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 1be0a6c7eb29e96768b63f038fd0d1d00ff6b0a3..11c6e92928619348ad360a8ade61c9c013a47009 100644 (file)
--- a/libavdevice/alsa-audio-dec.c
+++ b/libavdevice/alsa-audio-dec.c
@@ -21,7 +21,7 @@
  */
 
 /**
- * @file alsa-audio-dec.c
+ * @file
  * ALSA input and output: input
  * @author Luca Abeni ( lucabe72 email it )
  * @author Benoit Fouet ( benoit fouet free fr )
@@ -45,32 +45,21 @@
  * plugin.
  */
 
-#include "libavformat/avformat.h"
 #include <alsa/asoundlib.h>
+#include "libavutil/opt.h"
+#include "libavutil/mathematics.h"
 
+#include "avdevice.h"
 #include "alsa-audio.h"
 
-av_cold static int audio_read_header(AVFormatContext *s1,
+static av_cold int audio_read_header(AVFormatContext *s1,
                                      AVFormatParameters *ap)
 {
     AlsaData *s = s1->priv_data;
     AVStream *st;
     int ret;
-    unsigned int sample_rate;
-    int codec_id;
-    snd_pcm_sw_params_t *sw_params;
-
-    if (ap->sample_rate <= 0) {
-        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
-
-        return AVERROR(EIO);
-    }
-
-    if (ap->channels <= 0) {
-        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
-
-        return AVERROR(EIO);
-    }
+    enum CodecID codec_id;
+    double o;
 
     st = av_new_stream(s1, 0);
     if (!st) {
@@ -78,44 +67,25 @@ av_cold static int audio_read_header(AVFormatContext *s1,
 
         return AVERROR(ENOMEM);
     }
-    sample_rate = ap->sample_rate;
-    codec_id    = ap->audio_codec_id;
+    codec_id    = s1->audio_codec_id;
 
-    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
         &codec_id);
     if (ret < 0) {
         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  = CODEC_TYPE_AUDIO;
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id    = codec_id;
-    st->codec->sample_rate = sample_rate;
-    st->codec->channels    = ap->channels;
+    st->codec->sample_rate = s->sample_rate;
+    st->codec->channels    = s->channels;
     av_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;
 
@@ -127,16 +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);
 
@@ -149,21 +118,33 @@ 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;
 
     return 0;
 }
 
-AVInputFormat alsa_demuxer = {
+static const AVOption options[] = {
+    { "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { "channels",    "", offsetof(AlsaData, channels),    FF_OPT_TYPE_INT, {.dbl = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { NULL },
+};
+
+static const AVClass alsa_demuxer_class = {
+    .class_name     = "ALSA demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+};
+
+AVInputFormat ff_alsa_demuxer = {
     "alsa",
     NULL_IF_CONFIG_SMALL("ALSA audio input"),
     sizeof(AlsaData),
@@ -172,4 +153,5 @@ AVInputFormat alsa_demuxer = {
     audio_read_packet,
     ff_alsa_close,
     .flags = AVFMT_NOFILE,
+    .priv_class = &alsa_demuxer_class,
 };