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authorMichael Niedermayer <michaelni@gmx.at>2012-05-08 21:10:56 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-05-08 22:02:59 +0200
commitb4178a3f13784604281dc3da31383783676b8fec (patch)
tree5cbcf5e4bf9288ee4743e47e90a3dcacb45f3df0 /libavcodec/utils.c
parentb4b58485135dbc37a6cf8a57196157b1d67d13e1 (diff)
parentb2e495afa8e23b46536e25e892157104437f4020 (diff)
downloadffmpeg-b4178a3f13784604281dc3da31383783676b8fec.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: rtmp: Support 'rtmp_live', an option which specifies if the media is a live stream. av_samples_fill_array: Mark unmodified function argument as const. lagarith: add YUY2 decoding support Support decoding unaligned rgb24 lagarith. dv: Split profile handling code into a separate file. flvenc: use AVFormatContext, not AVCodecContext for logging. mov: Remove write-only variable in mov_read_chan(). fate: Change the probe-format refs to match the final text format committed. fate: Add avprobe as a make dependency Add probe fate tests to test for regressions in detecting media types. fate: Add oneline comparison method qdm2: clip array indices returned by qdm2_get_vlc(). avplay: properly close/reopen AVAudioResampleContext on channel layout change avcodec: do not needlessly set packet size to 0 in avcodec_encode_audio2() avcodec: for audio encoding, reset output packet when it is not valid avcodec: refactor avcodec_encode_audio2() to merge common branches avcodec: remove fallbacks for AVCodec.encode() in avcodec_encode_audio2() Conflicts: ffplay.c libavcodec/Makefile libavcodec/dvdata.c libavcodec/dvdata.h libavcodec/qdm2.c libavcodec/utils.c libavformat/flvenc.c libavformat/mov.c tests/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/utils.c')
-rw-r--r--libavcodec/utils.c83
1 files changed, 9 insertions, 74 deletions
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index addae0413a..79b355472e 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -1008,7 +1008,6 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
{
int ret;
AVPacket user_pkt = *avpkt;
- int nb_samples;
int needs_realloc = !user_pkt.data;
*got_packet_ptr = 0;
@@ -1016,27 +1015,23 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
if (!(avctx->codec->capabilities & CODEC_CAP_DELAY) && !frame) {
av_free_packet(avpkt);
av_init_packet(avpkt);
- avpkt->size = 0;
return 0;
}
/* 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)
+ if (frame->nb_samples > avctx->frame_size)
return AVERROR(EINVAL);
} else if (!(avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) {
- if (nb_samples != avctx->frame_size)
+ if (frame->nb_samples != avctx->frame_size)
return AVERROR(EINVAL);
}
- } else {
- nb_samples = avctx->frame_size;
}
- if (avctx->codec->encode2) {
- ret = avctx->codec->encode2(avctx, avpkt, frame, got_packet_ptr);
- if (!ret && *got_packet_ptr) {
+ ret = avctx->codec->encode2(avctx, avpkt, frame, got_packet_ptr);
+ if (!ret) {
+ if (*got_packet_ptr) {
if (!(avctx->codec->capabilities & CODEC_CAP_DELAY)) {
if (avpkt->pts == AV_NOPTS_VALUE)
avpkt->pts = frame->pts;
@@ -1048,69 +1043,6 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
} 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_pkt.data) {
- if (avctx->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) {
- av_assert0(av_get_bits_per_sample(avctx->codec_id) != 0);
- if (!frame)
- return AVERROR(EINVAL);
- 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 += 2*FF_MIN_BUFFER_SIZE;
- }
- }
- if ((ret = ff_alloc_packet2(avctx, 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_pkt.data && avpkt->data != avctx->internal->byte_buffer)
- 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 = ff_samples_to_time_base(avctx,
- avctx->frame_size);
- }
- }
- avpkt->size = ret;
- *got_packet_ptr = (ret > 0);
- ret = 0;
- }
-
- if (fs_tmp)
- avctx->frame_size = fs_tmp;
}
if (avpkt->data && avpkt->data == avctx->internal->byte_buffer) {
needs_realloc = 0;
@@ -1141,8 +1073,11 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
avctx->frame_number++;
}
- if (ret < 0 || !*got_packet_ptr)
+ if (ret < 0 || !*got_packet_ptr) {
av_free_packet(avpkt);
+ av_init_packet(avpkt);
+ return ret;
+ }
/* 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