diff options
Diffstat (limited to 'libavdevice/pulse_audio_dec.c')
| -rw-r--r-- | libavdevice/pulse_audio_dec.c | 376 |
1 files changed, 376 insertions, 0 deletions
diff --git a/libavdevice/pulse_audio_dec.c b/libavdevice/pulse_audio_dec.c new file mode 100644 index 0000000000..aa0800b40b --- /dev/null +++ b/libavdevice/pulse_audio_dec.c @@ -0,0 +1,376 @@ +/* + * Pulseaudio input + * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org> + * Copyright 2004-2006 Lennart Poettering + * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <pulse/rtclock.h> +#include <pulse/error.h> + +#include "libavutil/internal.h" +#include "libavutil/opt.h" +#include "libavutil/time.h" + +#include "libavformat/avformat.h" +#include "libavformat/internal.h" +#include "pulse_audio_common.h" +#include "timefilter.h" + +#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) + +typedef struct PulseData { + AVClass *class; + char *server; + char *name; + char *stream_name; + int sample_rate; + int channels; + int frame_size; + int fragment_size; + + pa_threaded_mainloop *mainloop; + pa_context *context; + pa_stream *stream; + + TimeFilter *timefilter; + int last_period; + int wallclock; +} PulseData; + + +#define CHECK_SUCCESS_GOTO(rerror, expression, label) \ + do { \ + if (!(expression)) { \ + rerror = AVERROR_EXTERNAL; \ + goto label; \ + } \ + } while (0) + +#define CHECK_DEAD_GOTO(p, rerror, label) \ + do { \ + if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \ + !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \ + rerror = AVERROR_EXTERNAL; \ + goto label; \ + } \ + } while (0) + +static void context_state_cb(pa_context *c, void *userdata) { + PulseData *p = userdata; + + switch (pa_context_get_state(c)) { + case PA_CONTEXT_READY: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + pa_threaded_mainloop_signal(p->mainloop, 0); + break; + } +} + +static void stream_state_cb(pa_stream *s, void * userdata) { + PulseData *p = userdata; + + switch (pa_stream_get_state(s)) { + case PA_STREAM_READY: + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + pa_threaded_mainloop_signal(p->mainloop, 0); + break; + } +} + +static void stream_request_cb(pa_stream *s, size_t length, void *userdata) { + PulseData *p = userdata; + + pa_threaded_mainloop_signal(p->mainloop, 0); +} + +static void stream_latency_update_cb(pa_stream *s, void *userdata) { + PulseData *p = userdata; + + pa_threaded_mainloop_signal(p->mainloop, 0); +} + +static av_cold int pulse_close(AVFormatContext *s) +{ + PulseData *pd = s->priv_data; + + if (pd->mainloop) + pa_threaded_mainloop_stop(pd->mainloop); + + if (pd->stream) + pa_stream_unref(pd->stream); + pd->stream = NULL; + + if (pd->context) { + pa_context_disconnect(pd->context); + pa_context_unref(pd->context); + } + pd->context = NULL; + + if (pd->mainloop) + pa_threaded_mainloop_free(pd->mainloop); + pd->mainloop = NULL; + + ff_timefilter_destroy(pd->timefilter); + pd->timefilter = NULL; + + return 0; +} + +static av_cold int pulse_read_header(AVFormatContext *s) +{ + PulseData *pd = s->priv_data; + AVStream *st; + char *device = NULL; + int ret; + enum AVCodecID codec_id = + s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; + const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id), + pd->sample_rate, + pd->channels }; + + pa_buffer_attr attr = { -1 }; + + st = avformat_new_stream(s, NULL); + + if (!st) { + av_log(s, AV_LOG_ERROR, "Cannot add stream\n"); + return AVERROR(ENOMEM); + } + + attr.fragsize = pd->fragment_size; + + if (s->filename[0] != '\0' && strcmp(s->filename, "default")) + device = s->filename; + + if (!(pd->mainloop = pa_threaded_mainloop_new())) { + pulse_close(s); + return AVERROR_EXTERNAL; + } + + if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) { + pulse_close(s); + return AVERROR_EXTERNAL; + } + + pa_context_set_state_callback(pd->context, context_state_cb, pd); + + if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) { + pulse_close(s); + return AVERROR(pa_context_errno(pd->context)); + } + + pa_threaded_mainloop_lock(pd->mainloop); + + if (pa_threaded_mainloop_start(pd->mainloop) < 0) { + ret = -1; + goto unlock_and_fail; + } + + for (;;) { + pa_context_state_t state; + + state = pa_context_get_state(pd->context); + + if (state == PA_CONTEXT_READY) + break; + + if (!PA_CONTEXT_IS_GOOD(state)) { + ret = AVERROR(pa_context_errno(pd->context)); + goto unlock_and_fail; + } + + /* Wait until the context is ready */ + pa_threaded_mainloop_wait(pd->mainloop); + } + + if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) { + ret = AVERROR(pa_context_errno(pd->context)); + goto unlock_and_fail; + } + + pa_stream_set_state_callback(pd->stream, stream_state_cb, pd); + pa_stream_set_read_callback(pd->stream, stream_request_cb, pd); + pa_stream_set_write_callback(pd->stream, stream_request_cb, pd); + pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd); + + ret = pa_stream_connect_record(pd->stream, device, &attr, + PA_STREAM_INTERPOLATE_TIMING + |PA_STREAM_ADJUST_LATENCY + |PA_STREAM_AUTO_TIMING_UPDATE); + + if (ret < 0) { + ret = AVERROR(pa_context_errno(pd->context)); + goto unlock_and_fail; + } + + for (;;) { + pa_stream_state_t state; + + state = pa_stream_get_state(pd->stream); + + if (state == PA_STREAM_READY) + break; + + if (!PA_STREAM_IS_GOOD(state)) { + ret = AVERROR(pa_context_errno(pd->context)); + goto unlock_and_fail; + } + + /* Wait until the stream is ready */ + pa_threaded_mainloop_wait(pd->mainloop); + } + + pa_threaded_mainloop_unlock(pd->mainloop); + + /* take real parameters */ + st->codec->codec_type = AVMEDIA_TYPE_AUDIO; + st->codec->codec_id = codec_id; + st->codec->sample_rate = pd->sample_rate; + st->codec->channels = pd->channels; + avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ + + pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate, + 1000, 1.5E-6); + + if (!pd->timefilter) { + pulse_close(s); + return AVERROR(ENOMEM); + } + + return 0; + +unlock_and_fail: + pa_threaded_mainloop_unlock(pd->mainloop); + + pulse_close(s); + return ret; +} + +static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) +{ + PulseData *pd = s->priv_data; + int ret; + size_t read_length; + const void *read_data = NULL; + int64_t dts; + pa_usec_t latency; + int negative; + + pa_threaded_mainloop_lock(pd->mainloop); + + CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); + + while (!read_data) { + int r; + + r = pa_stream_peek(pd->stream, &read_data, &read_length); + CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); + + if (read_length <= 0) { + pa_threaded_mainloop_wait(pd->mainloop); + CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); + } else if (!read_data) { + /* There's a hole in the stream, skip it. We could generate + * silence, but that wouldn't work for compressed streams. */ + r = pa_stream_drop(pd->stream); + CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); + } + } + + if (av_new_packet(pkt, read_length) < 0) { + ret = AVERROR(ENOMEM); + goto unlock_and_fail; + } + + dts = av_gettime(); + pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL)); + + if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) { + enum AVCodecID codec_id = + s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; + int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels); + int frame_duration = read_length / frame_size; + + + if (negative) { + dts += latency; + } else + dts -= latency; + if (pd->wallclock) + pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period); + + pd->last_period = frame_duration; + } else { + av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n"); + } + + memcpy(pkt->data, read_data, read_length); + pa_stream_drop(pd->stream); + + pa_threaded_mainloop_unlock(pd->mainloop); + return 0; + +unlock_and_fail: + pa_threaded_mainloop_unlock(pd->mainloop); + return ret; +} + +static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) +{ + PulseData *s = h->priv_data; + return ff_pulse_audio_get_devices(device_list, s->server, 0); +} + +#define OFFSET(a) offsetof(PulseData, a) +#define D AV_OPT_FLAG_DECODING_PARAM + +static const AVOption options[] = { + { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D }, + { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D }, + { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D }, + { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D }, + { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D }, + { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D }, + { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D }, + { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D }, + { NULL }, +}; + +static const AVClass pulse_demuxer_class = { + .class_name = "Pulse demuxer", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, + .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, +}; + +AVInputFormat ff_pulse_demuxer = { + .name = "pulse", + .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"), + .priv_data_size = sizeof(PulseData), + .read_header = pulse_read_header, + .read_packet = pulse_read_packet, + .read_close = pulse_close, + .get_device_list = pulse_get_device_list, + .flags = AVFMT_NOFILE, + .priv_class = &pulse_demuxer_class, +}; |