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Remove audioflingersink

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Remove audioflingersink, it's in gst-android now.
This commit is contained in:
Reynaldo H. Verdejo Pinochet 2011-04-13 15:05:15 -04:00 committed by Tim-Philipp Müller
parent fb67a0d126
commit 290d598748
8 changed files with 0 additions and 2476 deletions
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1
Android.mk
View file

@ -111,7 +111,6 @@ CONFIGURE_TARGETS += gst-plugins-bad-configure
-include $(GST_PLUGINS_BAD_TOP)/gst/asfmux/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/videoparsers/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/ext/faad/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/sys/audioflingersink/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/sdp/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/hls/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/jp2kdecimator/Android.mk

69
sys/audioflingersink/Android.mk
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@ -1,69 +0,0 @@
# external/gstreamer/gstplayer/Android.mk
#
# Copyright 2009 STN wireless
#
#ifeq ($(USE_HARDWARE_MM),true)
LOCAL_PATH:= $(call my-dir)
# -------------------------------------
# gstaudioflinger library
#
include $(CLEAR_VARS)
LOCAL_ARM_MODE := arm
gstaudioflinger_FILES := \
audioflinger_wrapper.cpp \
gstaudioflingersink.c \
GstAndroid.cpp
LOCAL_SRC_FILES := $(gstaudioflinger_FILES)
LOCAL_C_INCLUDES = $(LOCAL_PATH) \
$(LOCAL_PATH)/include \
$(TOP)/frameworks/base
LOCAL_CFLAGS += -DHAVE_CONFIG_H
LOCAL_CFLAGS += -Wall -Wdeclaration-after-statement -g -O2
LOCAL_CFLAGS += -DANDROID_USE_GSTREAMER \
$(shell $(PKG_CONFIG) gstreamer-plugins-bad-0.10 --cflags) \
$(shell $(PKG_CONFIG) gstreamer-audio-0.10 --cflags)
ifeq ($(USE_AUDIO_PURE_CODEC),true)
LOCAL_CFLAGS += -DAUDIO_PURE_CODEC
endif
LOCAL_SHARED_LIBRARIES += libdl
LOCAL_SHARED_LIBRARIES += \
libgstreamer-0.10 \
libgstbase-0.10 \
libglib-2.0 \
libgthread-2.0 \
libgmodule-2.0 \
libgobject-2.0 \
libgstvideo-0.10 \
libgstaudio-0.10
LOCAL_LDFLAGS := -L$(SYSROOT)/usr/lib -llog
LOCAL_SHARED_LIBRARIES += \
libutils \
libcutils \
libui \
libhardware \
libandroid_runtime \
libmedia
LOCAL_MODULE:= libgstaudioflinger
LOCAL_MODULE_PATH := $(TARGET_OUT)/lib/gstreamer-0.10
#
# define LOCAL_PRELINK_MODULE to false to not use pre-link map
#
LOCAL_PRELINK_MODULE := false
LOCAL_MODULE_TAGS := eng debug
include $(BUILD_SHARED_LIBRARY)
#endif # USE_HARDWARE_MM == true

36
sys/audioflingersink/GstAndroid.cpp
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@ -1,36 +0,0 @@
#include <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <unistd.h>
#include <poll.h>
#include <sys/ioctl.h>
#include <string.h>
#include <sys/mman.h>
/* Helper functions */
#include <gst/gst.h>
/* Object header */
#include "gstaudioflingersink.h"
static gboolean plugin_init (GstPlugin * plugin)
{
gboolean ret = TRUE;
ret &= gst_audioflinger_sink_plugin_init (plugin);
return ret;
}
/* Version number of package */
#define VERSION "0.0.1"
/* package name */
#define PACKAGE "Android ST-ERICSSON"
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioflinger",
"Android audioflinger library for gstreamer",
plugin_init, VERSION, "LGPL", "libgstaudioflinger.so", "http://www.stericsson.com")

470
sys/audioflingersink/audioflinger_wrapper.cpp
View file

@ -1,470 +0,0 @@
/* 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;
}

85
sys/audioflingersink/audioflinger_wrapper.h
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@ -1,85 +0,0 @@
/* 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.
*/
/*
* This file defines APIs to convert C++ AudioFlinder/AudioTrack
* interface to C interface
*/
#ifndef __AUDIOFLINGER_WRAPPER_H__
#define __AUDIOFLINGER_WRAPPER_H__
#define LATE 0x80000002
#ifdef __cplusplus
extern "C" {
#endif
typedef void* AudioFlingerDeviceHandle;
AudioFlingerDeviceHandle audioflinger_device_create();
AudioFlingerDeviceHandle audioflinger_device_open(void* audio_sink);
int audioflinger_device_set (AudioFlingerDeviceHandle handle,
int streamType, int channelCount, uint32_t sampleRate, int bufferCount);
void audioflinger_device_release(AudioFlingerDeviceHandle handle);
void audioflinger_device_start(AudioFlingerDeviceHandle handle);
void audioflinger_device_stop(AudioFlingerDeviceHandle handle);
ssize_t audioflinger_device_write(AudioFlingerDeviceHandle handle,
const void* buffer, size_t size);
void audioflinger_device_flush(AudioFlingerDeviceHandle handle);
void audioflinger_device_pause(AudioFlingerDeviceHandle handle);
void audioflinger_device_mute(AudioFlingerDeviceHandle handle, int mute);
int audioflinger_device_muted(AudioFlingerDeviceHandle handle);
void audioflinger_device_set_volume(AudioFlingerDeviceHandle handle,
float left, float right);
int audioflinger_device_frameCount(AudioFlingerDeviceHandle handle);
int audioflinger_device_frameSize(AudioFlingerDeviceHandle handle);
int64_t audioflinger_device_latency(AudioFlingerDeviceHandle handle);
int audioflinger_device_format(AudioFlingerDeviceHandle handle);
int audioflinger_device_channelCount(AudioFlingerDeviceHandle handle);
uint32_t audioflinger_device_sampleRate(AudioFlingerDeviceHandle handle);
int audioflinger_device_obtain_buffer (AudioFlingerDeviceHandle handle,
void **buffer_handle, int8_t **data, size_t *samples, uint64_t offset);
void audioflinger_device_release_buffer (AudioFlingerDeviceHandle handle,
void *buffer_handle);
uint32_t audioflinger_device_get_position (AudioFlingerDeviceHandle handle);
#ifdef __cplusplus
}
#endif
#endif /* __AUDIOFLINGER_WRAPPER_H__ */

90
sys/audioflingersink/gstaudioflingerringbuffer.h
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@ -1,90 +0,0 @@
/* GStreamer
* Copyright (C) 2010 Alessandro Decina <alessandro.decina@collabora.co.uk>
*
* 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.
*/
#ifndef GST_AUDIO_FLINGER_RING_BUFFER_H
#define GST_AUDIO_FLINGER_RING_BUFFER_H
#include <string.h>
#include "gstaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
#define GST_CAT_DEFAULT gst_audio_sink_debug
#define GST_TYPE_AUDIORING_BUFFER \
(gst_audioringbuffer_get_type())
#define GST_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
#define GST_AUDIORING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_CAST(obj) \
((GstAudioRingBuffer *)obj)
#define GST_IS_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
#define GST_IS_AUDIORING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
#define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
#define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
#define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf)))
struct _GstAudioRingBuffer
{
GstRingBuffer object;
gboolean running;
gint queuedseg;
GCond *cond;
};
struct _GstAudioRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_dispose (GObject * object);
static void gst_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_pause (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_activate (GstRingBuffer * buf,
gboolean active);
GType gst_audioringbuffer_get_type (void);
#endif /* GST_AUDIO_FLINGER_RING_BUFFER_H */

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sys/audioflingersink/gstaudioflingersink.c
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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.
*/
#ifndef __GST_AUDIOFLINGERSINK_H__
#define __GST_AUDIOFLINGERSINK_H__
#include <gst/gst.h>
#include "gstaudiosink.h"
#include "audioflinger_wrapper.h"
G_BEGIN_DECLS
#define GST_TYPE_AUDIOFLINGERSINK (gst_audioflinger_sink_get_type())
#define GST_AUDIOFLINGERSINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOFLINGERSINK,GstAudioFlingerSink))
#define GST_AUDIOFLINGERSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOFLINGERSINK,GstAudioFlingerSinkClass))
#define GST_IS_AUDIOFLINGERSINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOFLINGERSINK))
#define GST_IS_AUDIOFLINGERSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFLINGERSINK))
typedef struct _GstAudioFlingerSink GstAudioFlingerSink;
typedef struct _GstAudioFlingerSinkClass GstAudioFlingerSinkClass;
struct _GstAudioFlingerSink {
GstAudioSink sink;
AudioFlingerDeviceHandle audioflinger_device;
gboolean m_init;
gint bytes_per_sample;
gdouble m_volume;
gboolean m_mute;
gpointer m_audiosink;
GstCaps *probed_caps;
gboolean eos;
GstClock *audio_clock;
GstClock *system_clock;
GstClock *system_audio_clock;
GstClock *exported_clock;
gboolean export_system_audio_clock;
gboolean may_provide_clock;
gboolean slaving_disabled;
guint64 last_resync_sample;
};
struct _GstAudioFlingerSinkClass {
GstAudioSinkClass parent_class;
};
GType gst_audioflinger_sink_get_type(void);
gboolean gst_audioflinger_sink_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_AUDIOFLINGERSINK_H__ */