lavc: remove disabled FF_API_OLD_ENCODE_AUDIO cruft

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);

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);

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);

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 },

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.
  *

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;
 }
 

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);

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 },

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.

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) {

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);

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;

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)"),

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;

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;
 }
 

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;

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;

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;
 }
 

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);

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[]){

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);

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);

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);

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)

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

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);

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;
 }
 

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;