Merge remote-tracking branch 'qatar/master'
* qatar/master: lavc: remove "legacy" mpegvideo decoder. iv8: assemble packets to return complete frames pulse: documentation pulse: introduce pulseaudio input remove the zork pcm seek test Conflicts: configure libavdevice/Makefile libavdevice/alldevices.c libavdevice/avdevice.h libavdevice/pulse.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
commit
8d4e44993a
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@ -390,6 +390,89 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
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For more information about OSS see:
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@url{http://manuals.opensound.com/usersguide/dsp.html}
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@section pulse
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pulseaudio input device.
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To enable this input device during configuration you need libpulse-simple
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installed in your system.
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The filename to provide to the input device is a source device or the
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string "default"
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To list the pulse source devices and their properties you can invoke
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the command @file{pactl list sources}.
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@example
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avconv -f pulse -i default /tmp/pulse.wav
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@end example
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@subsection @var{server} AVOption
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The syntax is:
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@example
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-server @var{server name}
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@end example
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Connects to a specific server.
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@subsection @var{name} AVOption
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The syntax is:
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@example
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-name @var{application name}
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@end example
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Specify the application name pulse will use when showing active clients,
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by default it is "libav"
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@subsection @var{stream_name} AVOption
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The syntax is:
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@example
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-stream_name @var{stream name}
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@end example
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Specify the stream name pulse will use when showing active streams,
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by default it is "record"
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@subsection @var{sample_rate} AVOption
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The syntax is:
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@example
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-sample_rate @var{samplerate}
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@end example
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Specify the samplerate in Hz, by default 48kHz is used.
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@subsection @var{channels} AVOption
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The syntax is:
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@example
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-channels @var{N}
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@end example
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Specify the channels in use, by default 2 (stereo) is set.
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@subsection @var{frame_size} AVOption
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The syntax is:
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@example
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-frame_size @var{bytes}
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@end example
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Specify the number of byte per frame, by default it is set to 1024.
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@subsection @var{fragment_size} AVOption
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The syntax is:
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@example
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-fragment_size @var{bytes}
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@end example
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Specify the minimal buffering fragment in pulseaudio, it will affect the
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audio latency. By default it is unset.
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@section sndio
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sndio input device.
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@ -21,11 +21,9 @@
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/**
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* @file
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* Pulseaudio input
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* PulseAudio input using the simple API.
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* @author Luca Barbato <lu_zero@gentoo.org>
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*
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* This avdevice decoder allows to capture audio from a Pulseaudio device using
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* the simple api.
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*/
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#include <pulse/simple.h>
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@ -95,9 +93,9 @@ static av_cold int pulse_read_header(AVFormatContext *s,
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device = s->filename;
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pd->s = pa_simple_new(pd->server, pd->name,
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PA_STREAM_RECORD,
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device, pd->stream_name, &ss,
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NULL, &attr, &ret);
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PA_STREAM_RECORD,
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device, pd->stream_name, &ss,
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NULL, &attr, &ret);
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if (!pd->s) {
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av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
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@ -122,7 +120,7 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
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int res;
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pa_usec_t latency;
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uint64_t frame_duration =
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(pd->frame_size*1000000LL)/(pd->sample_rate * pd->channels);
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(pd->frame_size*1000000LL) / (pd->sample_rate * pd->channels);
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if (av_new_packet(pkt, pd->frame_size) < 0) {
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return AVERROR(ENOMEM);
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@ -145,10 +143,10 @@ static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
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pd->pts = -latency;
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}
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pd->pts += frame_duration;
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pkt->pts = pd->pts;
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pd->pts += frame_duration;
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return 0;
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}
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@ -163,20 +161,13 @@ static av_cold int pulse_close(AVFormatContext *s)
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#define D AV_OPT_FLAG_DECODING_PARAM
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static const AVOption options[] = {
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{ "server", "pulse server name",
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OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
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{ "name", "application name",
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OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
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{ "stream_name", "stream description",
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OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
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{ "sample_rate", "",
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OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
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{ "channels", "",
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OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
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{ "frame_size", "",
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OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
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{ "fragment_size", "buffering size, affects latency and cpu usage",
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OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D },
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{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
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{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
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{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
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{ "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
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{ "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
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{ "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
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{ "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D },
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{ NULL },
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};
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@ -55,33 +55,56 @@ static int read_header(AVFormatContext *s, AVFormatParameters *ap)
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static int read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, size, pts, type;
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retry:
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type= avio_rb16(s->pb); // 257 or 258
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size= avio_rb16(s->pb);
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int ret, size, pts, type, flags;
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int first_pkt = 0;
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int frame_complete = 0;
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avio_rb16(s->pb); //some flags, 0x80 indicates end of frame
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avio_rb16(s->pb); //packet number
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pts=avio_rb32(s->pb);
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avio_rb32(s->pb); //6A 13 E3 88
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while (!frame_complete) {
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size -= 12;
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if(size<1)
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return -1;
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type = avio_rb16(s->pb); // 257 or 258
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size = avio_rb16(s->pb);
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flags = avio_rb16(s->pb); //some flags, 0x80 indicates end of frame
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avio_rb16(s->pb); //packet number
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pts = avio_rb32(s->pb);
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avio_rb32(s->pb); //6A 13 E3 88
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if(type==258){
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avio_skip(s->pb, size);
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goto retry;
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frame_complete = flags & 0x80;
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size -= 12;
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if (size < 1)
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return -1;
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if (type == 258) {
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avio_skip(s->pb, size);
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frame_complete = 0;
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continue;
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}
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if (!first_pkt) {
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ret = av_get_packet(s->pb, pkt, size);
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if (ret < 0)
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return ret;
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first_pkt = 1;
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pkt->pts = pts;
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pkt->pos -= 16;
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} else {
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ret = av_append_packet(s->pb, pkt, size);
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if (ret < 0) {
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av_log(s, AV_LOG_ERROR, "failed to grow packet\n");
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av_free_packet(pkt);
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return ret;
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}
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}
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if (ret < size) {
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av_log(s, AV_LOG_ERROR, "Truncated packet! Read %d of %d bytes\n",
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ret, size);
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pkt->flags |= AV_PKT_FLAG_CORRUPT;
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break;
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}
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}
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ret= av_get_packet(s->pb, pkt, size);
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pkt->pts= pts;
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pkt->pos-=16;
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pkt->stream_index = 0;
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return ret;
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return 0;
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}
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AVInputFormat ff_iv8_demuxer = {
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