+++ /dev/null
-/* 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.
- */
-#define ENABLE_GST_PLAYER_LOG
-#include <media/AudioTrack.h>
-#include <utils/Log.h>
-#include <AudioFlinger.h>
-#include <MediaPlayerInterface.h>
-#include <MediaPlayerService.h>
-#include "audioflinger_wrapper.h"
-#include <glib/glib.h>
-//#include <GstLog.h>
-
-
-#define LOG_NDEBUG 0
-
-#undef LOG_TAG
-#define LOG_TAG "audioflinger_wrapper"
-
-
-using namespace android;
-
-
-typedef struct _AudioFlingerDevice
-{
- AudioTrack* audio_track;
- bool init;
- sp<MediaPlayerBase::AudioSink> audio_sink;
- bool audio_sink_specified;
-} AudioFlingerDevice;
-
-
-/* commonly used macro */
-#define AUDIO_FLINGER_DEVICE(handle) ((AudioFlingerDevice*)handle)
-#define AUDIO_FLINGER_DEVICE_TRACK(handle) \
- (AUDIO_FLINGER_DEVICE(handle)->audio_track)
-#define AUDIO_FLINGER_DEVICE_SINK(handle) \
- (AUDIO_FLINGER_DEVICE(handle)->audio_sink)
-
-
-AudioFlingerDeviceHandle audioflinger_device_create()
-{
- AudioFlingerDevice* audiodev = NULL;
- AudioTrack *audiotr = NULL;
-
- // create a new instance of AudioFlinger
- audiodev = new AudioFlingerDevice;
- if (audiodev == NULL) {
- LOGE("Error to create AudioFlingerDevice\n");
- return NULL;
- }
-
- // create AudioTrack
- audiotr = new AudioTrack ();
- if (audiotr == NULL) {
- LOGE("Error to create AudioTrack\n");
- return NULL;
- }
-
- audiodev->init = false;
- audiodev->audio_track = (AudioTrack *) audiotr;
- audiodev->audio_sink = 0;
- audiodev->audio_sink_specified = false;
- LOGD("Create AudioTrack successfully %p\n",audiodev);
-
- return (AudioFlingerDeviceHandle)audiodev;
-}
-
-AudioFlingerDeviceHandle audioflinger_device_open(void* audio_sink)
-{
- AudioFlingerDevice* audiodev = NULL;
-
- // audio_sink shall be an MediaPlayerBase::AudioSink instance
- if(audio_sink == NULL)
- return NULL;
-
- // create a new instance of AudioFlinger
- audiodev = new AudioFlingerDevice;
- if (audiodev == NULL) {
- LOGE("Error to create AudioFlingerDevice\n");
- return NULL;
- }
-
- // set AudioSink
-
- audiodev->audio_sink = (MediaPlayerBase::AudioSink*)audio_sink;
- audiodev->audio_track = NULL;
- audiodev->init = false;
- audiodev->audio_sink_specified = true;
- LOGD("Open AudioSink successfully : %p\n",audiodev);
-
- return (AudioFlingerDeviceHandle)audiodev;
-}
-
-int audioflinger_device_set (AudioFlingerDeviceHandle handle,
- int streamType, int channelCount, uint32_t sampleRate, int bufferCount)
-{
- status_t status = NO_ERROR;
-#ifndef STECONF_ANDROID_VERSION_DONUT
- uint32_t channels = 0;
-#endif
-
- int format = AudioSystem::PCM_16_BIT;
-
- if (handle == NULL)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- // bufferCount is not the number of internal buffer, but the internal
- // buffer size
-#ifdef STECONF_ANDROID_VERSION_DONUT
- status = AUDIO_FLINGER_DEVICE_TRACK(handle)->set(streamType, sampleRate,
- format, channelCount);
- LOGD("SET : handle : %p : Set AudioTrack, status: %d, streamType: %d, sampleRate: %d, "
- "channelCount: %d, bufferCount: %d\n",handle, status, streamType, sampleRate,
- channelCount, bufferCount);
-#else
- switch (channelCount)
- {
- case 1:
- channels = AudioSystem::CHANNEL_OUT_FRONT_LEFT;
- break;
- case 2:
- channels = AudioSystem::CHANNEL_OUT_STEREO;
- break;
- case 0:
- default:
- channels = 0;
- break;
- }
- status = AUDIO_FLINGER_DEVICE_TRACK(handle)->set(streamType, sampleRate,
- format, channels/*, bufferCount*/);
- LOGD("SET handle : %p : Set AudioTrack, status: %d, streamType: %d, sampleRate: %d, "
- "channelCount: %d(%d), bufferCount: %d\n",handle, status, streamType, sampleRate,
- channelCount, channels, bufferCount);
-#endif
- AUDIO_FLINGER_DEVICE_TRACK(handle)->setPositionUpdatePeriod(bufferCount);
-
- }
- else if(AUDIO_FLINGER_DEVICE_SINK(handle).get()) {
-#ifdef STECONF_ANDROID_VERSION_DONUT
- status = AUDIO_FLINGER_DEVICE_SINK(handle)->open(sampleRate, channelCount,
- format/*, bufferCount*/); //SDA
-
- LOGD("OPEN : handle : %p : Set AudioSink, status: %d, streamType: %d, sampleRate: %d,"
- "channelCount: %d, bufferCount: %d\n", handle, status, streamType, sampleRate,
- channelCount, bufferCount);
-#else
- channels = channelCount;
- status = AUDIO_FLINGER_DEVICE_SINK(handle)->open(sampleRate, channels,
- format/*, bufferCount*/);
- LOGD("OPEN handle : %p : Set AudioSink, status: %d, streamType: %d, sampleRate: %d,"
- "channelCount: %d(%d), bufferCount: %d\n", handle, status, streamType, sampleRate,
- channelCount, channels, bufferCount);
-#endif
- AUDIO_FLINGER_DEVICE_TRACK(handle) = (AudioTrack *)(AUDIO_FLINGER_DEVICE_SINK(handle)->getTrack());
- if(AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->setPositionUpdatePeriod(bufferCount);
- }
- }
-
- if (status != NO_ERROR)
- return -1;
-
- AUDIO_FLINGER_DEVICE(handle)->init = true;
-
- return 0;
-}
-
-void audioflinger_device_release (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL)
- return;
-
- LOGD("Enter\n");
- if(! AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified ) {
- if (AUDIO_FLINGER_DEVICE_TRACK(handle) ) {
- LOGD("handle : %p Release AudioTrack\n", handle);
- delete AUDIO_FLINGER_DEVICE_TRACK(handle);
- }
- }
- if (AUDIO_FLINGER_DEVICE_SINK(handle).get()) {
- LOGD("handle : %p Release AudioSink\n", handle);
- AUDIO_FLINGER_DEVICE_SINK(handle).clear();
- AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified = false;
- }
-
- delete AUDIO_FLINGER_DEVICE(handle);
-}
-
-
-void audioflinger_device_start (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Start Device\n", handle);
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- AUDIO_FLINGER_DEVICE_SINK(handle)->start();
- }
- else {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->start();
- }
-}
-
-void audioflinger_device_stop (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Stop Device\n", handle);
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- AUDIO_FLINGER_DEVICE_SINK(handle)->stop();
- }
- else {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->stop();
- }
-
-}
-
-void audioflinger_device_flush (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Flush device\n", handle);
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- AUDIO_FLINGER_DEVICE_SINK(handle)->flush();
- }
- else {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->flush();
- }
-}
-
-void audioflinger_device_pause (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Pause Device\n", handle);
-
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- AUDIO_FLINGER_DEVICE_SINK(handle)->pause();
- }
- else {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->pause();
- }
-
-}
-
-void audioflinger_device_mute (AudioFlingerDeviceHandle handle, int mute)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Mute Device\n", handle);
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- // do nothing here, because the volume/mute is set in media service layer
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->mute((bool)mute);
- }
-}
-
-int audioflinger_device_muted (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- // do nothing here, because the volume/mute is set in media service layer
- return -1;
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int) AUDIO_FLINGER_DEVICE_TRACK(handle)->muted ();
- }
- return -1;
-}
-
-
-void audioflinger_device_set_volume (AudioFlingerDeviceHandle handle, float left,
- float right)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return;
-
- LOGD("handle : %p Set volume Device %f,%f\n", handle,left,right);
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- // do nothing here, because the volume/mute is set in media service layer
- return ;
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- AUDIO_FLINGER_DEVICE_TRACK(handle)->setVolume (left, right);
- }
-}
-
-ssize_t audioflinger_device_write (AudioFlingerDeviceHandle handle, const void *buffer,
- size_t size)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- return AUDIO_FLINGER_DEVICE_SINK(handle)->write(buffer, size);
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return AUDIO_FLINGER_DEVICE_TRACK(handle)->write(buffer, size);
- }
-#ifndef STECONF_ANDROID_VERSION_DONUT
- return -1;
-#endif
-}
-
-int audioflinger_device_frameCount (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- return (int)AUDIO_FLINGER_DEVICE_SINK(handle)->frameCount();
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int)AUDIO_FLINGER_DEVICE_TRACK(handle)->frameCount();
- }
- return -1;
-}
-
-int audioflinger_device_frameSize (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- return (int)AUDIO_FLINGER_DEVICE_SINK(handle)->frameSize();
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int)AUDIO_FLINGER_DEVICE_TRACK(handle)->frameSize();
- }
-#ifndef STECONF_ANDROID_VERSION_DONUT
- return -1;
-#endif
-}
-
-int64_t audioflinger_device_latency (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- return (int64_t)AUDIO_FLINGER_DEVICE_SINK(handle)->latency();
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int64_t)AUDIO_FLINGER_DEVICE_TRACK(handle)->latency();
- }
- return -1;
-}
-
-int audioflinger_device_format (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
-
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- // do nothing here, MediaPlayerBase::AudioSink doesn't provide format()
- // interface
- return -1;
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int)AUDIO_FLINGER_DEVICE_TRACK(handle)->format();
- }
- return -1;
-}
-
-int audioflinger_device_channelCount (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return -1;
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- return (int)AUDIO_FLINGER_DEVICE_SINK(handle)->channelCount();
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int)AUDIO_FLINGER_DEVICE_TRACK(handle)->channelCount();
- }
- return -1;
-}
-
-uint32_t audioflinger_device_sampleRate (AudioFlingerDeviceHandle handle)
-{
- if (handle == NULL || AUDIO_FLINGER_DEVICE(handle)->init == false)
- return 0;
- if(AUDIO_FLINGER_DEVICE(handle)->audio_sink_specified) {
- // do nothing here, MediaPlayerBase::AudioSink doesn't provide sampleRate()
- // interface
- return -1;
- }
- else if (AUDIO_FLINGER_DEVICE_TRACK(handle)) {
- return (int)AUDIO_FLINGER_DEVICE_TRACK(handle)->getSampleRate();
-}
- return(-1);
-}
-
-int audioflinger_device_obtain_buffer (AudioFlingerDeviceHandle handle,
- void **buffer_handle, int8_t **data, size_t *samples, uint64_t offset)
-{
- AudioTrack *track = AUDIO_FLINGER_DEVICE_TRACK (handle);
- status_t res;
- AudioTrack::Buffer *audioBuffer;
-
- if(track == 0) return(-1);
- audioBuffer = new AudioTrack::Buffer();
- audioBuffer->frameCount = *samples;
- res = track->obtainBufferAtOffset (audioBuffer, offset, -1);
- if (res < 0) {
- delete audioBuffer;
-
- return (int) res;
- }
-
- *samples = audioBuffer->frameCount;
- *buffer_handle = static_cast<void *> (audioBuffer);
- *data = audioBuffer->i8;
-
- return res;
-}
-
-void audioflinger_device_release_buffer (AudioFlingerDeviceHandle handle,
- void *buffer_handle)
-{
- AudioTrack *track = AUDIO_FLINGER_DEVICE_TRACK (handle);
- AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(buffer_handle);
-
- if(track == 0) return;
-
- track->releaseBuffer (audioBuffer);
- delete audioBuffer;
-}
-
-uint32_t audioflinger_device_get_position (AudioFlingerDeviceHandle handle)
-{
- status_t status;
- uint32_t ret = -1;
- AudioTrack *track = AUDIO_FLINGER_DEVICE_TRACK (handle);
-
- if(track == 0) return(-1);
-
- status = track->getPosition (&ret);
-
- return ret;
-}
+++ /dev/null
-/* 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;
-}
projects / platform / upstream / gstreamer.git / commitdiff