+ }
+
+ /* check for valid frame size */
+ if (frame) {
+ nb_samples = frame->nb_samples;
+ if (avctx->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) {
+ if (nb_samples > avctx->frame_size)
+ return AVERROR(EINVAL);
+ } else if (!(avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
+ if (nb_samples != avctx->frame_size)
+ return AVERROR(EINVAL);
+ }
+ } else {
+ nb_samples = avctx->frame_size;
+ }
+
+ if (avctx->codec->encode2) {
+ *got_packet_ptr = 0;
+ ret = avctx->codec->encode2(avctx, avpkt, frame, got_packet_ptr);
+ if (!ret && *got_packet_ptr) {
+ if (!(avctx->codec->capabilities & CODEC_CAP_DELAY)) {
+ avpkt->pts = frame->pts;
+ avpkt->duration = av_rescale_q(frame->nb_samples,
+ (AVRational){ 1, avctx->sample_rate },
+ avctx->time_base);
+ }
+ avpkt->dts = avpkt->pts;
+ } else {
+ avpkt->size = 0;
+ }
+ } else {
+ /* for compatibility with encoders not supporting encode2(), we need to
+ allocate a packet buffer if the user has not provided one or check
+ the size otherwise */
+ int fs_tmp = 0;
+ int buf_size = avpkt->size;
+ if (!user_packet) {
+ if (avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) {
+ av_assert0(av_get_bits_per_sample(avctx->codec_id) != 0);
+ buf_size = nb_samples * avctx->channels *
+ av_get_bits_per_sample(avctx->codec_id) / 8;
+ } else {
+ /* this is a guess as to the required size.
+ if an encoder needs more than this, it should probably
+ implement encode2() */
+ buf_size = 2 * avctx->frame_size * avctx->channels *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ buf_size += FF_MIN_BUFFER_SIZE;
+ }
+ }
+ if ((ret = ff_alloc_packet(avpkt, buf_size)))
+ return ret;
+
+ /* Encoders using AVCodec.encode() that support
+ CODEC_CAP_SMALL_LAST_FRAME require avctx->frame_size to be set to
+ the smaller size when encoding the last frame.
+ This code can be removed once all encoders supporting
+ CODEC_CAP_SMALL_LAST_FRAME use encode2() */
+ if ((avctx->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) &&
+ nb_samples < avctx->frame_size) {
+ fs_tmp = avctx->frame_size;
+ avctx->frame_size = nb_samples;
+ }
+
+ /* encode the frame */
+ ret = avctx->codec->encode(avctx, avpkt->data, avpkt->size,
+ frame ? frame->data[0] : NULL);
+ if (ret >= 0) {
+ if (!ret) {
+ /* no output. if the packet data was allocated by libavcodec,
+ free it */
+ if (!user_packet)
+ av_freep(&avpkt->data);
+ } else {
+ if (avctx->coded_frame)
+ avpkt->pts = avpkt->dts = avctx->coded_frame->pts;
+ /* Set duration for final small packet. This can be removed
+ once all encoders supporting CODEC_CAP_SMALL_LAST_FRAME use
+ encode2() */
+ if (fs_tmp) {
+ avpkt->duration = av_rescale_q(avctx->frame_size,
+ (AVRational){ 1, avctx->sample_rate },
+ avctx->time_base);
+ }
+ }
+ avpkt->size = ret;
+ *got_packet_ptr = (ret > 0);
+ ret = 0;
+ }
+
+ if (fs_tmp)
+ avctx->frame_size = fs_tmp;
+ }
+ if (!ret)
+ avctx->frame_number++;
+
+ /* NOTE: if we add any audio encoders which output non-keyframe packets,
+ this needs to be moved to the encoders, but for now we can do it
+ here to simplify things */
+ avpkt->flags |= AV_PKT_FLAG_KEY;
+
+ return ret;
+}
+
+#if FF_API_OLD_DECODE_AUDIO
+int attribute_align_arg avcodec_encode_audio(AVCodecContext *avctx,
+ uint8_t *buf, int buf_size,
+ const short *samples)
+{
+ AVPacket pkt;
+ AVFrame frame0;
+ 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,
+ 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 = av_rescale_q(avctx->internal->sample_count,
+ (AVRational){ 1, avctx->sample_rate },
+ avctx->time_base);
+ 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;