Merge commit '2df0c32e'

* commit '2df0c32e':
  lavc: use a separate field for exporting audio encoder padding

Conflicts:
	libavcodec/audio_frame_queue.c
	libavcodec/avcodec.h
	libavcodec/libvorbisenc.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmaenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

doc/APIchanges

@@ -15,6 +15,10 @@ libavutil:     2014-08-09
 
 API changes, most recent first:
 
+2014-10-13 - xxxxxxx - lavc 55.03.0 - avcodec.h
+  Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay
+  for audio encoding.
+
 2014-09-xx - xxxxxxx - lavu 54.04.0 - pixdesc.h
   Add API to return the name of frame and context color properties.
 

libavcodec/aacenc.c

@@ -776,7 +776,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     for (i = 0; i < 428; i++)
         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
 
-    avctx->delay = 1024;
+    avctx->initial_padding = 1024;
     ff_af_queue_init(avctx, &s->afq);
 
     return 0;

libavcodec/ac3enc.c

@@ -2434,7 +2434,7 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
         return ret;
 
     avctx->frame_size = AC3_BLOCK_SIZE * s->num_blocks;
-    avctx->delay      = AC3_BLOCK_SIZE;
+    avctx->initial_padding = AC3_BLOCK_SIZE;
 
     s->bitstream_mode = avctx->audio_service_type;
     if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)

libavcodec/audio_frame_queue.c

@@ -28,8 +28,8 @@
 av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
 {
     afq->avctx = avctx;
-    afq->remaining_delay   = avctx->delay;
-    afq->remaining_samples = avctx->delay;
+    afq->remaining_delay   = avctx->initial_padding;
+    afq->remaining_samples = avctx->initial_padding;
     afq->frame_count       = 0;
 }
 

libavcodec/avcodec.h

@@ -1373,16 +1373,7 @@ typedef struct AVCodecContext {
      *   encoded input.
      *
      * Audio:
-     *   For encoding, this is the number of "priming" samples added by the
-     *   encoder to the beginning of the stream. The decoded output will be
-     *   delayed by this many samples relative to the input to the encoder (or
-     *   more, if the decoder adds its own padding).
-     *   The timestamps on the output packets are adjusted by the encoder so
-     *   that they always refer to the first sample of the data actually
-     *   contained in the packet, including any added padding.
-     *   E.g. if the timebase is 1/samplerate and the timestamp of the first
-     *   input sample is 0, the timestamp of the first output packet will be
-     *   -delay.
+     *   For encoding, this field is unused (see initial_padding).
      *
      *   For decoding, this is the number of samples the decoder needs to
      *   output before the decoder's output is valid. When seeking, you should
@@ -2980,6 +2971,23 @@ typedef struct AVCodecContext {
      */
     int side_data_only_packets;
 
+    /**
+     * Audio only. The number of "priming" samples (padding) inserted by the
+     * encoder at the beginning of the audio. I.e. this number of leading
+     * decoded samples must be discarded by the caller to get the original audio
+     * without leading padding.
+     *
+     * - decoding: unused
+     * - encoding: Set by libavcodec. The timestamps on the output packets are
+     *             adjusted by the encoder so that they always refer to the
+     *             first sample of the data actually contained in the packet,
+     *             including any added padding.  E.g. if the timebase is
+     *             1/samplerate and the timestamp of the first input sample is
+     *             0, the timestamp of the first output packet will be
+     *             -initial_padding.
+     */
+    int initial_padding;
+
     /**
      * Timebase in which pkt_dts/pts and AVPacket.dts/pts are.
      * Code outside libavcodec should access this field using:

libavcodec/g722enc.c

@@ -107,7 +107,7 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
            a common packet size for VoIP applications */
         avctx->frame_size = 320;
     }
-    avctx->delay = 22;
+    avctx->initial_padding = 22;
 
     if (avctx->trellis) {
         /* validate trellis */
@@ -374,7 +374,7 @@ static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
 
     if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
     *got_packet_ptr = 1;
     return 0;
 }

libavcodec/libfaac.c

@@ -167,7 +167,7 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
         }
     }
 
-    avctx->delay = FAAC_DELAY_SAMPLES;
+    avctx->initial_padding = FAAC_DELAY_SAMPLES;
     ff_af_queue_init(avctx, &s->afq);
 
     return 0;

libavcodec/libfdk-aacenc.c

@@ -286,7 +286,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     }
 
     avctx->frame_size = info.frameLength;
-    avctx->delay      = info.encoderDelay;
+    avctx->initial_padding = info.encoderDelay;
     ff_af_queue_init(avctx, &s->afq);
 
     if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {

libavcodec/libmp3lame.c

@@ -138,7 +138,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
     }
 
     /* get encoder delay */
-    avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
+    avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
     ff_af_queue_init(avctx, &s->afq);
 
     avctx->frame_size  = lame_get_framesize(s->gfp);

libavcodec/libopencore-amr.c

@@ -199,7 +199,7 @@ static av_cold int amr_nb_encode_init(AVCodecContext *avctx)
     }
 
     avctx->frame_size  = 160;
-    avctx->delay       =  50;
+    avctx->initial_padding = 50;
     ff_af_queue_init(avctx, &s->afq);
 
     s->enc_state = Encoder_Interface_init(s->enc_dtx);
@@ -246,7 +246,7 @@ static int amr_nb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                 return AVERROR(ENOMEM);
             memcpy(flush_buf, samples, frame->nb_samples * sizeof(*flush_buf));
             samples = flush_buf;
-            if (frame->nb_samples < avctx->frame_size - avctx->delay)
+            if (frame->nb_samples < avctx->frame_size - avctx->initial_padding)
                 s->enc_last_frame = -1;
         }
         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) {

libavcodec/libopusenc.c

@@ -87,7 +87,7 @@ static void libopus_write_header(AVCodecContext *avctx, int stream_count,
     bytestream_put_buffer(&p, "OpusHead", 8);
     bytestream_put_byte(&p, 1); /* Version */
     bytestream_put_byte(&p, channels);
-    bytestream_put_le16(&p, avctx->delay); /* Lookahead samples at 48kHz */
+    bytestream_put_le16(&p, avctx->initial_padding); /* Lookahead samples at 48kHz */
     bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
     bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
 
@@ -284,7 +284,7 @@ static av_cold int libopus_encode_init(AVCodecContext *avctx)
         goto fail;
     }
 
-    ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->delay));
+    ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->initial_padding));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to get number of lookahead samples: %s\n",

libavcodec/libspeexenc.c

@@ -236,7 +236,7 @@ static av_cold int encode_init(AVCodecContext *avctx)
     s->header.frames_per_packet = s->frames_per_packet;
 
     /* set encoding delay */
-    speex_encoder_ctl(s->enc_state, SPEEX_GET_LOOKAHEAD, &avctx->delay);
+    speex_encoder_ctl(s->enc_state, SPEEX_GET_LOOKAHEAD, &avctx->initial_padding);
     ff_af_queue_init(avctx, &s->afq);
 
     /* create header packet bytes from header struct */

libavcodec/libtwolame.c

@@ -60,7 +60,7 @@ static av_cold int twolame_encode_init(AVCodecContext *avctx)
     int ret;
 
     avctx->frame_size = TWOLAME_SAMPLES_PER_FRAME;
-    avctx->delay      = 512 - 32 + 1;
+    avctx->initial_padding = 512 - 32 + 1;
 
     s->glopts = twolame_init();
     if (!s->glopts)
@@ -155,7 +155,7 @@ static int twolame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     if (frame) {
         avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
         if (frame->pts != AV_NOPTS_VALUE)
-            avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+            avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
     } else {
         avpkt->pts = s->next_pts;
     }

libavcodec/libvo-aacenc.c

@@ -61,7 +61,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     int index, ret;
 
     avctx->frame_size = FRAME_SIZE;
-    avctx->delay      = ENC_DELAY;
+    avctx->initial_padding = ENC_DELAY;
     s->last_frame     = 2;
     ff_af_queue_init(avctx, &s->afq);
 

libavcodec/libvo-amrwbenc.c

@@ -93,7 +93,7 @@ static av_cold int amr_wb_encode_init(AVCodecContext *avctx)
     s->last_bitrate    = avctx->bit_rate;
 
     avctx->frame_size  = 320;
-    avctx->delay       =  80;
+    avctx->initial_padding =  80;
 
     s->state     = E_IF_init();
 
@@ -129,7 +129,7 @@ static int amr_wb_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     }
 
     if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
 
     avpkt->size = size;
     *got_packet_ptr = 1;

libavcodec/libvorbisenc.c

@@ -345,8 +345,8 @@ static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     if (duration > 0) {
         /* we do not know encoder delay until we get the first packet from
          * libvorbis, so we have to update the AudioFrameQueue counts */
-        if (!avctx->delay && s->afq.frames) {
-            avctx->delay              = duration;
+        if (!avctx->initial_padding && s->afq.frames) {
+            avctx->initial_padding    = duration;
             av_assert0(!s->afq.remaining_delay);
             s->afq.frames->duration  += duration;
             if (s->afq.frames->pts != AV_NOPTS_VALUE)

libavcodec/mpegaudioenc_template.c

@@ -89,7 +89,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
     bitrate = bitrate / 1000;
     s->nb_channels = channels;
     avctx->frame_size = MPA_FRAME_SIZE;
-    avctx->delay      = 512 - 32 + 1;
+    avctx->initial_padding = 512 - 32 + 1;
 
     /* encoding freq */
     s->lsf = 0;
@@ -771,7 +771,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     encode_frame(s, bit_alloc, padding);
 
     if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
 
     avpkt->size = put_bits_count(&s->pb) / 8;
     *got_packet_ptr = 1;

libavcodec/nellymoserenc.c

@@ -165,7 +165,7 @@ static av_cold int encode_init(AVCodecContext *avctx)
     }
 
     avctx->frame_size = NELLY_SAMPLES;
-    avctx->delay      = NELLY_BUF_LEN;
+    avctx->initial_padding = NELLY_BUF_LEN;
     ff_af_queue_init(avctx, &s->afq);
     s->avctx = avctx;
     if ((ret = ff_mdct_init(&s->mdct_ctx, 8, 0, 32768.0)) < 0)

libavcodec/ra144enc.c

@@ -55,7 +55,7 @@ static av_cold int ra144_encode_init(AVCodecContext * avctx)
         return -1;
     }
     avctx->frame_size = NBLOCKS * BLOCKSIZE;
-    avctx->delay      = avctx->frame_size;
+    avctx->initial_padding = avctx->frame_size;
     avctx->bit_rate = 8000;
     ractx = avctx->priv_data;
     ractx->lpc_coef[0] = ractx->lpc_tables[0];

libavcodec/utils.c

@@ -1609,6 +1609,11 @@ int attribute_align_arg avcodec_open2(AVCodecContext *avctx, const AVCodec *code
 
     ret=0;
 
+#if FF_API_AUDIOENC_DELAY
+    if (av_codec_is_encoder(avctx->codec))
+        avctx->delay = avctx->initial_padding;
+#endif
+
     if (av_codec_is_decoder(avctx->codec)) {
         if (!avctx->bit_rate)
             avctx->bit_rate = get_bit_rate(avctx);
@@ -1921,6 +1926,10 @@ end:
     av_frame_free(&padded_frame);
     av_free(extended_frame);
 
+#if FF_API_AUDIOENC_DELAY
+    avctx->delay = avctx->initial_padding;
+#endif
+
     return ret;
 }
 

libavcodec/version.h

@@ -29,8 +29,8 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 56
-#define LIBAVCODEC_VERSION_MINOR  4
-#define LIBAVCODEC_VERSION_MICRO 101
+#define LIBAVCODEC_VERSION_MINOR  5
+#define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \
@@ -175,5 +175,8 @@
 #ifndef FF_API_DV_FRAME_PROFILE
 #define FF_API_DV_FRAME_PROFILE  (LIBAVCODEC_VERSION_MAJOR < 57)
 #endif
+#ifndef FF_API_AUDIOENC_DELAY
+#define FF_API_AUDIOENC_DELAY    (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
 
 #endif /* AVCODEC_VERSION_H */

libavcodec/wmaenc.c

@@ -88,9 +88,7 @@ static av_cold int encode_init(AVCodecContext *avctx)
                          (avctx->sample_rate * 8);
     block_align        = FFMIN(block_align, MAX_CODED_SUPERFRAME_SIZE);
     avctx->block_align = block_align;
-
-    avctx->frame_size  =
-    avctx->delay       = s->frame_len;
+    avctx->frame_size = avctx->initial_padding = s->frame_len;
 
     return 0;
 }
@@ -403,7 +401,7 @@ static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
     av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == avctx->block_align);
 
     if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
 
     avpkt->size     = avctx->block_align;
     *got_packet_ptr = 1;