Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavdevice / alsa-audio-dec.c

diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
index b6a7ac20feeebb66f14e3b7865b097f7061290d8..f2dd21655b0c83500ab80c853ae5f8735ed595c6 100644 (file)
--- a/libavdevice/alsa-audio-dec.c
+++ b/libavdevice/alsa-audio-dec.c
@@ -3,20 +3,20 @@
  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -46,10 +46,11 @@
  */
 
 #include <alsa/asoundlib.h>
-#include "libavformat/avformat.h"
 #include "libavformat/internal.h"
 #include "libavutil/opt.h"
+#include "libavutil/mathematics.h"
 
+#include "avdevice.h"
 #include "alsa-audio.h"
 
 static av_cold int audio_read_header(AVFormatContext *s1)
@@ -58,7 +59,7 @@ static av_cold int audio_read_header(AVFormatContext *s1)
     AVStream *st;
     int ret;
     enum CodecID codec_id;
-    snd_pcm_sw_params_t *sw_params;
+    double o;
 
     st = avformat_new_stream(s1, NULL);
     if (!st) {
@@ -74,35 +75,17 @@ static av_cold int audio_read_header(AVFormatContext *s1)
         return AVERROR(EIO);
     }
 
-    if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
-        av_log(s1, AV_LOG_WARNING,
-               "capture with some ALSA plugins, especially dsnoop, "
-               "may hang.\n");
-
-    ret = snd_pcm_sw_params_malloc(&sw_params);
-    if (ret < 0) {
-        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
-               snd_strerror(ret));
-        goto fail;
-    }
-
-    snd_pcm_sw_params_current(s->h, sw_params);
-    snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
-
-    ret = snd_pcm_sw_params(s->h, sw_params);
-    snd_pcm_sw_params_free(sw_params);
-    if (ret < 0) {
-        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
-               snd_strerror(ret));
-        goto fail;
-    }
-
     /* take real parameters */
     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id    = codec_id;
     st->codec->sample_rate = s->sample_rate;
     st->codec->channels    = s->channels;
     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+    o = 2 * M_PI * s->period_size / s->sample_rate * 1.5; // bandwidth: 1.5Hz
+    s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
+                                      sqrt(2 * o), o * o);
+    if (!s->timefilter)
+        goto fail;
 
     return 0;
 
@@ -114,16 +97,15 @@ fail:
 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AlsaData *s  = s1->priv_data;
-    AVStream *st = s1->streams[0];
     int res;
-    snd_htimestamp_t timestamp;
-    snd_pcm_uframes_t ts_delay;
+    int64_t dts;
+    snd_pcm_sframes_t delay = 0;
 
-    if (av_new_packet(pkt, s->period_size) < 0) {
+    if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
         return AVERROR(EIO);
     }
 
-    while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
+    while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
         if (res == -EAGAIN) {
             av_free_packet(pkt);
 
@@ -136,14 +118,13 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 
             return AVERROR(EIO);
         }
+        ff_timefilter_reset(s->timefilter);
     }
 
-    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
-    ts_delay += res;
-    pkt->pts = timestamp.tv_sec * 1000000LL
-               + (timestamp.tv_nsec * st->codec->sample_rate
-                  - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
-               / (st->codec->sample_rate * 1000LL);
+    dts = av_gettime();
+    snd_pcm_delay(s->h, &delay);
+    dts -= av_rescale(delay + res, 1000000, s->sample_rate);
+    pkt->pts = ff_timefilter_update(s->timefilter, dts, res);
 
     pkt->size = res * s->frame_size;