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Merge remote-tracking branch 'qatar/master'

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* 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>
This commit is contained in:
Michael Niedermayer
2012-05-08 21:10:56 +02:00
parent b4b5848513 b2e495afa8
commit b4178a3f13
19 changed files with 591 additions and 492 deletions
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83
libavcodec/utils.c
View File

@ -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