Commit e052f065 authored by Michael Niedermayer's avatar Michael Niedermayer

Merge commit '0f24a3ca'

* commit '0f24a3ca':
  lavc: remove disabled FF_API_OLD_ENCODE_VIDEO cruft
  lavc: remove disabled FF_API_OLD_ENCODE_AUDIO cruft
  lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft

Conflicts:
	libavcodec/flacenc.c
	libavcodec/libgsm.c
	libavcodec/utils.c
	libavcodec/version.h

The compatibility wrapers are left as they likely sre still
in wide use. They will be removed when they break or otherwise
cause work without an volunteer being available.
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents eba6a04e 0f24a3ca
......@@ -679,9 +679,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
av_freep(&s->buffer.samples);
av_freep(&s->cpe);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -715,11 +712,6 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto alloc_fail;
#endif
return 0;
alloc_fail:
return AVERROR(ENOMEM);
......
......@@ -2049,9 +2049,6 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
s->mdct_end(s);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -2481,14 +2478,6 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
if (ret)
goto init_fail;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto init_fail;
}
#endif
ff_dsputil_init(&s->dsp, avctx);
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
......
......@@ -144,11 +144,6 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto error;
#endif
return 0;
error:
adpcm_encode_close(avctx);
......@@ -158,9 +153,6 @@ error:
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
......
......@@ -107,14 +107,6 @@ static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
return HEADER_SIZE;
}
#if FF_API_OLD_ENCODE_AUDIO
static av_cold int adx_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
#endif
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
ADXContext *c = avctx->priv_data;
......@@ -125,11 +117,6 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
}
avctx->frame_size = BLOCK_SAMPLES;
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
#endif
/* the cutoff can be adjusted, but this seems to work pretty well */
c->cutoff = 500;
ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
......@@ -175,9 +162,6 @@ AVCodec ff_adpcm_adx_encoder = {
.id = AV_CODEC_ID_ADPCM_ADX,
.priv_data_size = sizeof(ADXContext),
.init = adx_encode_init,
#if FF_API_OLD_ENCODE_AUDIO
.close = adx_encode_close,
#endif
.encode2 = adx_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
......
......@@ -547,11 +547,6 @@ typedef struct AVCodecDescriptor {
*/
#define AV_CODEC_PROP_BITMAP_SUB (1 << 16)
#if FF_API_OLD_DECODE_AUDIO
/* in bytes */
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
#endif
/**
* @ingroup lavc_decoding
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
......
......@@ -399,12 +399,6 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->frame_count = 0;
s->min_framesize = s->max_framesize;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
if (channels == 3 &&
avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
channels == 4 &&
......@@ -1310,9 +1304,6 @@ static av_cold int flac_encode_close(AVCodecContext *avctx)
}
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......
......@@ -53,9 +53,6 @@ static av_cold int g722_encode_close(AVCodecContext *avctx)
av_freep(&c->node_buf[i]);
av_freep(&c->nodep_buf[i]);
}
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -123,14 +120,6 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
g722_encode_close(avctx);
......
......@@ -331,13 +331,6 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
g726_reset(c);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
avctx->coded_frame->key_frame = 1;
#endif
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
......@@ -345,14 +338,6 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
return 0;
}
#if FF_API_OLD_ENCODE_AUDIO
static av_cold int g726_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
#endif
static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
......@@ -402,9 +387,6 @@ AVCodec ff_adpcm_g726_encoder = {
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
.encode2 = g726_encode_frame,
#if FF_API_OLD_ENCODE_AUDIO
.close = g726_encode_close,
#endif
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
......
......@@ -45,9 +45,6 @@ static av_cold int Faac_encode_close(AVCodecContext *avctx)
{
FaacAudioContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
......@@ -132,14 +129,6 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
avctx->frame_size = samples_input / avctx->channels;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* Set decoder specific info */
avctx->extradata_size = 0;
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
......
......@@ -97,9 +97,6 @@ static int aac_encode_close(AVCodecContext *avctx)
if (s->handle)
aacEncClose(&s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
......@@ -275,13 +272,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
avctx->frame_size = info.frameLength;
avctx->delay = info.encoderDelay;
ff_af_queue_init(avctx, &s->afq);
......
......@@ -41,9 +41,6 @@
#include "gsm.h"
static av_cold int libgsm_encode_close(AVCodecContext *avctx) {
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
gsm_destroy(avctx->priv_data);
avctx->priv_data = NULL;
return 0;
......@@ -88,12 +85,6 @@ static av_cold int libgsm_encode_init(AVCodecContext *avctx) {
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
goto error;
#endif
return 0;
error:
libgsm_encode_close(avctx);
......
......@@ -159,23 +159,10 @@ static av_cold int ilbc_encode_init(AVCodecContext *avctx)
avctx->block_align = s->encoder.no_of_bytes;
avctx->frame_size = s->encoder.blockl;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
return 0;
}
static av_cold int ilbc_encode_close(AVCodecContext *avctx)
{
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
static int ilbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
......@@ -204,7 +191,6 @@ AVCodec ff_libilbc_encoder = {
.priv_data_size = sizeof(ILBCEncContext),
.init = ilbc_encode_init,
.encode2 = ilbc_encode_frame,
.close = ilbc_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("iLBC (Internet Low Bitrate Codec)"),
......
......@@ -78,9 +78,6 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
......@@ -143,14 +140,6 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
avctx->frame_size = lame_get_framesize(s->gfp);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* allocate float sample buffers */
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
int ch;
......
......@@ -203,11 +203,6 @@ static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
avctx->frame_size = 160;
avctx->delay = 50;
ff_af_queue_init(avctx, &s->afq);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
s->enc_state = Encoder_Interface_init(s->enc_dtx);
if (!s->enc_state) {
......@@ -228,9 +223,6 @@ static av_cold int amr_nb_encode_close(AVCodecContext *avctx)
Encoder_Interface_exit(s->enc_state);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......
......@@ -252,16 +252,6 @@ static av_cold int encode_init(AVCodecContext *avctx)
av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
return AVERROR(ENOMEM);
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
av_freep(&avctx->extradata);
speex_header_free(header_data);
speex_encoder_destroy(s->enc_state);
av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
return AVERROR(ENOMEM);
}
#endif
/* copy header packet to extradata */
memcpy(avctx->extradata, header_data, header_size);
......@@ -328,9 +318,6 @@ static av_cold int encode_close(AVCodecContext *avctx)
speex_encoder_destroy(s->enc_state);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0;
......
......@@ -47,9 +47,6 @@ static int aac_encode_close(AVCodecContext *avctx)
AACContext *s = avctx->priv_data;
s->codec_api.Uninit(s->handle);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
av_freep(&s->end_buffer);
......@@ -63,11 +60,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
AACENC_PARAM params = { 0 };
int index, ret;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
avctx->frame_size = FRAME_SIZE;
avctx->delay = ENC_DELAY;
s->last_frame = 2;
......
......@@ -94,11 +94,6 @@ static av_cold int amr_wb_encode_init(AVCodecContext *avctx)
avctx->frame_size = 320;
avctx->delay = 80;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
s->state = E_IF_init();
......@@ -110,7 +105,6 @@ static int amr_wb_encode_close(AVCodecContext *avctx)
AMRWBContext *s = avctx->priv_data;
E_IF_exit(s->state);
av_freep(&avctx->coded_frame);
return 0;
}
......
......@@ -187,9 +187,6 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
av_fifo_free(s->pkt_fifo);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0;
......@@ -267,14 +264,6 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
oggvorbis_encode_close(avctx);
......
......@@ -184,12 +184,6 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
total_quant_bits[i] = 12 * v;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
#endif
return 0;
}
......@@ -769,14 +763,6 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
return 0;
}
static av_cold int MPA_encode_close(AVCodecContext *avctx)
{
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
static const AVCodecDefault mp2_defaults[] = {
{ "b", "128k" },
{ NULL },
......@@ -789,7 +775,6 @@ AVCodec ff_mp2_encoder = {
.priv_data_size = sizeof(MpegAudioContext),
.init = MPA_encode_init,
.encode2 = MPA_encode_frame,
.close = MPA_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = (const int[]){
......
......@@ -140,9 +140,6 @@ static av_cold int encode_end(AVCodecContext *avctx)
av_free(s->path);
}
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -187,14 +184,6 @@ static av_cold int encode_init(AVCodecContext *avctx)
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
encode_end(avctx);
......
......@@ -40,9 +40,6 @@ static av_cold int ra144_encode_close(AVCodecContext *avctx)
RA144Context *ractx = avctx->priv_data;
ff_lpc_end(&ractx->lpc_ctx);
ff_af_queue_close(&ractx->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......@@ -71,14 +68,6 @@ static av_cold int ra144_encode_init(AVCodecContext * avctx)
ff_af_queue_init(avctx, &ractx->afq);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
ra144_encode_close(avctx);
......
......@@ -46,9 +46,6 @@ static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
{
ROQDPCMContext *context = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&context->frame_buffer);
return 0;
......@@ -81,14 +78,6 @@ static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
context->lastSample[0] = context->lastSample[1] = 0;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
roq_dpcm_encode_close(avctx);
......
......@@ -58,14 +58,14 @@
#define FF_API_AVCODEC_OPEN (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
#ifndef FF_API_OLD_DECODE_AUDIO
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
#ifndef FF_API_OLD_TIMECODE
#define FF_API_OLD_TIMECODE (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
#ifndef FF_API_OLD_ENCODE_AUDIO
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
#ifndef FF_API_OLD_ENCODE_VIDEO
#define FF_API_OLD_ENCODE_VIDEO (LIBAVCODEC_VERSION_MAJOR < 56)
......
......@@ -1153,9 +1153,6 @@ static av_cold int vorbis_encode_close(AVCodecContext *avctx)
ff_mdct_end(&venc->mdct[0]);
ff_mdct_end(&venc->mdct[1]);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0 ;
......@@ -1187,14 +1184,6 @@ static av_cold int vorbis_encode_init(AVCodecContext *avctx)
avctx->frame_size = 1 << (venc->log2_blocksize[0] - 1);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
vorbis_encode_close(avctx);
......
......@@ -390,11 +390,6 @@ int ff_wma_end(AVCodecContext *avctx)
av_free(s->int_table[i]);
}
#if FF_API_OLD_ENCODE_AUDIO
if (av_codec_is_encoder(avctx->codec))
av_freep(&avctx->coded_frame);
#endif
return 0;
}
......
......@@ -50,11 +50,6 @@ static int encode_init(AVCodecContext * avctx){
return AVERROR(EINVAL);
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
#endif
/* extract flag infos */
flags1 = 0;
flags2 = 1;
......
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