[FFmpeg-cvslog] avdevice/pulse_audio_dec: reimplement using the non simple API
Michael Niedermayer
git at videolan.org
Wed Jul 9 01:44:50 CEST 2014
ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Tue Jul 8 04:28:33 2014 +0200| [98eab9815961289b913ebd1b4230a661d190bc69] | committer: Michael Niedermayer
avdevice/pulse_audio_dec: reimplement using the non simple API
This fixes timestamps
Based-on: code from pulseaudio
Reviewed-by: Lukasz Marek <lukasz.m.luki2 at gmail.com>
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=98eab9815961289b913ebd1b4230a661d190bc69
---
libavdevice/pulse_audio_dec.c | 287 +++++++++++++++++++++++++++++++++--------
libavdevice/version.h | 2 +-
2 files changed, 235 insertions(+), 54 deletions(-)
diff --git a/libavdevice/pulse_audio_dec.c b/libavdevice/pulse_audio_dec.c
index 414d73b..01ff6d1 100644
--- a/libavdevice/pulse_audio_dec.c
+++ b/libavdevice/pulse_audio_dec.c
@@ -1,6 +1,8 @@
/*
* Pulseaudio input
* Copyright (c) 2011 Luca Barbato <lu_zero at gentoo.org>
+ * Copyright 2004-2006 Lennart Poettering
+ * Copyright (c) 2014 Michael Niedermayer <michaelni at gmx.at>
*
* This file is part of FFmpeg.
*
@@ -19,19 +21,14 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file
- * PulseAudio input using the simple API.
- * @author Luca Barbato <lu_zero at gentoo.org>
- */
-
-#include <pulse/simple.h>
#include <pulse/rtclock.h>
#include <pulse/error.h>
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "libavutil/opt.h"
+#include "libavutil/time.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)
@@ -44,17 +41,102 @@ typedef struct PulseData {
int channels;
int frame_size;
int fragment_size;
- pa_simple *s;
- int64_t pts;
- int64_t frame_duration;
+
+ pa_threaded_mainloop *mainloop;
+ pa_context *context;
+ pa_stream *stream;
+
+ TimeFilter *timefilter;
+ int last_period;
} 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, sample_bytes;
+ 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),
@@ -75,77 +157,176 @@ static av_cold int pulse_read_header(AVFormatContext *s)
if (strcmp(s->filename, "default"))
device = s->filename;
- pd->s = pa_simple_new(pd->server, pd->name,
- PA_STREAM_RECORD,
- device, pd->stream_name, &ss,
- NULL, &attr, &ret);
+ 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 (!pd->s) {
- av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
- pa_strerror(ret));
- return AVERROR(EIO);
+ 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, pd->sample_rate); /* 64 bits pts in us */
+ avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
- pd->pts = AV_NOPTS_VALUE;
- sample_bytes = (av_get_bits_per_sample(codec_id) >> 3) * pd->channels;
+ pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
+ 1000, 1.5E-6);
- if (pd->frame_size % sample_bytes) {
- av_log(s, AV_LOG_WARNING, "frame_size %i is not divisible by %i "
- "(channels * bytes_per_sample) \n", pd->frame_size, sample_bytes);
+ if (!pd->timefilter) {
+ pulse_close(s);
+ return AVERROR(ENOMEM);
}
- pd->frame_duration = pd->frame_size / sample_bytes;
-
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 res;
+ int ret;
+ size_t read_length;
+ const void *read_data = NULL;
+ int64_t dts;
+ pa_usec_t latency;
+ int negative;
- if (av_new_packet(pkt, pd->frame_size) < 0) {
- return AVERROR(ENOMEM);
- }
+ pa_threaded_mainloop_lock(pd->mainloop);
- if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
- av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
- pa_strerror(res));
- av_free_packet(pkt);
- return AVERROR(EIO);
- }
+ CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
- if (pd->pts == AV_NOPTS_VALUE) {
- pa_usec_t latency;
+ while (!read_data) {
+ int r;
- if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
- av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
- pa_strerror(res));
- return AVERROR(EIO);
+ 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);
}
+ }
- pd->pts = -latency;
+ if (av_new_packet(pkt, read_length) < 0) {
+ ret = AVERROR(ENOMEM);
+ goto unlock_and_fail;
}
- pkt->pts = pd->pts;
+ dts = av_gettime();
+ pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
- pd->pts += pd->frame_duration;
+ 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;
- return 0;
-}
-static av_cold int pulse_close(AVFormatContext *s)
-{
- PulseData *pd = s->priv_data;
- pa_simple_free(pd->s);
- pd->s = NULL;
+ if (negative) {
+ dts += latency;
+ } else
+ dts -= latency;
+ 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)
diff --git a/libavdevice/version.h b/libavdevice/version.h
index 163a4c6..a098549 100644
--- a/libavdevice/version.h
+++ b/libavdevice/version.h
@@ -29,7 +29,7 @@
#define LIBAVDEVICE_VERSION_MAJOR 55
#define LIBAVDEVICE_VERSION_MINOR 13
-#define LIBAVDEVICE_VERSION_MICRO 101
+#define LIBAVDEVICE_VERSION_MICRO 102
#define LIBAVDEVICE_VERSION_INT AV_VERSION_INT(LIBAVDEVICE_VERSION_MAJOR, \
LIBAVDEVICE_VERSION_MINOR, \
More information about the ffmpeg-cvslog
mailing list