Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mov: Don't av_malloc(0).
  avconv: only allocate 1 AVFrame per input stream
  avconv: fix memleaks due to not freeing the AVFrame for audio
  h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg).
  misc Doxygen markup improvements
  doxygen: eliminate Qt-style doxygen syntax
  g722: Add a regression test for muxing/demuxing in wav
  g722: Change bits per sample to 4
  g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample
  api-example: update to use avcodec_decode_audio4()
  avplay: use avcodec_decode_audio4()
  avplay: use a separate buffer for playing silence
  avformat: use avcodec_decode_audio4() in avformat_find_stream_info()
  avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3()
  mov: Allow empty stts atom.
  doc: document preferred Doxygen syntax and make patcheck detect it

Conflicts:
	avconv.c
	ffplay.c
	libavcodec/mlpdec.c
	libavcodec/version.h
	libavformat/mov.c
	tests/codec-regression.sh
	tests/fate/h264.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 25 insertions, 21 deletions

diff --git a/ffplay.c b/ffplay.c
index c1fe937ec4..976ac06d7b 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -151,11 +151,10 @@ typedef struct VideoState {
     AVStream *audio_st;
     PacketQueue audioq;
     int audio_hw_buf_size;
-    /* samples output by the codec. we reserve more space for avsync
-       compensation, resampling and format conversion */
-    DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
     DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
+    uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
     uint8_t *audio_buf;
+    uint8_t *audio_buf1;
     unsigned int audio_buf_size; /* in bytes */
     int audio_buf_index; /* in bytes */
     int audio_write_buf_size;
@@ -174,6 +173,7 @@ typedef struct VideoState {
     double audio_current_pts_drift;
     int frame_drops_early;
     int frame_drops_late;
+    AVFrame *frame;
 
     enum ShowMode {
         SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -1998,8 +1998,8 @@ static int synchronize_audio(VideoState *is, short *samples,
                     max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
                     if (wanted_size < min_size)
                         wanted_size = min_size;
-                    else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)))
-                        wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2));
+                    else if (wanted_size > FFMIN3(max_size, samples_size, sizeof(is->audio_buf2)))
+                        wanted_size = FFMIN3(max_size, samples_size, sizeof(is->audio_buf2));
 
                     /* add or remove samples to correction the synchro */
                     if (wanted_size < samples_size) {
@@ -2043,7 +2043,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     AVPacket *pkt = &is->audio_pkt;
     AVCodecContext *dec= is->audio_st->codec;
     int len1, len2, data_size, resampled_data_size;
-    int64_t dec_channel_layout;
+    int64_t dec_channel_layout, got_frame;
     double pts;
     int new_packet = 0;
     int flush_complete = 0;
@@ -2051,13 +2051,16 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     for(;;) {
         /* NOTE: the audio packet can contain several frames */
         while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
+            if (!is->frame) {
+                if (!(is->frame = avcodec_alloc_frame()))
+                    return AVERROR(ENOMEM);
+            } else
+                avcodec_get_frame_defaults(is->frame);
+
             if (flush_complete)
                 break;
             new_packet = 0;
-            data_size = sizeof(is->audio_buf1);
-            len1 = avcodec_decode_audio3(dec,
-                                        (int16_t *)is->audio_buf1, &data_size,
-                                        pkt_temp);
+            len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
             if (len1 < 0) {
                 /* if error, we skip the frame */
                 pkt_temp->size = 0;
@@ -2067,12 +2070,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             pkt_temp->data += len1;
             pkt_temp->size -= len1;
 
-            if (data_size <= 0) {
+            if (!got_frame) {
                 /* stop sending empty packets if the decoder is finished */
                 if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
                     flush_complete = 1;
                 continue;
             }
+            data_size = av_samples_get_buffer_size(NULL, dec->channels,
+                                                   is->frame->nb_samples,
+                                                   dec->sample_fmt, 1);
 
             dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
 
@@ -2101,7 +2107,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
 
             resampled_data_size = data_size;
             if (is->swr_ctx) {
-                const uint8_t *in[] = {is->audio_buf1};
+                const uint8_t *in[] = { is->frame->data[0] };
                 uint8_t *out[] = {is->audio_buf2};
                 len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
                                                 in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
@@ -2116,7 +2122,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                 is->audio_buf = is->audio_buf2;
                 resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
             } else {
-                is->audio_buf= is->audio_buf1;
+                is->audio_buf = is->frame->data[0];
             }
 
             /* if no pts, then compute it */
@@ -2150,11 +2156,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
         if (pkt->data == flush_pkt.data)
             avcodec_flush_buffers(dec);
 
-        pkt_temp->data = pkt->data;
-        pkt_temp->size = pkt->size;
-        pkt_temp->flags           = pkt->flags;
-        pkt_temp->side_data       = pkt->side_data;
-        pkt_temp->side_data_elems = pkt->side_data_elems;
+        *pkt_temp = *pkt;
 
         /* if update the audio clock with the pts */
         if (pkt->pts != AV_NOPTS_VALUE) {
@@ -2178,9 +2180,8 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
            audio_size = audio_decode_frame(is, &pts);
            if (audio_size < 0) {
                 /* if error, just output silence */
-               is->audio_buf = is->audio_buf1;
-               is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
-               memset(is->audio_buf, 0, is->audio_buf_size);
+               is->audio_buf      = is->silence_buf;
+               is->audio_buf_size = sizeof(is->silence_buf);
            } else {
                if (is->show_mode != SHOW_MODE_VIDEO)
                    update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2356,6 +2357,9 @@ static void stream_component_close(VideoState *is, int stream_index)
         if (is->swr_ctx)
             swr_free(&is->swr_ctx);
         av_free_packet(&is->audio_pkt);
+        av_freep(&is->audio_buf1);
+        is->audio_buf = NULL;
+        av_freep(&is->frame);
 
         if (is->rdft) {
             av_rdft_end(is->rdft);