diff --git a/libavdevice/alsa-audio-common.c b/libavdevice/alsa-audio-common.c index 8c5be3c864..6188721f48 100644 --- a/libavdevice/alsa-audio-common.c +++ b/libavdevice/alsa-audio-common.c @@ -316,6 +316,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1) AlsaData *s = s1->priv_data; av_freep(&s->reorder_buf); + ff_timefilter_destroy(s->timefilter); snd_pcm_close(s->h); return 0; } diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c index e3ad98b7f3..f8977a10f9 100644 --- a/libavdevice/alsa-audio-dec.c +++ b/libavdevice/alsa-audio-dec.c @@ -59,6 +59,7 @@ static av_cold int audio_read_header(AVFormatContext *s1, int ret; enum CodecID codec_id; snd_pcm_sw_params_t *sw_params; + double o; #if FF_API_FORMAT_PARAMETERS if (ap->sample_rate > 0) @@ -82,35 +83,17 @@ 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; 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; @@ -124,8 +107,8 @@ 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 * s->frame_size) < 0) { return AVERROR(EIO); @@ -144,14 +127,13 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) return AVERROR(EIO); } + ff_timefilter_reset(s->timefilter); } - snd_pcm_htimestamp(s->h, &ts_delay, ×tamp); - 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; diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h index 9b1ecb1696..0226632479 100644 --- a/libavdevice/alsa-audio.h +++ b/libavdevice/alsa-audio.h @@ -33,6 +33,7 @@ #include #include "config.h" #include "libavutil/log.h" +#include "libavformat/timefilter.h" #include "avdevice.h" /* XXX: we make the assumption that the soundcard accepts this format */ @@ -49,6 +50,7 @@ typedef struct { int period_size; ///< preferred size for reads and writes, in frames int sample_rate; ///< sample rate set by user int channels; ///< number of channels set by user + TimeFilter *timefilter; void (*reorder_func)(const void *, void *, int); void *reorder_buf; int reorder_buf_size; ///< in frames diff --git a/libavformat/Makefile b/libavformat/Makefile index 3d9017b3e4..6d7a342c9e 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -338,6 +338,7 @@ OBJS-$(CONFIG_TCP_PROTOCOL) += tcp.o OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o # libavdevice dependencies +OBJS-$(CONFIG_ALSA_INDEV) += timefilter.o OBJS-$(CONFIG_JACK_INDEV) += timefilter.o TESTPROGS = timefilter