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gstreamer/sys/audioflingersink/gstaudioflingersink.c
Benjamin Gaignard b4ff7c94d7 sys: Add android audioflingersink
2010-12-03 17:46:27 +01:00

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/* GStreamer
* Copyright (C) <2009> Prajnashi S <prajnashi@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-audioflindersink
*
* This element lets you output sound using the Audio Flinger system in Android
*
* Note that you should almost always use generic audio conversion elements
* like audioconvert and audioresample in front of an audiosink to make sure
* your pipeline works under all circumstances (those conversion elements will
* act in passthrough-mode if no conversion is necessary).
*/
#ifdef HAVE_CONFIG_H
//#include "config.h"
#endif
#include "gstaudioflingersink.h"
#include <utils/Log.h>
#define LOG_NDEBUG 0
#undef LOG_TAG
#define LOG_TAG "GstAudioFlingerSink"
#define DEFAULT_BUFFERTIME (500*GST_MSECOND) / (GST_USECOND)
#define DEFAULT_LATENCYTIME (50*GST_MSECOND) / (GST_USECOND)
#define DEFAULT_VOLUME 10.0
#define DEFAULT_MUTE FALSE
#define DEFAULT_EXPORT_SYSTEM_AUDIO_CLOCK TRUE
/*
* PROPERTY_ID
*/
enum
{
PROP_NULL,
PROP_VOLUME,
PROP_MUTE,
PROP_AUDIO_SINK,
};
GST_DEBUG_CATEGORY_STATIC (audioflinger_debug);
#define GST_CAT_DEFAULT audioflinger_debug
/* elementfactory information */
static const GstElementDetails gst_audioflinger_sink_details =
GST_ELEMENT_DETAILS ("Audio Sink (AudioFlinger)",
"Sink/Audio",
"Output to android's AudioFlinger",
"Prajnashi S <prajnashi@gmail.com>, "
"Alessandro Decina <alessandro.decina@collabora.co.uk>");
#define GST_TYPE_ANDROID_AUDIORING_BUFFER \
(gst_android_audioringbuffer_get_type())
#define GST_ANDROID_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_ANDROID_AUDIORING_BUFFER,GstAndroidAudioRingBuffer))
#define GST_ANDROID_AUDIORING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_ANDROID_AUDIORING_BUFFER,GstAndroidAudioRingBufferClass))
#define GST_ANDROID_AUDIORING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_ANDROID_AUDIORING_BUFFER, GstAndroidAudioRingBufferClass))
#define GST_ANDROID_AUDIORING_BUFFER_CAST(obj) \
((GstAndroidAudioRingBuffer *)obj)
#define GST_IS_ANDROID_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_ANDROID_AUDIORING_BUFFER))
#define GST_IS_ANDROID_AUDIORING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_ANDROID_AUDIORING_BUFFER))
typedef struct _GstAndroidAudioRingBuffer GstAndroidAudioRingBuffer;
typedef struct _GstAndroidAudioRingBufferClass GstAndroidAudioRingBufferClass;
#define GST_ANDROID_AUDIORING_BUFFER_GET_COND(buf) (((GstAndroidAudioRingBuffer *)buf)->cond)
#define GST_ANDROID_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_ANDROID_ANDROID_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_ANDROID_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_ANDROID_ANDROID_AUDIORING_BUFFER_GET_COND (buf)))
#define GST_ANDROID_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_ANDROID_ANDROID_AUDIORING_BUFFER_GET_COND (buf)))
struct _GstAndroidAudioRingBuffer
{
GstRingBuffer object;
gboolean running;
gint queuedseg;
GCond *cond;
};
struct _GstAndroidAudioRingBufferClass
{
GstRingBufferClass parent_class;
};
static void
gst_android_audioringbuffer_class_init (GstAndroidAudioRingBufferClass * klass);
static void gst_android_audioringbuffer_init (GstAndroidAudioRingBuffer *
ringbuffer, GstAndroidAudioRingBufferClass * klass);
static void gst_android_audioringbuffer_dispose (GObject * object);
static void gst_android_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_android_audioringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_close_device (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_android_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_pause (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_stop (GstRingBuffer * buf);
static gboolean gst_android_audioringbuffer_activate (GstRingBuffer * buf,
gboolean active);
static void gst_android_audioringbuffer_clear (GstRingBuffer * buf);
static guint gst_android_audioringbuffer_commit (GstRingBuffer * buf,
guint64 * sample, guchar * data, gint in_samples, gint out_samples,
gint * accum);
static void gst_audioflinger_sink_base_init (gpointer g_class);
static void gst_audioflinger_sink_class_init (GstAudioFlingerSinkClass * klass);
static void gst_audioflinger_sink_init (GstAudioFlingerSink *
audioflinger_sink);
static void gst_audioflinger_sink_dispose (GObject * object);
static void gst_audioflinger_sink_finalise (GObject * object);
static void gst_audioflinger_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audioflinger_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static GstCaps *gst_audioflinger_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_audioflinger_sink_open (GstAudioFlingerSink * asink);
static gboolean gst_audioflinger_sink_close (GstAudioFlingerSink * asink);
static gboolean gst_audioflinger_sink_prepare (GstAudioFlingerSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_audioflinger_sink_unprepare (GstAudioFlingerSink * asink);
static void gst_audioflinger_sink_reset (GstAudioFlingerSink * asink,
gboolean create_clock);
static void gst_audioflinger_sink_set_mute (GstAudioFlingerSink *
audioflinger_sink, gboolean mute);
static void gst_audioflinger_sink_set_volume (GstAudioFlingerSink *
audioflinger_sink, float volume);
static gboolean gst_audioflinger_sink_event (GstBaseSink * bsink,
GstEvent * event);
static GstRingBuffer *gst_audioflinger_sink_create_ringbuffer (GstBaseAudioSink
* sink);
static GstClockTime gst_audioflinger_sink_get_time (GstClock * clock,
gpointer user_data);
static GstFlowReturn gst_audioflinger_sink_preroll (GstBaseSink * bsink,
GstBuffer * buffer);
static GstClockTime gst_audioflinger_sink_system_audio_clock_get_time (GstClock
* clock, gpointer user_data);
static GstClock *gst_audioflinger_sink_provide_clock (GstElement * elem);
static GstStateChangeReturn gst_audioflinger_sink_change_state (GstElement *
element, GstStateChange transition);
static GstStaticPadTemplate audioflingersink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; ")
);
static GType
gst_android_audioringbuffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstAndroidAudioRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_android_audioringbuffer_class_init,
NULL,
NULL,
sizeof (GstAndroidAudioRingBuffer),
0,
(GInstanceInitFunc) gst_android_audioringbuffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER,
"GstAndroidAudioSinkRingBuffer", &ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_android_audioringbuffer_class_init (GstAndroidAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = G_OBJECT_CLASS (klass);
gstringbuffer_class = GST_RING_BUFFER_CLASS (klass);
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_android_audioringbuffer_dispose;
gobject_class->finalize = gst_android_audioringbuffer_finalize;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_release);
gstringbuffer_class->start =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_start);
gstringbuffer_class->pause =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_pause);
gstringbuffer_class->resume =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_start);
gstringbuffer_class->stop =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_stop);
gstringbuffer_class->clear_all =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_clear);
gstringbuffer_class->commit =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_commit);
#if 0
gstringbuffer_class->delay =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_delay);
#endif
gstringbuffer_class->activate =
GST_DEBUG_FUNCPTR (gst_android_audioringbuffer_activate);
}
static void
gst_android_audioringbuffer_init (G_GNUC_UNUSED GstAndroidAudioRingBuffer *
ringbuffer, G_GNUC_UNUSED GstAndroidAudioRingBufferClass * g_class)
{
}
static void
gst_android_audioringbuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_android_audioringbuffer_finalize (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_android_audioringbuffer_open_device (GstRingBuffer * buf)
{
GstAudioFlingerSink *sink;
gboolean result = TRUE;
LOGD (">gst_android_audioringbuffer_open_device");
sink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (buf));
result = gst_audioflinger_sink_open (sink);
if (!result)
goto could_not_open;
return result;
could_not_open:
{
GST_DEBUG_OBJECT (sink, "could not open device");
LOGE ("could not open device");
return FALSE;
}
}
static gboolean
gst_android_audioringbuffer_close_device (GstRingBuffer * buf)
{
GstAudioFlingerSink *sink;
gboolean result = TRUE;
LOGD (">gst_android_audioringbuffer_close_device");
sink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (buf));
result = gst_audioflinger_sink_close (sink);
if (!result)
goto could_not_close;
return result;
could_not_close:
{
GST_DEBUG_OBJECT (sink, "could not close device");
LOGE ("could not close device");
return FALSE;
}
}
static gboolean
gst_android_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec)
{
GstAudioFlingerSink *sink;
gboolean result = FALSE;
LOGD (">gst_android_audioringbuffer_acquire");
sink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (buf));
result = gst_audioflinger_sink_prepare (sink, spec);
if (!result)
goto could_not_prepare;
return TRUE;
/* ERRORS */
could_not_prepare:
{
GST_DEBUG_OBJECT (sink, "could not prepare device");
LOGE ("could not close device");
return FALSE;
}
}
static gboolean
gst_android_audioringbuffer_activate (G_GNUC_UNUSED GstRingBuffer * buf,
G_GNUC_UNUSED gboolean active)
{
return TRUE;
}
/* function is called with LOCK */
static gboolean
gst_android_audioringbuffer_release (GstRingBuffer * buf)
{
GstAudioFlingerSink *sink;
gboolean result = FALSE;
LOGD (">gst_android_audioringbuffer_release");
sink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (buf));
result = gst_audioflinger_sink_unprepare (sink);
if (!result)
goto could_not_unprepare;
GST_DEBUG_OBJECT (sink, "unprepared");
LOGD ("unprepared");
return result;
could_not_unprepare:
{
GST_DEBUG_OBJECT (sink, "could not unprepare device");
LOGE ("could not unprepare device");
return FALSE;
}
}
static gboolean
gst_android_audioringbuffer_start (GstRingBuffer * buf)
{
GstAudioFlingerSink *asink;
GstAndroidAudioRingBuffer *abuf;
abuf = GST_ANDROID_AUDIORING_BUFFER_CAST (buf);
asink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (abuf));
GST_INFO_OBJECT (buf, "starting ringbuffer");
LOGD ("starting ringbuffer");
audioflinger_device_start (asink->audioflinger_device);
return TRUE;
}
static gboolean
gst_android_audioringbuffer_pause (GstRingBuffer * buf)
{
GstAudioFlingerSink *asink;
GstAndroidAudioRingBuffer *abuf;
abuf = GST_ANDROID_AUDIORING_BUFFER_CAST (buf);
asink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (abuf));
GST_INFO_OBJECT (buf, "pausing ringbuffer");
LOGD ("pausing ringbuffer");
audioflinger_device_pause (asink->audioflinger_device);
return TRUE;
}
static gboolean
gst_android_audioringbuffer_stop (GstRingBuffer * buf)
{
GstAudioFlingerSink *asink;
GstAndroidAudioRingBuffer *abuf;
abuf = GST_ANDROID_AUDIORING_BUFFER_CAST (buf);
asink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (abuf));
GST_INFO_OBJECT (buf, "stopping ringbuffer");
LOGD ("stopping ringbuffer");
audioflinger_device_stop (asink->audioflinger_device);
return TRUE;
}
#if 0
static guint
gst_android_audioringbuffer_delay (GstRingBuffer * buf)
{
return 0;
}
#endif
static void
gst_android_audioringbuffer_clear (GstRingBuffer * buf)
{
GstAudioFlingerSink *asink;
GstAndroidAudioRingBuffer *abuf;
abuf = GST_ANDROID_AUDIORING_BUFFER_CAST (buf);
asink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (abuf));
GST_INFO_OBJECT (buf, "clearing ringbuffer");
LOGD ("clearing ringbuffer");
if (asink->audioflinger_device == NULL)
return;
GST_INFO_OBJECT (asink, "resetting clock");
gst_audio_clock_reset (GST_AUDIO_CLOCK (asink->audio_clock), 0);
audioflinger_device_flush (asink->audioflinger_device);
}
#define FWD_SAMPLES(s,se,d,de) \
G_STMT_START { \
/* no rate conversion */ \
guint towrite = MIN (se + bps - s, de - d); \
/* simple copy */ \
if (!skip) \
memcpy (d, s, towrite); \
in_samples -= towrite / bps; \
out_samples -= towrite / bps; \
s += towrite; \
GST_LOG ("copy %u bytes", towrite); \
} G_STMT_END
/* in_samples >= out_samples, rate > 1.0 */
#define FWD_UP_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
memcpy (d, s, bps); \
s += bps; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* out_samples > in_samples, for rates smaller than 1.0 */
#define FWD_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
memcpy (d, s, bps); \
d += bps; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
s += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_UP_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
memcpy (d, se, bps); \
se -= bps; \
*accum += outr; \
while (d < de && (*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
if (!skip) \
memcpy (d, se, bps); \
d += bps; \
*accum += inr; \
while (s <= se && (*accum << 1) >= outr) { \
*accum -= outr; \
se -= bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
static guint
gst_android_audioringbuffer_commit (GstRingBuffer * buf, guint64 * sample,
guchar * data, gint in_samples, gint out_samples, gint * accum)
{
GstBaseAudioSink *baseaudiosink;
GstAudioFlingerSink *asink;
GstAndroidAudioRingBuffer *abuf;
guint result;
guint8 *data_end;
gboolean reverse;
gint *toprocess;
gint inr, outr, bps;
guint bufsize;
gboolean skip = FALSE;
guint32 position;
gboolean slaved;
guint64 corrected_sample;
gboolean sync;
abuf = GST_ANDROID_AUDIORING_BUFFER_CAST (buf);
asink = GST_AUDIOFLINGERSINK (GST_OBJECT_PARENT (abuf));
baseaudiosink = GST_BASE_AUDIO_SINK (asink);
sync = gst_base_sink_get_sync (GST_BASE_SINK_CAST (asink));
GST_LOG_OBJECT (asink, "entering commit");
/* make sure the ringbuffer is started */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (g_atomic_int_get (&buf->abidata.ABI.may_start) == FALSE))
goto no_start;
GST_LOG_OBJECT (buf, "start!");
LOGD ("start!");
if (!gst_ring_buffer_start (buf))
goto start_failed;
}
slaved = GST_ELEMENT_CLOCK (baseaudiosink) != asink->exported_clock;
if (asink->last_resync_sample == -1 ||
(gint64) baseaudiosink->next_sample == -1) {
if (slaved) {
/* we're writing a discont buffer. Disable slaving for a while in order to
* fill the initial buffer needed by the audio mixer thread. This avoids
* some cases where audioflinger removes us from the list of active tracks
* because we aren't writing enough data.
*/
GST_INFO_OBJECT (asink, "no previous sample, now %" G_GINT64_FORMAT
" disabling slaving", *sample);
LOGD ("no previous sample, now %ld disabling slaving", *sample);
asink->last_resync_sample = *sample;
g_object_set (asink, "slave-method", GST_BASE_AUDIO_SINK_SLAVE_NONE,
NULL);
asink->slaving_disabled = TRUE;
} else {
/* Trace displayed too much time : remove it
GST_INFO_OBJECT (asink, "no previous sample but not slaved");
LOGD("no previous sample but not slaved");
*/
}
}
if (slaved && asink->slaving_disabled) {
guint64 threshold;
threshold = gst_util_uint64_scale_int (buf->spec.rate, 5, 1);
threshold += asink->last_resync_sample;
if (*sample >= threshold) {
GST_INFO_OBJECT (asink, "last sync %" G_GINT64_FORMAT
" reached sample %" G_GINT64_FORMAT ", enabling slaving",
asink->last_resync_sample, *sample);
g_object_set (asink, "slave-method", GST_BASE_AUDIO_SINK_SLAVE_SKEW,
NULL);
asink->slaving_disabled = FALSE;
}
}
bps = buf->spec.bytes_per_sample;
bufsize = buf->spec.segsize * buf->spec.segtotal;
/* our toy resampler for trick modes */
reverse = out_samples < 0;
out_samples = ABS (out_samples);
if (in_samples >= out_samples)
toprocess = &in_samples;
else
toprocess = &out_samples;
inr = in_samples - 1;
outr = out_samples - 1;
GST_LOG_OBJECT (asink, "in %d, out %d reverse %d sync %d", inr, outr,
reverse, sync);
/* data_end points to the last sample we have to write, not past it. This is
* needed to properly handle reverse playback: it points to the last sample. */
data_end = data + (bps * inr);
while (*toprocess > 0) {
if (sync) {
size_t avail;
guint towrite;
gint err;
guint8 *d, *d_end;
gpointer buffer_handle;
position = audioflinger_device_get_position (asink->audioflinger_device);
avail = out_samples;
buffer_handle = NULL;
GST_LOG_OBJECT (asink, "calling obtain buffer, position %d"
" offset %" G_GINT64_FORMAT " samples %" G_GSSIZE_FORMAT,
position, *sample, avail);
err = audioflinger_device_obtain_buffer (asink->audioflinger_device,
&buffer_handle, (int8_t **) & d, &avail, *sample);
GST_LOG_OBJECT (asink, "obtain buffer returned");
if (err < 0) {
GST_LOG_OBJECT (asink, "obtain buffer error %d, state %d",
err, buf->state);
LOGD ("obtain buffer error 0x%x, state %d", err, buf->state);
if (err == LATE)
skip = TRUE;
else if (buf->state != GST_RING_BUFFER_STATE_STARTED)
goto done;
else
goto obtain_buffer_failed;
}
towrite = avail * bps;
d_end = d + towrite;
GST_LOG_OBJECT (asink, "writing %u samples at offset %" G_GUINT64_FORMAT,
(guint) avail, *sample);
if (G_LIKELY (inr == outr && !reverse)) {
FWD_SAMPLES (data, data_end, d, d_end);
} else if (!reverse) {
if (inr >= outr) {
/* forward speed up */
FWD_UP_SAMPLES (data, data_end, d, d_end);
} else {
/* forward slow down */
FWD_DOWN_SAMPLES (data, data_end, d, d_end);
}
} else {
if (inr >= outr)
/* reverse speed up */
REV_UP_SAMPLES (data, data_end, d, d_end);
else
/* reverse slow down */
REV_DOWN_SAMPLES (data, data_end, d, d_end);
}
*sample += avail;
if (buffer_handle)
audioflinger_device_release_buffer (asink->audioflinger_device,
buffer_handle);
} else {
gint written;
written = audioflinger_device_write (asink->audioflinger_device, data,
*toprocess * bps);
if (written > 0) {
*toprocess -= written / bps;
data += written;
} else {
LOGE ("Error to write buffer(error=%d)", written);
GST_LOG_OBJECT (asink, "Error to write buffer(error=%d)", written);
goto start_failed;
}
}
}
skip:
/* we consumed all samples here */
data = data_end + bps;
done:
result = inr - ((data_end - data) / bps);
GST_LOG_OBJECT (asink, "wrote %d samples", result);
return result;
/* ERRORS */
no_start:
{
GST_LOG_OBJECT (asink, "we can not start");
LOGE ("we can not start");
return 0;
}
start_failed:
{
GST_LOG_OBJECT (asink, "failed to start the ringbuffer");
LOGE ("failed to start the ringbuffer");
return 0;
}
obtain_buffer_failed:
{
GST_ELEMENT_ERROR (asink, RESOURCE, FAILED,
("obtain_buffer failed"), (NULL));
LOGE ("obtain_buffer failed");
return -1;
}
}
static GstElementClass *parent_class = NULL;
GType
gst_audioflinger_sink_get_type (void)
{
static GType audioflingersink_type = 0;
if (!audioflingersink_type) {
static const GTypeInfo audioflingersink_info = {
sizeof (GstAudioFlingerSinkClass),
gst_audioflinger_sink_base_init,
NULL,
(GClassInitFunc) gst_audioflinger_sink_class_init,
NULL,
NULL,
sizeof (GstAudioFlingerSink),
0,
(GInstanceInitFunc) gst_audioflinger_sink_init,
};
audioflingersink_type =
g_type_register_static (GST_TYPE_AUDIO_SINK, "GstAudioFlingerSink",
&audioflingersink_info, 0);
}
return audioflingersink_type;
}
static void
gst_audioflinger_sink_dispose (GObject * object)
{
GstAudioFlingerSink *audioflinger_sink = GST_AUDIOFLINGERSINK (object);
if (audioflinger_sink->probed_caps) {
gst_caps_unref (audioflinger_sink->probed_caps);
audioflinger_sink->probed_caps = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audioflinger_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_audioflinger_sink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&audioflingersink_sink_factory));
GST_DEBUG_CATEGORY_INIT (audioflinger_debug, "audioflingersink", 0,
"audioflinger sink trace");
}
static void
gst_audioflinger_sink_class_init (GstAudioFlingerSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioflinger_sink_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioflinger_sink_finalise);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_get_property);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_set_property);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_provide_clock);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_change_state);
gstbasesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_getcaps);
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_create_ringbuffer);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audioflinger_sink_event);
gstbasesink_class->preroll =
GST_DEBUG_FUNCPTR (gst_audioflinger_sink_preroll);
/* Install properties */
g_object_class_install_property (gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute output", DEFAULT_MUTE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"control volume size", 0.0, 10.0, DEFAULT_VOLUME, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_AUDIO_SINK,
g_param_spec_pointer ("audiosink", "AudioSink",
"The pointer of MediaPlayerBase::AudioSink", G_PARAM_WRITABLE));
}
static void
gst_audioflinger_sink_init (GstAudioFlingerSink * audioflinger_sink)
{
GST_DEBUG_OBJECT (audioflinger_sink, "initializing audioflinger_sink");
LOGD ("initializing audioflinger_sink");
audioflinger_sink->audio_clock = NULL;
audioflinger_sink->system_clock = NULL;
audioflinger_sink->system_audio_clock = NULL;
audioflinger_sink->exported_clock = NULL;
audioflinger_sink->export_system_audio_clock =
DEFAULT_EXPORT_SYSTEM_AUDIO_CLOCK;
gst_audioflinger_sink_reset (audioflinger_sink, TRUE);
}
static void
gst_audioflinger_sink_reset (GstAudioFlingerSink * sink, gboolean create_clocks)
{
if (sink->audioflinger_device != NULL) {
audioflinger_device_release (sink->audioflinger_device);
sink->audioflinger_device = NULL;
}
sink->audioflinger_device = NULL;
sink->m_volume = DEFAULT_VOLUME;
sink->m_mute = DEFAULT_MUTE;
sink->m_init = FALSE;
sink->m_audiosink = NULL;
sink->eos = FALSE;
sink->may_provide_clock = TRUE;
sink->last_resync_sample = -1;
if (sink->system_clock) {
GstClock *clock = sink->system_clock;
GST_INFO_OBJECT (sink, "destroying system_clock %d",
GST_OBJECT_REFCOUNT (sink->system_clock));
gst_clock_set_master (sink->system_clock, NULL);
gst_object_replace ((GstObject **) & sink->system_clock, NULL);
GST_INFO_OBJECT (sink, "destroyed system_clock");
GST_INFO_OBJECT (sink, "destroying system_audio_clock %d",
GST_OBJECT_REFCOUNT (sink->system_audio_clock));
gst_object_replace ((GstObject **) & sink->system_audio_clock, NULL);
GST_INFO_OBJECT (sink, "destroyed system_audio_clock");
}
if (sink->audio_clock) {
GST_INFO_OBJECT (sink, "destroying audio clock %d",
GST_OBJECT_REFCOUNT (sink->audio_clock));
gst_object_replace ((GstObject **) & sink->audio_clock, NULL);
}
if (sink->exported_clock) {
GST_INFO_OBJECT (sink, "destroying exported clock %d",
GST_OBJECT_REFCOUNT (sink->exported_clock));
gst_object_replace ((GstObject **) & sink->exported_clock, NULL);
GST_INFO_OBJECT (sink, "destroyed exported clock");
}
if (create_clocks) {
GstClockTime external, internal;
/* create the audio clock that uses the ringbuffer as its audio source */
sink->audio_clock = gst_audio_clock_new ("GstAudioFlingerSinkClock",
gst_audioflinger_sink_get_time, sink);
/* always set audio_clock as baseaudiosink's provided_clock */
gst_object_replace ((GstObject **) &
GST_BASE_AUDIO_SINK (sink)->provided_clock,
GST_OBJECT (sink->audio_clock));
/* create the system_audio_clock, which is an *audio clock* that uses an
* instance of the system clock as its time source */
sink->system_audio_clock =
gst_audio_clock_new ("GstAudioFlingerSystemAudioClock",
gst_audioflinger_sink_system_audio_clock_get_time, sink);
/* create an instance of the system clock, that we slave to
* sink->audio_clock to have an audio clock with an higher resolution than
* the segment size (50ms) */
sink->system_clock = g_object_new (GST_TYPE_SYSTEM_CLOCK,
"name", "GstAudioFlingerSystemClock", NULL);
/* calibrate the clocks */
external = gst_clock_get_time (sink->audio_clock);
internal = gst_clock_get_internal_time (sink->system_clock);
gst_clock_set_calibration (sink->system_clock, internal, external, 1, 1);
/* slave the system clock to the audio clock */
GST_OBJECT_FLAG_SET (sink->system_clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
g_object_set (sink->system_clock, "timeout", 50 * GST_MSECOND, NULL);
gst_clock_set_master (sink->system_clock, sink->audio_clock);
}
}
static void
gst_audioflinger_sink_finalise (GObject * object)
{
GstAudioFlingerSink *audioflinger_sink = GST_AUDIOFLINGERSINK (object);
GST_INFO_OBJECT (object, "finalize");
gst_audioflinger_sink_reset (audioflinger_sink, FALSE);
G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));
}
static GstRingBuffer *
gst_audioflinger_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
LOGD ("creating ringbuffer");
buffer = g_object_new (GST_TYPE_ANDROID_AUDIORING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
LOGD ("created ringbuffer @%p", buffer);
return buffer;
}
static void
gst_audioflinger_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioFlingerSink *audioflinger_sink;
audioflinger_sink = GST_AUDIOFLINGERSINK (object);
g_return_if_fail (audioflinger_sink != NULL);
switch (prop_id) {
case PROP_MUTE:
g_value_set_boolean (value, audioflinger_sink->m_mute);
GST_DEBUG_OBJECT (audioflinger_sink, "get mute: %d",
audioflinger_sink->m_mute);
LOGD ("get mute: %d", audioflinger_sink->m_mute);
break;
case PROP_VOLUME:
g_value_set_double (value, audioflinger_sink->m_volume);
GST_DEBUG_OBJECT (audioflinger_sink, "get volume: %f",
audioflinger_sink->m_volume);
LOGD ("get volume: %f", audioflinger_sink->m_volume);
break;
case PROP_AUDIO_SINK:
GST_ERROR_OBJECT (audioflinger_sink, "Shall not go here!");
LOGD ("Shall not go here!");
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioflinger_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFlingerSink *audioflinger_sink;
audioflinger_sink = GST_AUDIOFLINGERSINK (object);
g_return_if_fail (audioflinger_sink != NULL);
GST_OBJECT_LOCK (audioflinger_sink);
switch (prop_id) {
case PROP_MUTE:
audioflinger_sink->m_mute = g_value_get_boolean (value);
GST_DEBUG_OBJECT (audioflinger_sink, "set mute: %d",
audioflinger_sink->m_mute);
LOGD ("set mute: %d", audioflinger_sink->m_mute);
/* set device if it's initialized */
if (audioflinger_sink->audioflinger_device && audioflinger_sink->m_init)
gst_audioflinger_sink_set_mute (audioflinger_sink,
(int) (audioflinger_sink->m_mute));
break;
case PROP_VOLUME:
audioflinger_sink->m_volume = g_value_get_double (value);
GST_DEBUG_OBJECT (audioflinger_sink, "set volume: %f",
audioflinger_sink->m_volume);
LOGD ("set volume: %f", audioflinger_sink->m_volume);
/* set device if it's initialized */
if (audioflinger_sink->audioflinger_device && audioflinger_sink->m_init)
gst_audioflinger_sink_set_volume (audioflinger_sink,
(float) audioflinger_sink->m_volume);
break;
case PROP_AUDIO_SINK:
audioflinger_sink->m_audiosink = g_value_get_pointer (value);
GST_DEBUG_OBJECT (audioflinger_sink, "set audiosink: %p",
audioflinger_sink->m_audiosink);
LOGD ("set audiosink: %p", audioflinger_sink->m_audiosink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (audioflinger_sink);
}
static GstCaps *
gst_audioflinger_sink_getcaps (GstBaseSink * bsink)
{
GstAudioFlingerSink *audioflinger_sink;
GstCaps *caps;
audioflinger_sink = GST_AUDIOFLINGERSINK (bsink);
GST_DEBUG_OBJECT (audioflinger_sink, "enter,%p",
audioflinger_sink->audioflinger_device);
LOGD ("gst_audioflinger_sink_getcaps,%p",
audioflinger_sink->audioflinger_device);
if (audioflinger_sink->audioflinger_device == NULL
|| audioflinger_sink->m_init == FALSE) {
caps =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
(bsink)));
} else if (audioflinger_sink->probed_caps) {
caps = gst_caps_copy (audioflinger_sink->probed_caps);
} else {
caps = gst_caps_new_any ();
if (caps && !gst_caps_is_empty (caps)) {
audioflinger_sink->probed_caps = gst_caps_copy (caps);
}
}
return caps;
}
static gboolean
gst_audioflinger_sink_open (GstAudioFlingerSink * audioflinger)
{
GstBaseAudioSink *baseaudiosink = (GstBaseAudioSink *) audioflinger;
GST_DEBUG_OBJECT (audioflinger, "enter");
LOGD ("gst_audioflinger_sink_open");
g_return_val_if_fail (audioflinger != NULL, FALSE);
baseaudiosink->buffer_time = DEFAULT_BUFFERTIME;
baseaudiosink->latency_time = DEFAULT_LATENCYTIME;
if (audioflinger->audioflinger_device == NULL) {
if (audioflinger->m_audiosink) {
if (!(audioflinger->audioflinger_device =
audioflinger_device_open (audioflinger->m_audiosink)))
goto failed_creation;
GST_DEBUG_OBJECT (audioflinger, "open an existed flinger, %p",
audioflinger->audioflinger_device);
LOGD ("open an existed flinger, %p", audioflinger->audioflinger_device);
} else {
if (!(audioflinger->audioflinger_device = audioflinger_device_create ()))
goto failed_creation;
GST_DEBUG_OBJECT (audioflinger, "create a new flinger, %p",
audioflinger->audioflinger_device);
LOGD ("create a new flinger, %p", audioflinger->audioflinger_device);
}
}
return TRUE;
/* ERRORS */
failed_creation:
{
GST_ELEMENT_ERROR (audioflinger, RESOURCE, SETTINGS, (NULL),
("Failed to create AudioFlinger"));
LOGE ("Failed to create AudioFlinger");
return FALSE;
}
}
static gboolean
gst_audioflinger_sink_close (GstAudioFlingerSink * audioflinger)
{
GST_DEBUG_OBJECT (audioflinger, "enter");
LOGD ("gst_audioflinger_sink_close");
if (audioflinger->audioflinger_device != NULL) {
GST_DEBUG_OBJECT (audioflinger, "release flinger device");
LOGD ("release flinger device");
audioflinger_device_stop (audioflinger->audioflinger_device);
audioflinger_device_release (audioflinger->audioflinger_device);
audioflinger->audioflinger_device = NULL;
}
return TRUE;
}
static gboolean
gst_audioflinger_sink_prepare (GstAudioFlingerSink * audioflinger,
GstRingBufferSpec * spec)
{
GST_DEBUG_OBJECT (audioflinger, "enter");
LOGD ("gst_audioflinger_sink_prepare");
/* FIXME:
*
* Pipeline crashes in audioflinger_device_set(), after releasing audio
* flinger device and creating it again. In most cases, it will happen when
* playing the same audio again.
*
* It seems the root cause is we create and release audio flinger sink in
* different thread in playbin2. Till now, I haven't found way to
* create/release device in the same thread. Fortunately, it will not effect
* the gst-launch usage
*/
if (audioflinger_device_set (audioflinger->audioflinger_device,
3, spec->channels, spec->rate, spec->segsize) == -1)
goto failed_creation;
audioflinger->m_init = TRUE;
// gst_audioflinger_sink_set_volume (audioflinger, audioflinger->m_volume);
// gst_audioflinger_sink_set_mute (audioflinger, audioflinger->m_mute);
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
audioflinger->bytes_per_sample = spec->bytes_per_sample;
spec->segsize =
audioflinger_device_frameCount (audioflinger->audioflinger_device);
GST_DEBUG_OBJECT (audioflinger,
"channels: %d, rate: %d, width: %d, got segsize: %d, segtotal: %d, "
"frame count: %d, frame size: %d",
spec->channels, spec->rate, spec->width, spec->segsize, spec->segtotal,
audioflinger_device_frameCount (audioflinger->audioflinger_device),
audioflinger_device_frameSize (audioflinger->audioflinger_device)
);
LOGD ("channels: %d, rate: %d, width: %d, got segsize: %d, segtotal: %d, "
"frame count: %d, frame size: %d",
spec->channels, spec->rate, spec->width, spec->segsize, spec->segtotal,
audioflinger_device_frameCount (audioflinger->audioflinger_device),
audioflinger_device_frameSize (audioflinger->audioflinger_device)
);
#if 0
GST_DEBUG_OBJECT (audioflinger, "pause device");
LOGD ("pause device");
audioflinger_device_pause (audioflinger->audioflinger_device);
#endif
return TRUE;
/* ERRORS */
failed_creation:
{
GST_ELEMENT_ERROR (audioflinger, RESOURCE, SETTINGS, (NULL),
("Failed to create AudioFlinger for format %d", spec->format));
LOGE ("Failed to create AudioFlinger for format %d", spec->format);
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (audioflinger, RESOURCE, SETTINGS, (NULL),
("Unhandled width %d", spec->width));
LOGE ("Unhandled width %d", spec->width);
return FALSE;
}
}
static gboolean
gst_audioflinger_sink_unprepare (GstAudioFlingerSink * audioflinger)
{
GST_DEBUG_OBJECT (audioflinger, "enter");
LOGD ("gst_audioflinger_sink_unprepare");
if (audioflinger->audioflinger_device != NULL) {
GST_DEBUG_OBJECT (audioflinger, "release flinger device");
LOGD ("release flinger device");
audioflinger_device_stop (audioflinger->audioflinger_device);
audioflinger->m_init = FALSE;
}
return TRUE;
}
static void
gst_audioflinger_sink_set_mute (GstAudioFlingerSink * audioflinger_sink,
gboolean mute)
{
GST_DEBUG_OBJECT (audioflinger_sink, "set PROP_MUTE = %d\n", mute);
LOGD ("set PROP_MUTE = %d\n", mute);
if (audioflinger_sink->audioflinger_device)
audioflinger_device_mute (audioflinger_sink->audioflinger_device, mute);
audioflinger_sink->m_mute = mute;
}
static void
gst_audioflinger_sink_set_volume (GstAudioFlingerSink * audioflinger_sink,
float volume)
{
GST_DEBUG_OBJECT (audioflinger_sink, "set PROP_VOLUME = %f\n", volume);
LOGD ("set PROP_VOLUME = %f\n", volume);
if (audioflinger_sink->audioflinger_device != NULL) {
audioflinger_device_set_volume (audioflinger_sink->audioflinger_device,
volume, volume);
}
}
gboolean
gst_audioflinger_sink_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "audioflingersink", GST_RANK_PRIMARY,
GST_TYPE_AUDIOFLINGERSINK);
}
/*
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "audioflingersink",
"audioflinger sink audio", plugin_init, VERSION, "LGPL", "GStreamer",
"http://gstreamer.net/")
*/
static GstClock *
gst_audioflinger_sink_provide_clock (GstElement * elem)
{
GstBaseAudioSink *sink;
GstAudioFlingerSink *asink;
GstClock *clock;
sink = GST_BASE_AUDIO_SINK (elem);
asink = GST_AUDIOFLINGERSINK (elem);
/* we have no ringbuffer (must be NULL state) */
if (sink->ringbuffer == NULL)
goto wrong_state;
if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (sink);
if (!asink->may_provide_clock)
goto already_playing;
if (!sink->provide_clock)
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (asink->exported_clock));
GST_INFO_OBJECT (asink, "providing clock %p %s", clock,
clock == NULL ? NULL : GST_OBJECT_NAME (clock));
GST_OBJECT_UNLOCK (sink);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
LOGD ("ringbuffer not acquired");
return NULL;
}
already_playing:
{
GST_INFO_OBJECT (sink, "we went to playing already");
GST_OBJECT_UNLOCK (sink);
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (sink, "clock provide disabled");
LOGD ("clock provide disabled");
GST_OBJECT_UNLOCK (sink);
return NULL;
}
}
static GstStateChangeReturn
gst_audioflinger_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
GstClockTime time;
GstAudioFlingerSink *sink = GST_AUDIOFLINGERSINK (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
sink->may_provide_clock = FALSE;
if (sink->exported_clock == sink->system_audio_clock) {
GstClockTime cinternal, cexternal, crate_num, crate_denom;
/* take the slave lock to make sure that the slave_callback doesn't run
* while we're moving sink->audio_clock forward, causing
* sink->system_clock to jump as well */
GST_CLOCK_SLAVE_LOCK (sink->system_clock);
gst_clock_get_calibration (sink->audio_clock, NULL, NULL,
&crate_num, &crate_denom);
cinternal = gst_clock_get_internal_time (sink->audio_clock);
cexternal = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
gst_clock_set_calibration (sink->audio_clock, cinternal, cexternal,
crate_num, crate_denom);
/* reset observations */
sink->system_clock->filling = TRUE;
sink->system_clock->time_index = 0;
GST_CLOCK_SLAVE_UNLOCK (sink->system_clock);
time = gst_clock_get_time (sink->audio_clock);
GST_INFO_OBJECT (sink, "PAUSED_TO_PLAYING,"
" base_time %" GST_TIME_FORMAT
" after %" GST_TIME_FORMAT
" internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_ELEMENT (sink)->base_time),
GST_TIME_ARGS (time),
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
}
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
default:
break;
}
return ret;
}
static GstFlowReturn
gst_audioflinger_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
{
GstFlowReturn ret;
gboolean us_live = FALSE;
GstQuery *query;
GstAudioFlingerSink *asink = GST_AUDIOFLINGERSINK (bsink);
GstBaseAudioSink *baseaudiosink = GST_BASE_AUDIO_SINK (bsink);
GstClock *clock;
GST_INFO_OBJECT (bsink, "preroll");
ret = GST_BASE_SINK_CLASS (parent_class)->preroll (bsink, buffer);
if (ret != GST_FLOW_OK)
goto done;
if (asink->exported_clock != NULL) {
GST_INFO_OBJECT (bsink, "clock already exported");
goto done;
}
query = gst_query_new_latency ();
/* ask the peer for the latency */
if (gst_pad_peer_query (bsink->sinkpad, query)) {
/* get upstream min and max latency */
gst_query_parse_latency (query, &us_live, NULL, NULL);
GST_INFO_OBJECT (bsink, "query result live: %d", us_live);
} else {
GST_WARNING_OBJECT (bsink, "latency query failed");
}
gst_query_unref (query);
if (!us_live && asink->export_system_audio_clock) {
clock = asink->system_audio_clock;
/* set SLAVE_NONE so that baseaudiosink doesn't try to slave audio_clock to
* system_audio_clock
*/
g_object_set (asink, "slave-method", GST_BASE_AUDIO_SINK_SLAVE_NONE, NULL);
} else {
clock = asink->audio_clock;
}
GST_INFO_OBJECT (bsink, "using %s clock",
clock == asink->audio_clock ? "audio" : "system_audio");
gst_object_replace ((GstObject **) & asink->exported_clock,
GST_OBJECT (clock));
GST_OBJECT_UNLOCK (asink);
done:
return ret;
}
static gboolean
gst_audioflinger_sink_event (GstBaseSink * bsink, GstEvent * event)
{
GstAudioFlingerSink *asink = GST_AUDIOFLINGERSINK (bsink);
GstBaseAudioSink *baseaudiosink = GST_BASE_AUDIO_SINK (bsink);
GstRingBuffer *ringbuf = baseaudiosink->ringbuffer;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
GST_INFO_OBJECT (asink, "got EOS");
asink->eos = TRUE;
if (baseaudiosink->next_sample) {
guint64 next_sample, sample;
gint sps;
GstFlowReturn ret;
GstBuffer *buf;
sps = ringbuf->spec.segsize / ringbuf->spec.bytes_per_sample;
sample = baseaudiosink->next_sample;
next_sample = baseaudiosink->next_sample / sps;
if (next_sample < ringbuf->spec.segsize) {
gint samples, out_samples, accum, size;
GstClockTime timestamp, before, after;
guchar *data, *data_start;
gint64 drift_tolerance;
guint written;
gint64 offset;
samples = (ringbuf->spec.segsize - next_sample) * 4;
size = samples * ringbuf->spec.bytes_per_sample;
timestamp = gst_util_uint64_scale_int (baseaudiosink->next_sample,
GST_SECOND, ringbuf->spec.rate);
before = gst_clock_get_internal_time (asink->audio_clock);
GST_INFO_OBJECT (asink, "%" G_GINT64_FORMAT " < %d, "
"padding with silence, samples %d size %d ts %" GST_TIME_FORMAT,
next_sample, ringbuf->spec.segsize, samples, size,
GST_TIME_ARGS (timestamp));
LOGD ("PADDING");
data_start = data = g_malloc0 (size);
offset = baseaudiosink->next_sample;
out_samples = samples;
GST_STATE_LOCK (bsink);
do {
written =
gst_ring_buffer_commit_full (ringbuf, &offset, data, samples,
out_samples, &accum);
GST_DEBUG_OBJECT (bsink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (written == samples)
break;
/* else something interrupted us and we wait for preroll. */
if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
break;
/* update the output samples. FIXME, this will just skip them when pausing
* during trick mode */
if (out_samples > written) {
out_samples -= written;
accum = 0;
} else
break;
samples -= written;
data += written * ringbuf->spec.bytes_per_sample;
} while (TRUE);
GST_STATE_UNLOCK (bsink);
g_free (data_start);
after = gst_clock_get_internal_time (asink->audio_clock);
GST_INFO_OBJECT (asink, "padded, left %d before %" GST_TIME_FORMAT
" after %" GST_TIME_FORMAT, samples,
GST_TIME_ARGS (before), GST_TIME_ARGS (after));
} else {
LOGD ("NOT PADDING 1");
}
} else {
LOGD ("NOT PADDING 2");
}
break;
case GST_EVENT_BUFFERING_START:
GST_INFO_OBJECT (asink, "buffering start");
break;
case GST_EVENT_BUFFERING_STOP:
{
gboolean slaved;
GstClockTime cinternal, cexternal, crate_num, crate_denom;
GstClockTime before, after;
gst_clock_get_calibration (asink->audio_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
before = gst_clock_get_time (asink->audio_clock);
cinternal = gst_clock_get_internal_time (asink->audio_clock);
cexternal = gst_clock_get_time (GST_ELEMENT_CLOCK (asink));
gst_clock_set_calibration (asink->audio_clock, cinternal,
cexternal, crate_num, crate_denom);
after = gst_clock_get_time (asink->audio_clock);
GST_INFO_OBJECT (asink, "buffering stopped, clock recalibrated"
" before %" GST_TIME_FORMAT " after %" GST_TIME_FORMAT,
GST_TIME_ARGS (before), GST_TIME_ARGS (after));
/* force baseaudiosink to resync from the next buffer */
GST_BASE_AUDIO_SINK (asink)->next_sample = -1;
/* reset this so we allow some time before enabling slaving again */
asink->last_resync_sample = -1;
slaved = GST_ELEMENT_CLOCK (asink) != asink->exported_clock;
if (slaved) {
GST_INFO_OBJECT (asink, "disabling slaving");
g_object_set (asink, "slave-method", GST_BASE_AUDIO_SINK_SLAVE_NONE,
NULL);
asink->slaving_disabled = TRUE;
}
g_object_set (asink, "drift-tolerance", 200 * GST_MSECOND, NULL);
break;
}
default:
break;
}
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
}
static GstClockTime
gst_audioflinger_sink_get_time (GstClock * clock, gpointer user_data)
{
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (user_data);
uint32_t position = -1;
GstAudioFlingerSink *asink = GST_AUDIOFLINGERSINK (sink);
GstClockTime time = GST_CLOCK_TIME_NONE;
GstClockTime ptime = GST_CLOCK_TIME_NONE;
GstClockTime system_audio_clock_time = GST_CLOCK_TIME_NONE;
GstClockTime offset = GST_CLOCK_TIME_NONE;
GstClockTime adjusted_time = GST_CLOCK_TIME_NONE;
GstClockTime cinternal, cexternal, crate_num, crate_denom;
gst_clock_get_calibration (clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
if (!asink->audioflinger_device || !asink->m_init) {
GST_DEBUG_OBJECT (sink, "device not created yet");
goto out;
}
if (!asink->audioflinger_device || !asink->m_init) {
GST_DEBUG_OBJECT (sink, "device not created yet");
goto out;
}
if (!sink->ringbuffer) {
GST_DEBUG_OBJECT (sink, "NULL ringbuffer");
goto out;
}
if (!sink->ringbuffer->acquired) {
GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
goto out;
}
position = audioflinger_device_get_position (asink->audioflinger_device);
if (position == -1)
goto out;
time = gst_util_uint64_scale_int (position, GST_SECOND,
sink->ringbuffer->spec.rate);
offset = gst_audio_clock_adjust (GST_CLOCK (clock), 0);
adjusted_time = gst_audio_clock_adjust (GST_CLOCK (clock), time);
if (asink->system_audio_clock)
system_audio_clock_time = gst_clock_get_time (asink->system_audio_clock);
if (GST_ELEMENT_CLOCK (asink)
&& asink->audio_clock != GST_ELEMENT_CLOCK (asink))
ptime = gst_clock_get_time (GST_ELEMENT_CLOCK (asink));
out:
GST_DEBUG_OBJECT (sink,
"clock %s processed samples %" G_GINT32_FORMAT " offset %" GST_TIME_FORMAT
" time %" GST_TIME_FORMAT " pipeline time %" GST_TIME_FORMAT
" system audio clock %" GST_TIME_FORMAT " adjusted_time %" GST_TIME_FORMAT
" cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
GST_OBJECT_NAME (clock), position, GST_TIME_ARGS (offset),
GST_TIME_ARGS (time), GST_TIME_ARGS (ptime),
GST_TIME_ARGS (system_audio_clock_time), GST_TIME_ARGS (adjusted_time),
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
return time;
}
static GstClockTime
gst_audioflinger_sink_system_audio_clock_get_time (GstClock * clock,
gpointer user_data)
{
GstClockTime time, offset;
GstAudioFlingerSink *sink = GST_AUDIOFLINGERSINK (user_data);
time = gst_clock_get_time (sink->system_clock);
offset = gst_audio_clock_adjust (clock, (GstClockTime) 0);
time -= offset;
return time;
}
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