;
s->palctrl = NULL;
s->reget_buffer= avcodec_default_reget_buffer;

20
libavcodec/pcm-mpeg.c
View File

@@ -72,8 +72,8 @@ static int pcm_bluray_parse_header(AVCodecContext *avctx,
av_log(avctx, AV_LOG_ERROR, "unsupported sample depth (0)\n");
return -1;
}
avctx->sample_fmt = avctx->bits_per_coded_sample == 16 ? SAMPLE_FMT_S16 :
SAMPLE_FMT_S32;
avctx->sample_fmt = avctx->bits_per_coded_sample == 16 ? AV_SAMPLE_FMT_S16 :
AV_SAMPLE_FMT_S32;
/* get the sample rate. Not all values are known or exist. */
switch (header[2] & 0x0f) {
@@ -146,7 +146,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
samples = buf_size / sample_size;
output_size = samples * avctx->channels *
(avctx->sample_fmt == SAMPLE_FMT_S32 ? 4 : 2);
(avctx->sample_fmt == AV_SAMPLE_FMT_S32 ? 4 : 2);
if (output_size > *data_size) {
av_log(avctx, AV_LOG_ERROR,
"Insufficient output buffer space (%d bytes, needed %d bytes)\n",
@@ -162,7 +162,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
case CH_LAYOUT_4POINT0:
case CH_LAYOUT_2_2:
samples *= num_source_channels;
if (SAMPLE_FMT_S16 == avctx->sample_fmt) {
if (AV_SAMPLE_FMT_S16 == avctx->sample_fmt) {
#if HAVE_BIGENDIAN
memcpy(dst16, src, output_size);
#else
@@ -181,7 +181,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
case CH_LAYOUT_SURROUND:
case CH_LAYOUT_2_1:
case CH_LAYOUT_5POINT0:
if (SAMPLE_FMT_S16 == avctx->sample_fmt) {
if (AV_SAMPLE_FMT_S16 == avctx->sample_fmt) {
do {
#if HAVE_BIGENDIAN
memcpy(dst16, src, avctx->channels * 2);
@@ -207,7 +207,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
break;
/* remapping: L, R, C, LBack, RBack, LF */
case CH_LAYOUT_5POINT1:
if (SAMPLE_FMT_S16 == avctx->sample_fmt) {
if (AV_SAMPLE_FMT_S16 == avctx->sample_fmt) {
do {
dst16[0] = bytestream_get_be16(&src);
dst16[1] = bytestream_get_be16(&src);
@@ -231,7 +231,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
break;
/* remapping: L, R, C, LSide, LBack, RBack, RSide, <unused> */
case CH_LAYOUT_7POINT0:
if (SAMPLE_FMT_S16 == avctx->sample_fmt) {
if (AV_SAMPLE_FMT_S16 == avctx->sample_fmt) {
do {
dst16[0] = bytestream_get_be16(&src);
dst16[1] = bytestream_get_be16(&src);
@@ -259,7 +259,7 @@ static int pcm_bluray_decode_frame(AVCodecContext *avctx,
break;
/* remapping: L, R, C, LSide, LBack, RBack, RSide, LF */
case CH_LAYOUT_7POINT1:
if (SAMPLE_FMT_S16 == avctx->sample_fmt) {
if (AV_SAMPLE_FMT_S16 == avctx->sample_fmt) {
do {
dst16[0] = bytestream_get_be16(&src);
dst16[1] = bytestream_get_be16(&src);
@@ -304,7 +304,7 @@ AVCodec pcm_bluray_decoder = {
NULL,
NULL,
pcm_bluray_decode_frame,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16, SAMPLE_FMT_S32,
SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 16|20|24-bit big-endian for Blu-ray media"),
};

56
libavcodec/pcm.c
View File

@@ -228,7 +228,7 @@ static av_cold int pcm_decode_init(AVCodecContext * avctx)
avctx->sample_fmt = avctx->codec->sample_fmts[0];
if (avctx->sample_fmt == SAMPLE_FMT_S32)
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
avctx->bits_per_raw_sample = av_get_bits_per_sample(avctx->codec->id);
return 0;
@@ -475,7 +475,7 @@ AVCodec name_ ## _encoder = { \
.init = pcm_encode_init, \
.encode = pcm_encode_frame, \
.close = pcm_encode_close, \
.sample_fmts = (const enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
.sample_fmts = (const enum AVSampleFormat[]){sample_fmt_,AV_SAMPLE_FMT_NONE}, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
};
#else
@@ -491,7 +491,7 @@ AVCodec name_ ## _decoder = { \
.priv_data_size = sizeof(PCMDecode), \
.init = pcm_decode_init, \
.decode = pcm_decode_frame, \
.sample_fmts = (const enum SampleFormat[]){sample_fmt_,SAMPLE_FMT_NONE}, \
.sample_fmts = (const enum AVSampleFormat[]){sample_fmt_,AV_SAMPLE_FMT_NONE}, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
};
#else
@@ -502,28 +502,28 @@ AVCodec name_ ## _decoder = { \
PCM_ENCODER(id,sample_fmt_,name,long_name_) PCM_DECODER(id,sample_fmt_,name,long_name_)
/* Note: Do not forget to add new entries to the Makefile as well. */
PCM_CODEC (CODEC_ID_PCM_ALAW, SAMPLE_FMT_S16, pcm_alaw, "PCM A-law");
PCM_CODEC (CODEC_ID_PCM_DVD, SAMPLE_FMT_S32, pcm_dvd, "PCM signed 20|24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_F32BE, SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
PCM_CODEC (CODEC_ID_PCM_F32LE, SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
PCM_CODEC (CODEC_ID_PCM_F64BE, SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
PCM_CODEC (CODEC_ID_PCM_F64LE, SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
PCM_DECODER(CODEC_ID_PCM_LXF, SAMPLE_FMT_S32, pcm_lxf, "PCM signed 20-bit little-endian planar");
PCM_CODEC (CODEC_ID_PCM_MULAW, SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law");
PCM_CODEC (CODEC_ID_PCM_S8, SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
PCM_CODEC (CODEC_ID_PCM_S16BE, SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S16LE, SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, SAMPLE_FMT_S16, pcm_s16le_planar, "PCM 16-bit little-endian planar");
PCM_CODEC (CODEC_ID_PCM_S24BE, SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S24DAUD, SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
PCM_CODEC (CODEC_ID_PCM_S24LE, SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_S32BE, SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S32LE, SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U8, SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
PCM_CODEC (CODEC_ID_PCM_U16BE, SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U16LE, SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U24BE, SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U24LE, SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U32BE, SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U32LE, SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_ZORK, SAMPLE_FMT_S16, pcm_zork, "PCM Zork");
PCM_CODEC (CODEC_ID_PCM_ALAW, AV_SAMPLE_FMT_S16, pcm_alaw, "PCM A-law");
PCM_CODEC (CODEC_ID_PCM_DVD, AV_SAMPLE_FMT_S32, pcm_dvd, "PCM signed 20|24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_F32BE, AV_SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
PCM_CODEC (CODEC_ID_PCM_F32LE, AV_SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
PCM_CODEC (CODEC_ID_PCM_F64BE, AV_SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
PCM_CODEC (CODEC_ID_PCM_F64LE, AV_SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
PCM_DECODER(CODEC_ID_PCM_LXF, AV_SAMPLE_FMT_S32, pcm_lxf, "PCM signed 20-bit little-endian planar");
PCM_CODEC (CODEC_ID_PCM_MULAW, AV_SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law");
PCM_CODEC (CODEC_ID_PCM_S8, AV_SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
PCM_CODEC (CODEC_ID_PCM_S16BE, AV_SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S16LE, AV_SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, AV_SAMPLE_FMT_S16, pcm_s16le_planar, "PCM 16-bit little-endian planar");
PCM_CODEC (CODEC_ID_PCM_S24BE, AV_SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S24DAUD, AV_SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
PCM_CODEC (CODEC_ID_PCM_S24LE, AV_SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_S32BE, AV_SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_S32LE, AV_SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U8, AV_SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
PCM_CODEC (CODEC_ID_PCM_U16BE, AV_SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U16LE, AV_SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
PCM_CODEC (CODEC_ID_PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
PCM_CODEC (CODEC_ID_PCM_ZORK, AV_SAMPLE_FMT_S16, pcm_zork, "PCM Zork");

2
libavcodec/qcelpdec.c
View File

@@ -92,7 +92,7 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx)
QCELPContext *q = avctx->priv_data;
int i;
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
for(i=0; i<10; i++)
q->prev_lspf[i] = (i+1)/11.;

2
libavcodec/qdm2.c
View File

@@ -1866,7 +1866,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
qdm2_init(s);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
// dump_context(s);
return 0;

2
libavcodec/ra144dec.c
View File

@@ -37,7 +37,7 @@ static av_cold int ra144_decode_init(AVCodecContext * avctx)
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/ra144enc.c
View File

@@ -38,7 +38,7 @@ static av_cold int ra144_encode_init(AVCodecContext * avctx)
{
RA144Context *ractx;
if (avctx->sample_fmt != SAMPLE_FMT_S16) {
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
return -1;
}

2
libavcodec/ra288.c
View File

@@ -54,7 +54,7 @@ typedef struct {
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
return 0;
}

22
libavcodec/resample.c
View File

@@ -47,7 +47,7 @@ struct ReSampleContext {
/* channel convert */
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum SampleFormat sample_fmt[2]; ///< input and output sample format
enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
unsigned sample_size[2]; ///< size of one sample in sample_fmt
short *buffer[2]; ///< buffers used for conversion to S16
unsigned buffer_size[2]; ///< sizes of allocated buffers
@@ -144,8 +144,8 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum SampleFormat sample_fmt_out,
enum SampleFormat sample_fmt_in,
enum AVSampleFormat sample_fmt_out,
enum AVSampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff)
{
@@ -178,8 +178,8 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3;
s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3;
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
s->sample_fmt[0], 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert %s sample format to s16 sample format\n",
@@ -189,9 +189,9 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
}
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
SAMPLE_FMT_S16, 1, NULL, 0))) {
AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
av_get_sample_fmt_name(s->sample_fmt[1]));
@@ -224,7 +224,7 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
{
return av_audio_resample_init(output_channels, input_channels,
output_rate, input_rate,
SAMPLE_FMT_S16, SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
TAPS, 10, 0, 0.8);
}
#endif
@@ -246,7 +246,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
return nb_samples;
}
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
int istride[1] = { s->sample_size[0] };
int ostride[1] = { 2 };
const void *ibuf[1] = { input };
@@ -276,7 +276,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
lenout= 4*nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
output_bak = output;
if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
@@ -341,7 +341,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
int istride[1] = { 2 };
int ostride[1] = { s->sample_size[1] };
const void *ibuf[1] = { output };

4
libavcodec/roqaudioenc.c
View File

@@ -49,7 +49,7 @@ static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
return -1;
}
if (avctx->sample_fmt != SAMPLE_FMT_S16) {
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
return -1;
}
@@ -162,6 +162,6 @@ AVCodec roq_dpcm_encoder = {
roq_dpcm_encode_frame,
roq_dpcm_encode_close,
NULL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
};

2
libavcodec/shorten.c
View File

@@ -105,7 +105,7 @@ static av_cold int shorten_decode_init(AVCodecContext * avctx)
{
ShortenContext *s = avctx->priv_data;
s->avctx = avctx;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/sipr.c
View File

@@ -493,7 +493,7 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx)
for (i = 0; i < 4; i++)
ctx->energy_history[i] = -14;
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
dsputil_init(&ctx->dsp, avctx);

2
libavcodec/smacker.c
View File

@@ -555,7 +555,7 @@ static av_cold int decode_end(AVCodecContext *avctx)
static av_cold int smka_decode_init(AVCodecContext *avctx)
{
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
avctx->sample_fmt = avctx->bits_per_coded_sample == 8 ? SAMPLE_FMT_U8 : SAMPLE_FMT_S16;
avctx->sample_fmt = avctx->bits_per_coded_sample == 8 ? AV_SAMPLE_FMT_U8 : AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/sonic.c
View File

@@ -825,7 +825,7 @@ static av_cold int sonic_decode_init(AVCodecContext *avctx)
}
s->int_samples = av_mallocz(4* s->frame_size);
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

2
libavcodec/truespeech.c
View File

@@ -56,7 +56,7 @@ static av_cold int truespeech_decode_init(AVCodecContext * avctx)
{
// TSContext *c = avctx->priv_data;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

10
libavcodec/tta.c
View File

@@ -246,15 +246,15 @@ static av_cold int tta_decode_init(AVCodecContext * avctx)
if (s->is_float)
{
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
av_log(s->avctx, AV_LOG_ERROR, "Unsupported sample format. Please contact the developers.\n");
return -1;
}
else switch(s->bps) {
// case 1: avctx->sample_fmt = SAMPLE_FMT_U8; break;
case 2: avctx->sample_fmt = SAMPLE_FMT_S16; break;
// case 3: avctx->sample_fmt = SAMPLE_FMT_S24; break;
case 4: avctx->sample_fmt = SAMPLE_FMT_S32; break;
// case 1: avctx->sample_fmt = AV_SAMPLE_FMT_U8; break;
case 2: avctx->sample_fmt = AV_SAMPLE_FMT_S16; break;
// case 3: avctx->sample_fmt = AV_SAMPLE_FMT_S24; break;
case 4: avctx->sample_fmt = AV_SAMPLE_FMT_S32; break;
default:
av_log(s->avctx, AV_LOG_ERROR, "Invalid/unsupported sample format. Please contact the developers.\n");
return -1;

2
libavcodec/twinvq.c
View File

@@ -1068,7 +1068,7 @@ static av_cold int twin_decode_init(AVCodecContext *avctx)
int ibps = avctx->bit_rate/(1000 * avctx->channels);
tctx->avctx = avctx;
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->channels > CHANNELS_MAX) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %i\n",

4
libavcodec/utils.c
View File

@@ -923,7 +923,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
}
av_strlcat(buf, ", ", buf_size);
avcodec_get_channel_layout_string(buf + strlen(buf), buf_size - strlen(buf), enc->channels, enc->channel_layout);
if (enc->sample_fmt != SAMPLE_FMT_NONE) {
if (enc->sample_fmt != AV_SAMPLE_FMT_NONE) {
snprintf(buf + strlen(buf), buf_size - strlen(buf),
", %s", av_get_sample_fmt_name(enc->sample_fmt));
}
@@ -1067,7 +1067,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){
}
#if FF_API_OLD_SAMPLE_FMT
int av_get_bits_per_sample_format(enum SampleFormat sample_fmt) {
int av_get_bits_per_sample_format(enum AVSampleFormat sample_fmt) {
return av_get_bits_per_sample_fmt(sample_fmt);
}
#endif

2
libavcodec/vmdav.c
View File

@@ -446,7 +446,7 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
s->channels = avctx->channels;
s->bits = avctx->bits_per_coded_sample;
s->block_align = avctx->block_align;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
av_log(s->avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, block align = %d, sample rate = %d\n",
s->channels, s->bits, s->block_align, avctx->sample_rate);

2
libavcodec/vorbis_dec.c
View File

@@ -1006,7 +1006,7 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
avccontext->channels = vc->audio_channels;
avccontext->sample_rate = vc->audio_samplerate;
avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
avccontext->sample_fmt = SAMPLE_FMT_S16;
avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
return 0 ;
}

2
libavcodec/vorbis_enc.c
View File

@@ -1111,6 +1111,6 @@ AVCodec vorbis_encoder = {
vorbis_encode_frame,
vorbis_encode_close,
.capabilities= CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
};

58
libavcodec/wavpack.c
View File

@@ -494,7 +494,7 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
B = s->decorr[i].samplesB[pos];
j = (pos + t) & 7;
}
if(type != SAMPLE_FMT_S16){
if(type != AV_SAMPLE_FMT_S16){
L2 = L + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
R2 = R + ((s->decorr[i].weightB * (int64_t)B + 512) >> 10);
}else{
@@ -506,13 +506,13 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
s->decorr[i].samplesA[j] = L = L2;
s->decorr[i].samplesB[j] = R = R2;
}else if(t == -1){
if(type != SAMPLE_FMT_S16)
if(type != AV_SAMPLE_FMT_S16)
L2 = L + ((s->decorr[i].weightA * (int64_t)s->decorr[i].samplesA[0] + 512) >> 10);
else
L2 = L + ((s->decorr[i].weightA * s->decorr[i].samplesA[0] + 512) >> 10);
UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, s->decorr[i].samplesA[0], L);
L = L2;
if(type != SAMPLE_FMT_S16)
if(type != AV_SAMPLE_FMT_S16)
R2 = R + ((s->decorr[i].weightB * (int64_t)L2 + 512) >> 10);
else
R2 = R + ((s->decorr[i].weightB * L2 + 512) >> 10);
@@ -520,7 +520,7 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
R = R2;
s->decorr[i].samplesA[0] = R;
}else{
if(type != SAMPLE_FMT_S16)
if(type != AV_SAMPLE_FMT_S16)
R2 = R + ((s->decorr[i].weightB * (int64_t)s->decorr[i].samplesB[0] + 512) >> 10);
else
R2 = R + ((s->decorr[i].weightB * s->decorr[i].samplesB[0] + 512) >> 10);
@@ -532,7 +532,7 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
s->decorr[i].samplesA[0] = R;
}
if(type != SAMPLE_FMT_S16)
if(type != AV_SAMPLE_FMT_S16)
L2 = L + ((s->decorr[i].weightA * (int64_t)R2 + 512) >> 10);
else
L2 = L + ((s->decorr[i].weightA * R2 + 512) >> 10);
@@ -546,10 +546,10 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
L += (R -= (L >> 1));
crc = (crc * 3 + L) * 3 + R;
if(type == SAMPLE_FMT_FLT){
if(type == AV_SAMPLE_FMT_FLT){
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, L);
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, R);
} else if(type == SAMPLE_FMT_S32){
} else if(type == AV_SAMPLE_FMT_S32){
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, L);
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, R);
} else {
@@ -613,7 +613,7 @@ static inline int wv_unpack_mono(WavpackContext *s, GetBitContext *gb, void *dst
A = s->decorr[i].samplesA[pos];
j = (pos + t) & 7;
}
if(type != SAMPLE_FMT_S16)
if(type != AV_SAMPLE_FMT_S16)
S = T + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
else
S = T + ((s->decorr[i].weightA * A + 512) >> 10);
@@ -623,9 +623,9 @@ static inline int wv_unpack_mono(WavpackContext *s, GetBitContext *gb, void *dst
pos = (pos + 1) & 7;
crc = crc * 3 + S;
if(type == SAMPLE_FMT_FLT)
if(type == AV_SAMPLE_FMT_FLT)
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, S);
else if(type == SAMPLE_FMT_S32)
else if(type == AV_SAMPLE_FMT_S32)
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, S);
else
*dst16++ = wv_get_value_integer(s, &crc_extra_bits, S);
@@ -662,9 +662,9 @@ static av_cold int wavpack_decode_init(AVCodecContext *avctx)
s->avctx = avctx;
s->stereo = (avctx->channels == 2);
if(avctx->bits_per_coded_sample <= 16)
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
else
avctx->sample_fmt = SAMPLE_FMT_S32;
avctx->sample_fmt = AV_SAMPLE_FMT_S32;
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
wv_reset_saved_context(s);
@@ -708,13 +708,13 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
s->frame_flags = AV_RL32(buf); buf += 4;
if(s->frame_flags&0x80){
bpp = sizeof(float);
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
} else if((s->frame_flags&0x03) <= 1){
bpp = 2;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
} else {
bpp = 4;
avctx->sample_fmt = SAMPLE_FMT_S32;
avctx->sample_fmt = AV_SAMPLE_FMT_S32;
}
s->stereo_in = (s->frame_flags & WV_FALSE_STEREO) ? 0 : s->stereo;
s->joint = s->frame_flags & WV_JOINT_STEREO;
@@ -945,11 +945,11 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
av_log(avctx, AV_LOG_ERROR, "Packed samples not found\n");
return -1;
}
if(!got_float && avctx->sample_fmt == SAMPLE_FMT_FLT){
if(!got_float && avctx->sample_fmt == AV_SAMPLE_FMT_FLT){
av_log(avctx, AV_LOG_ERROR, "Float information not found\n");
return -1;
}
if(s->got_extra_bits && avctx->sample_fmt != SAMPLE_FMT_FLT){
if(s->got_extra_bits && avctx->sample_fmt != AV_SAMPLE_FMT_FLT){
const int size = get_bits_left(&s->gb_extra_bits);
const int wanted = s->samples * s->extra_bits << s->stereo_in;
if(size < wanted){
@@ -969,22 +969,22 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
}
if(s->stereo_in){
if(avctx->sample_fmt == SAMPLE_FMT_S16)
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_S16);
else if(avctx->sample_fmt == SAMPLE_FMT_S32)
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_S32);
if(avctx->sample_fmt == AV_SAMPLE_FMT_S16)
samplecount = wv_unpack_stereo(s, &s->gb, samples, AV_SAMPLE_FMT_S16);
else if(avctx->sample_fmt == AV_SAMPLE_FMT_S32)
samplecount = wv_unpack_stereo(s, &s->gb, samples, AV_SAMPLE_FMT_S32);
else
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_FLT);
samplecount = wv_unpack_stereo(s, &s->gb, samples, AV_SAMPLE_FMT_FLT);
}else{
if(avctx->sample_fmt == SAMPLE_FMT_S16)
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_S16);
else if(avctx->sample_fmt == SAMPLE_FMT_S32)
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_S32);
if(avctx->sample_fmt == AV_SAMPLE_FMT_S16)
samplecount = wv_unpack_mono(s, &s->gb, samples, AV_SAMPLE_FMT_S16);
else if(avctx->sample_fmt == AV_SAMPLE_FMT_S32)
samplecount = wv_unpack_mono(s, &s->gb, samples, AV_SAMPLE_FMT_S32);
else
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_FLT);
samplecount = wv_unpack_mono(s, &s->gb, samples, AV_SAMPLE_FMT_FLT);
if(s->stereo && avctx->sample_fmt == SAMPLE_FMT_S16){
if(s->stereo && avctx->sample_fmt == AV_SAMPLE_FMT_S16){
int16_t *dst = (int16_t*)samples + samplecount * 2;
int16_t *src = (int16_t*)samples + samplecount;
int cnt = samplecount;
@@ -993,7 +993,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
*--dst = *src;
}
samplecount *= 2;
}else if(s->stereo && avctx->sample_fmt == SAMPLE_FMT_S32){
}else if(s->stereo && avctx->sample_fmt == AV_SAMPLE_FMT_S32){
int32_t *dst = (int32_t*)samples + samplecount * 2;
int32_t *src = (int32_t*)samples + samplecount;
int cnt = samplecount;

2
libavcodec/wmadec.c
View File

@@ -123,7 +123,7 @@ static int wma_decode_init(AVCodecContext * avctx)
wma_lsp_to_curve_init(s, s->frame_len);
}
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

4
libavcodec/wmaenc.c
View File

@@ -392,7 +392,7 @@ AVCodec wmav1_encoder =
encode_init,
encode_superframe,
ff_wma_end,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
};
@@ -405,6 +405,6 @@ AVCodec wmav2_encoder =
encode_init,
encode_superframe,
ff_wma_end,
.sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};

2
libavcodec/wmaprodec.c
View File

@@ -276,7 +276,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx);
init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE);
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->extradata_size >= 18) {
s->decode_flags = AV_RL16(edata_ptr+14);

2
libavcodec/wmavoice.c
View File

@@ -425,7 +425,7 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
ctx->sample_fmt = SAMPLE_FMT_FLT;
ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
return 0;
}

2
libavcodec/ws-snd1.c
View File

@@ -43,7 +43,7 @@ static av_cold int ws_snd_decode_init(AVCodecContext * avctx)
{
// WSSNDContext *c = avctx->priv_data;
avctx->sample_fmt = SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}

4
libavfilter/avfilter.c
View File

@@ -115,7 +115,7 @@ int avfilter_link(AVFilterContext *src, unsigned srcpad,
link->srcpad = &src->output_pads[srcpad];
link->dstpad = &dst->input_pads[dstpad];
link->type = src->output_pads[srcpad].type;
assert(PIX_FMT_NONE == -1 && SAMPLE_FMT_NONE == -1);
assert(PIX_FMT_NONE == -1 && AV_SAMPLE_FMT_NONE == -1);
link->format = -1;
return 0;
@@ -268,7 +268,7 @@ AVFilterBufferRef *avfilter_get_video_buffer(AVFilterLink *link, int perms, int
}
AVFilterBufferRef *avfilter_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar)
{
AVFilterBufferRef *ret = NULL;

8
libavfilter/avfilter.h
View File

@@ -366,7 +366,7 @@ struct AVFilterPad {
* Input audio pads only.
*/
AVFilterBufferRef *(*get_audio_buffer)(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar);
/**
@@ -455,7 +455,7 @@ AVFilterBufferRef *avfilter_default_get_video_buffer(AVFilterLink *link,
/** default handler for get_audio_buffer() for audio inputs */
AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar);
/**
@@ -486,7 +486,7 @@ AVFilterBufferRef *avfilter_null_get_video_buffer(AVFilterLink *link,
/** get_audio_buffer() handler for filters which simply pass audio along */
AVFilterBufferRef *avfilter_null_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar);
/**
@@ -662,7 +662,7 @@ AVFilterBufferRef *avfilter_get_video_buffer(AVFilterLink *link, int perms,
* avfilter_unref_buffer when you are finished with it.
*/
AVFilterBufferRef *avfilter_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar);
/**

4
libavfilter/defaults.c
View File

@@ -82,7 +82,7 @@ fail:
}
AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int planar)
{
AVFilterBuffer *samples = av_mallocz(sizeof(AVFilterBuffer));
@@ -318,7 +318,7 @@ AVFilterBufferRef *avfilter_null_get_video_buffer(AVFilterLink *link, int perms,
}
AVFilterBufferRef *avfilter_null_get_audio_buffer(AVFilterLink *link, int perms,
enum SampleFormat sample_fmt, int size,
enum AVSampleFormat sample_fmt, int size,
int64_t channel_layout, int packed)
{
return avfilter_get_audio_buffer(link->dst->outputs[0], perms, sample_fmt,

2
libavfilter/formats.c
View File

@@ -108,7 +108,7 @@ AVFilterFormats *avfilter_all_formats(enum AVMediaType type)
AVFilterFormats *ret = NULL;
int fmt;
int num_formats = type == AVMEDIA_TYPE_VIDEO ? PIX_FMT_NB :
type == AVMEDIA_TYPE_AUDIO ? SAMPLE_FMT_NB : 0;
type == AVMEDIA_TYPE_AUDIO ? AV_SAMPLE_FMT_NB : 0;
for (fmt = 0; fmt < num_formats; fmt++)
if ((type != AVMEDIA_TYPE_VIDEO) ||

2
libavformat/flic.c
View File

@@ -157,7 +157,7 @@ static int flic_read_header(AVFormatContext *s,
ast->codec->codec_tag = 0;
ast->codec->sample_rate = FLIC_TFTD_SAMPLE_RATE;
ast->codec->channels = 1;
ast->codec->sample_fmt = SAMPLE_FMT_U8;
ast->codec->sample_fmt = AV_SAMPLE_FMT_U8;
ast->codec->bit_rate = st->codec->sample_rate * 8;
ast->codec->bits_per_coded_sample = 8;
ast->codec->channel_layout = CH_LAYOUT_MONO;

2
libavformat/output-example.c
View File

@@ -68,7 +68,7 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
c->codec_type = AVMEDIA_TYPE_AUDIO;
/* put sample parameters */
c->sample_fmt = SAMPLE_FMT_S16;
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;

2
libavformat/utils.c
View File

@@ -2015,7 +2015,7 @@ static int has_codec_parameters(AVCodecContext *enc)
int val;
switch(enc->codec_type) {
case AVMEDIA_TYPE_AUDIO:
val = enc->sample_rate && enc->channels && enc->sample_fmt != SAMPLE_FMT_NONE;
val = enc->sample_rate && enc->channels && enc->sample_fmt != AV_SAMPLE_FMT_NONE;
if(!enc->frame_size &&
(enc->codec_id == CODEC_ID_VORBIS ||
enc->codec_id == CODEC_ID_AAC ||