diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 5d15e85bde6fcc4e359d57f2055f8eb58ee417bb..60eca59ae0c705b7d5f6a73765758973f55fa21d 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -683,9 +683,6 @@ 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;
}
@@ -719,11 +716,6 @@ 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);
diff --git a/libavcodec/ac3enc.c b/libavcodec/ac3enc.c
index b3be2dbf107a144b02e2ff34dea21287a5a35dea..df86f0bd94228e5a5820e8a97616762b6901781e 100644
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -2052,9 +2052,6 @@ 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;
}
@@ -2484,14 +2481,6 @@ 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);
diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
index f81d7fde83844ffae2792c5c8bc9e161012860b1..9bcbc42b23b578bc366b6db7fd404beedd12c4ac 100644
--- a/libavcodec/adpcmenc.c
+++ b/libavcodec/adpcmenc.c
@@ -142,11 +142,6 @@ 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:
av_freep(&s->paths);
@@ -159,9 +154,6 @@ 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);
diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c
index 7a9c06a5915a29dce589dbfb9a4e1c765532a9ae..47620a22dd82ace12cdf676ce5b708259d7c1cf6 100644
--- a/libavcodec/adxenc.c
+++ b/libavcodec/adxenc.c
@@ -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);
@@ -177,9 +164,6 @@ 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 },
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 7bc7b618c751989ba271eb646bdc2fcafe94ae59..0d4ac318cdb47b55f4fa8eee3ff99c60889fcb2e 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3685,36 +3685,6 @@ AVCodec *avcodec_find_encoder(enum AVCodecID id);
*/
AVCodec *avcodec_find_encoder_by_name(const char *name);
-#if FF_API_OLD_ENCODE_AUDIO
-/**
- * Encode an audio frame from samples into buf.
- *
- * @deprecated Use avcodec_encode_audio2 instead.
- *
- * @note The output buffer should be at least FF_MIN_BUFFER_SIZE bytes large.
- * However, for codecs with avctx->frame_size equal to 0 (e.g. PCM) the user
- * will know how much space is needed because it depends on the value passed
- * in buf_size as described below. In that case a lower value can be used.
- *
- * @param avctx the codec context
- * @param[out] buf the output buffer
- * @param[in] buf_size the output buffer size
- * @param[in] samples the input buffer containing the samples
- * The number of samples read from this buffer is frame_size*channels,
- * both of which are defined in avctx.
- * For codecs which have avctx->frame_size equal to 0 (e.g. PCM) the number of
- * samples read from samples is equal to:
- * buf_size * 8 / (avctx->channels * av_get_bits_per_sample(avctx->codec_id))
- * This also implies that av_get_bits_per_sample() must not return 0 for these
- * codecs.
- * @return On error a negative value is returned, on success zero or the number
- * of bytes used to encode the data read from the input buffer.
- */
-int attribute_deprecated avcodec_encode_audio(AVCodecContext *avctx,
- uint8_t *buf, int buf_size,
- const short *samples);
-#endif
-
/**
* Encode a frame of audio.
*
diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 7808e2059c0a07a3e771353ab8766ba6e91f807d..1699312c8cca4effea5b64f77755a8e0d30e30f5 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -394,12 +394,6 @@ 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
-
ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON);
@@ -1285,9 +1279,6 @@ 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;
}
diff --git a/libavcodec/g722enc.c b/libavcodec/g722enc.c
index 11d3f209337e0d596bd21a6fe5e714d171dce1fc..b168a92d553dcb6264450c7a9f66a5c7758e5535 100644
--- a/libavcodec/g722enc.c
+++ b/libavcodec/g722enc.c
@@ -52,9 +52,6 @@ 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;
}
@@ -122,14 +119,6 @@ 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);
diff --git a/libavcodec/g726.c b/libavcodec/g726.c
index e1448a148749d08257041eab4aec7083a9def1a1..6db00feb31f5d79716154f3e556e2a955319c7aa 100644
--- a/libavcodec/g726.c
+++ b/libavcodec/g726.c
@@ -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)
{
@@ -404,9 +389,6 @@ 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 },
diff --git a/libavcodec/internal.h b/libavcodec/internal.h
index 3f28d4b00ac8e82fd7f81ad781dce7c72e2fa240..65437edc548e84a0a8d80453de2d2aed02fdb149 100644
--- a/libavcodec/internal.h
+++ b/libavcodec/internal.h
@@ -76,14 +76,6 @@ typedef struct AVCodecInternal {
*/
int allocate_progress;
-#if FF_API_OLD_ENCODE_AUDIO
- /**
- * Internal sample count used by avcodec_encode_audio() to fabricate pts.
- * Can be removed along with avcodec_encode_audio().
- */
- int sample_count;
-#endif
-
/**
* An audio frame with less than required samples has been submitted and
* padded with silence. Reject all subsequent frames.
diff --git a/libavcodec/libfaac.c b/libavcodec/libfaac.c
index d32e77667845b2fef5f6de141eb909b724f67c6b..63593441927fa4e765e6069fe57882927e355ff4 100644
--- a/libavcodec/libfaac.c
+++ b/libavcodec/libfaac.c
@@ -46,9 +46,6 @@ 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);
@@ -133,14 +130,6 @@ 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) {
diff --git a/libavcodec/libfdk-aacenc.c b/libavcodec/libfdk-aacenc.c
index db32e9fe4a5b814805a7fc549374c82c43f540e7..ef154c1adbade84a78fbd8709f610f81c203418e 100644
--- a/libavcodec/libfdk-aacenc.c
+++ b/libavcodec/libfdk-aacenc.c
@@ -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);
diff --git a/libavcodec/libgsm.c b/libavcodec/libgsm.c
index 3159f286da35a1fc90db6c6b67d97f9b93925f6e..ddf7b2367de286acb81f93882c5f6f18ef1436ee 100644
--- a/libavcodec/libgsm.c
+++ b/libavcodec/libgsm.c
@@ -77,21 +77,10 @@ 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) {
- gsm_destroy(avctx->priv_data);
- return AVERROR(ENOMEM);
- }
-#endif
-
return 0;
}
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;
diff --git a/libavcodec/libilbc.c b/libavcodec/libilbc.c
index d47227cadadc87ea43ea0998191a6d3bcaa0d535..714fdb02106aa8b7f2386196b585c03f664a487e 100644
--- a/libavcodec/libilbc.c
+++ b/libavcodec/libilbc.c
@@ -153,23 +153,10 @@ 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)
{
@@ -200,7 +187,6 @@ 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)"),
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 2e501cac0bf836274625fdaea00fc5013a3f4f94..5e7caf8c0ab03897d5472bc061e2285747ce90ad 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -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);
@@ -142,14 +139,6 @@ 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;
diff --git a/libavcodec/libopencore-amr.c b/libavcodec/libopencore-amr.c
index 71a0edbfdf862b02ed06de0259f7bc31e877be8f..d46ddd7c9d31c19c09a90b6389b7a22a73352f45 100644
--- a/libavcodec/libopencore-amr.c
+++ b/libavcodec/libopencore-amr.c
@@ -202,11 +202,6 @@ 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) {
@@ -227,9 +222,6 @@ 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;
}
diff --git a/libavcodec/libspeexenc.c b/libavcodec/libspeexenc.c
index 4277e62e4c96c02fab41b4fa5f0f150f72687789..9469a7628813d660a3ada095a0098e5cbb1946c5 100644
--- a/libavcodec/libspeexenc.c
+++ b/libavcodec/libspeexenc.c
@@ -251,16 +251,6 @@ 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);
@@ -329,9 +319,6 @@ 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;
diff --git a/libavcodec/libvo-aacenc.c b/libavcodec/libvo-aacenc.c
index 31822b5d733be7279ae38426d6bd3c5a41fddcbc..3e5efb3466ab23f0ddde0a244ea203a2f9a432db 100644
--- a/libavcodec/libvo-aacenc.c
+++ b/libavcodec/libvo-aacenc.c
@@ -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;
diff --git a/libavcodec/libvo-amrwbenc.c b/libavcodec/libvo-amrwbenc.c
index 650245646233e4849181515bd9faa1d5ec77cd2b..af6ddf4ed8fa9ba9c17489b598ce722230dbed77 100644
--- a/libavcodec/libvo-amrwbenc.c
+++ b/libavcodec/libvo-amrwbenc.c
@@ -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;
}
diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 092cbbc0a7fde660ace4005b36268f2436b55716..8e02057a38a5eaa441450286265a8e1e7d839f6d 100644
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -162,9 +162,6 @@ 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;
@@ -241,14 +238,6 @@ 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);
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index ba0d8cfcab480896ebec934ac9ccc1638691a208..76e7b88fce2d8bff3f3bf688afbff114fec3727d 100644
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -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;
}
@@ -771,14 +765,6 @@ 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 },
@@ -791,7 +777,6 @@ 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[]){
diff --git a/libavcodec/nellymoserenc.c b/libavcodec/nellymoserenc.c
index 8721c26f5f52cc0dc322c79ab3339ed160c98728..98fef23627160cdcfe749848bb6c2bdbe122dd8b 100644
--- a/libavcodec/nellymoserenc.c
+++ b/libavcodec/nellymoserenc.c
@@ -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);
diff --git a/libavcodec/ra144enc.c b/libavcodec/ra144enc.c
index b9473ac197603839f8646fd2f4da7eee7085a6c8..c05c2435d7ae7088b890001258cd1c76e0d36d16 100644
--- a/libavcodec/ra144enc.c
+++ b/libavcodec/ra144enc.c
@@ -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);
diff --git a/libavcodec/roqaudioenc.c b/libavcodec/roqaudioenc.c
index 3cc9931a333d67bdf340fc63d87a7d7a8f6526f0..d440fbe9448c055aa3bff11203c6a65c6bf5af5c 100644
--- a/libavcodec/roqaudioenc.c
+++ b/libavcodec/roqaudioenc.c
@@ -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);
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index ed39c0c3f14176e0203f456914de316c49830f98..cc7d6d01508439ad6e5f108fa0f79aa896fba7c9 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -1245,87 +1245,6 @@ end:
return ret;
}
-#if FF_API_OLD_ENCODE_AUDIO
-int attribute_align_arg avcodec_encode_audio(AVCodecContext *avctx,
- uint8_t *buf, int buf_size,
- const short *samples)
-{
- AVPacket pkt;
- AVFrame frame0 = { { 0 } };
- AVFrame *frame;
- int ret, samples_size, got_packet;
-
- av_init_packet(&pkt);
- pkt.data = buf;
- pkt.size = buf_size;
-
- if (samples) {
- frame = &frame0;
- avcodec_get_frame_defaults(frame);
-
- if (avctx->frame_size) {
- frame->nb_samples = avctx->frame_size;
- } else {
- /* if frame_size is not set, the number of samples must be
- * calculated from the buffer size */
- int64_t nb_samples;
- if (!av_get_bits_per_sample(avctx->codec_id)) {
- av_log(avctx, AV_LOG_ERROR, "avcodec_encode_audio() does not "
- "support this codec\n");
- return AVERROR(EINVAL);
- }
- nb_samples = (int64_t)buf_size * 8 /
- (av_get_bits_per_sample(avctx->codec_id) *
- avctx->channels);
- if (nb_samples >= INT_MAX)
- return AVERROR(EINVAL);
- frame->nb_samples = nb_samples;
- }
-
- /* it is assumed that the samples buffer is large enough based on the
- * relevant parameters */
- samples_size = av_samples_get_buffer_size(NULL, avctx->channels,
- frame->nb_samples,
- avctx->sample_fmt, 1);
- if ((ret = avcodec_fill_audio_frame(frame, avctx->channels,
- avctx->sample_fmt,
- (const uint8_t *)samples,
- samples_size, 1)))
- return ret;
-
- /* fabricate frame pts from sample count.
- * this is needed because the avcodec_encode_audio() API does not have
- * a way for the user to provide pts */
- frame->pts = ff_samples_to_time_base(avctx,
- avctx->internal->sample_count);
- avctx->internal->sample_count += frame->nb_samples;
- } else {
- frame = NULL;
- }
-
- got_packet = 0;
- ret = avcodec_encode_audio2(avctx, &pkt, frame, &got_packet);
- if (!ret && got_packet && avctx->coded_frame) {
- avctx->coded_frame->pts = pkt.pts;
- avctx->coded_frame->key_frame = !!(pkt.flags & AV_PKT_FLAG_KEY);
- }
- /* free any side data since we cannot return it */
- if (pkt.side_data_elems > 0) {
- int i;
- for (i = 0; i < pkt.side_data_elems; i++)
- av_free(pkt.side_data[i].data);
- av_freep(&pkt.side_data);
- pkt.side_data_elems = 0;
- }
-
- if (frame && frame->extended_data != frame->data)
- av_free(frame->extended_data);
-
- return ret ? ret : pkt.size;
-}
-
-#endif
-
#if FF_API_OLD_ENCODE_VIDEO
int attribute_align_arg avcodec_encode_video(AVCodecContext *avctx, uint8_t *buf, int buf_size,
const AVFrame *pict)
diff --git a/libavcodec/version.h b/libavcodec/version.h
index f1cc74961a808b81ef20f6c1c564bb786bded169..efd5d34688b35d57f6b2a95e5ba1bff56de35fce 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -49,9 +49,6 @@
#ifndef FF_API_REQUEST_CHANNELS
#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
-#ifndef FF_API_OLD_ENCODE_AUDIO
-#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
#ifndef FF_API_OLD_ENCODE_VIDEO
#define FF_API_OLD_ENCODE_VIDEO (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index 3d4f806ae5df4d638d2ab3b2a8e3926bbe311cdb..db0394a7dc24c051f0e2f400e0804a697221855f 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -1156,9 +1156,6 @@ 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 ;
@@ -1190,14 +1187,6 @@ 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);
diff --git a/libavcodec/wma.c b/libavcodec/wma.c
index ab7bcf53a80be750bd35baa298c0874e3388cd69..03e310bc942f45dd767f10ab42a703b67af2fcbd 100644
--- a/libavcodec/wma.c
+++ b/libavcodec/wma.c
@@ -386,11 +386,6 @@ 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;
}
diff --git a/libavcodec/wmaenc.c b/libavcodec/wmaenc.c
index f110f8946525e81908a9888cc08a8ea8f3691353..adaa7b37e21f5b33820feb7bc41cd41312d45e82 100644
--- a/libavcodec/wmaenc.c
+++ b/libavcodec/wmaenc.c
@@ -52,11 +52,6 @@ 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;