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vlc/modules/audio_output/audiotrack.c

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/*****************************************************************************
* audiotrack.c: Android Java AudioTrack audio output module
*****************************************************************************
* Copyright © 2012-2015 VLC authors and VideoLAN, VideoLabs
*
* Authors: Thomas Guillem <thomas@gllm.fr>
* Ming Hu <tewilove@gmail.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <assert.h>
#include <jni.h>
#include <dlfcn.h>
#include <stdbool.h>
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
#include "../video_output/android/utils.h"
#define SMOOTHPOS_SAMPLE_COUNT 10
#define SMOOTHPOS_INTERVAL_US INT64_C(30000) // 30ms
#define AUDIOTIMESTAMP_INTERVAL_US INT64_C(500000) // 500ms
static int Open( vlc_object_t * );
static void Close( vlc_object_t * );
static void Stop( audio_output_t * );
static int Start( audio_output_t *, audio_sample_format_t * );
static void *AudioTrack_Thread( void * );
/* There is an undefined behavior when configuring AudioTrack with SPDIF or
* more than 2 channels when there is no HDMI out. It may succeed and the
* Android ressampler will be used to downmix to stereo. It may fails cleanly,
* and this module will be able to recover and fallback to stereo. Finally, in
* some rare cases, it may crash during init or while ressampling. Because of
* the last case we don't try up to 8 channels and we use AT_DEV_STEREO device
* per default */
enum at_dev {
AT_DEV_STEREO = 0,
AT_DEV_PCM,
AT_DEV_ENCODED,
};
#define AT_DEV_DEFAULT AT_DEV_STEREO
#define AT_DEV_MAX_CHANNELS 8
static const struct {
const char *id;
const char *name;
enum at_dev at_dev;
} at_devs[] = {
{ "stereo", "Up to 2 channels (compat mode).", AT_DEV_STEREO },
{ "pcm", "Up to 8 channels.", AT_DEV_PCM },
/* With "encoded", the module will try to play every audio codecs via
* passthrough.
*
* With "encoded:ENCODING_FLAGS_MASK", the module will try to play only
* codecs specified by ENCODING_FLAGS_MASK. This extra value is a long long
* that contains binary-shifted AudioFormat.ENCODING_* values. */
{ "encoded", "Up to 8 channels, passthrough if available.", AT_DEV_ENCODED },
{ NULL, NULL, AT_DEV_DEFAULT },
};
struct aout_sys_t {
/* sw gain */
float soft_gain;
bool soft_mute;
enum at_dev at_dev;
jobject p_audiotrack; /* AudioTrack ref */
audio_sample_format_t fmt; /* fmt setup by Start */
struct {
unsigned int i_rate;
int i_channel_config;
int i_format;
int i_size;
} audiotrack_args;
/* Used by AudioTrack_getPlaybackHeadPosition */
struct {
uint32_t i_wrap_count;
uint32_t i_last;
} headpos;
/* Used by AudioTrack_GetTimestampPositionUs */
struct {
jobject p_obj; /* AudioTimestamp ref */
jlong i_frame_us;
jlong i_frame_pos;
mtime_t i_play_time; /* time when play was called */
mtime_t i_last_time;
} timestamp;
/* Used by AudioTrack_GetSmoothPositionUs */
struct {
uint32_t i_idx;
uint32_t i_count;
mtime_t p_us[SMOOTHPOS_SAMPLE_COUNT];
mtime_t i_us;
mtime_t i_last_time;
mtime_t i_latency_us;
} smoothpos;
uint32_t i_max_audiotrack_samples;
long long i_encoding_flags;
bool b_passthrough;
uint8_t i_chans_to_reorder; /* do we need channel reordering */
uint8_t p_chan_table[AOUT_CHAN_MAX];
enum {
WRITE_BYTEARRAY,
WRITE_BYTEARRAYV23,
WRITE_SHORTARRAYV23,
WRITE_BYTEBUFFER,
WRITE_FLOATARRAY
} i_write_type;
vlc_thread_t thread; /* AudioTrack_Thread */
vlc_mutex_t lock;
vlc_cond_t aout_cond; /* cond owned by aout */
vlc_cond_t thread_cond; /* cond owned by AudioTrack_Thread */
/* These variables need locking on read and write */
bool b_thread_running; /* Set to false by aout to stop the thread */
bool b_thread_paused; /* If true, the thread won't process any data, see
* Pause() */
bool b_thread_waiting; /* If true, the thread is waiting for enough spaces
* in AudioTrack internal buffers */
uint64_t i_samples_written; /* Number of samples written since last flush */
bool b_audiotrack_exception; /* True if audiotrack threw an exception */
bool b_error; /* generic error */
struct {
uint64_t i_read; /* Number of bytes read */
uint64_t i_write; /* Number of bytes written */
size_t i_size; /* Size of the circular buffer in bytes */
union {
jbyteArray p_bytearray;
jfloatArray p_floatarray;
jshortArray p_shortarray;
struct {
uint8_t *p_data;
jobject p_obj;
} bytebuffer;
} u;
} circular;
};
/* Soft volume helper */
#include "audio_output/volume.h"
// Don't use Float for now since 5.1/7.1 Float is down sampled to Stereo Float
//#define AUDIOTRACK_USE_FLOAT
//#define AUDIOTRACK_HW_LATENCY
/* Get AudioTrack native sample rate: if activated, most of the resampling
* will be done by VLC */
#define AUDIOTRACK_NATIVE_SAMPLERATE
vlc_module_begin ()
set_shortname( "AudioTrack" )
set_description( "Android AudioTrack audio output" )
set_capability( "audio output", 180 )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_AOUT )
add_sw_gain()
add_shortcut( "audiotrack" )
set_callbacks( Open, Close )
vlc_module_end ()
#define THREAD_NAME "android_audiotrack"
#define GET_ENV() android_getEnv( VLC_OBJECT(p_aout), THREAD_NAME )
static struct
{
struct {
jclass clazz;
jmethodID ctor;
jmethodID release;
jmethodID getState;
jmethodID play;
jmethodID stop;
jmethodID flush;
jmethodID pause;
jmethodID write;
jmethodID writeV23;
jmethodID writeShortV23;
jmethodID writeBufferV21;
jmethodID writeFloat;
jmethodID getPlaybackHeadPosition;
jmethodID getTimestamp;
jmethodID getMinBufferSize;
jmethodID getNativeOutputSampleRate;
jint STATE_INITIALIZED;
jint MODE_STREAM;
jint ERROR;
jint ERROR_BAD_VALUE;
jint ERROR_INVALID_OPERATION;
jint WRITE_NON_BLOCKING;
} AudioTrack;
struct {
jint ENCODING_PCM_8BIT;
jint ENCODING_PCM_16BIT;
jint ENCODING_PCM_FLOAT;
bool has_ENCODING_PCM_FLOAT;
jint ENCODING_AC3;
bool has_ENCODING_AC3;
jint ENCODING_E_AC3;
bool has_ENCODING_E_AC3;
jint ENCODING_DOLBY_TRUEHD;
bool has_ENCODING_DOLBY_TRUEHD;
jint ENCODING_DTS;
bool has_ENCODING_DTS;
jint ENCODING_DTS_HD;
bool has_ENCODING_DTS_HD;
jint ENCODING_IEC61937;
bool has_ENCODING_IEC61937;
jint CHANNEL_OUT_MONO;
jint CHANNEL_OUT_STEREO;
jint CHANNEL_OUT_FRONT_LEFT;
jint CHANNEL_OUT_FRONT_RIGHT;
jint CHANNEL_OUT_BACK_LEFT;
jint CHANNEL_OUT_BACK_RIGHT;
jint CHANNEL_OUT_FRONT_CENTER;
jint CHANNEL_OUT_LOW_FREQUENCY;
jint CHANNEL_OUT_BACK_CENTER;
jint CHANNEL_OUT_5POINT1;
jint CHANNEL_OUT_SIDE_LEFT;
jint CHANNEL_OUT_SIDE_RIGHT;
bool has_CHANNEL_OUT_SIDE;
} AudioFormat;
struct {
jint ERROR_DEAD_OBJECT;
bool has_ERROR_DEAD_OBJECT;
jint STREAM_MUSIC;
} AudioManager;
struct {
jclass clazz;
jmethodID getOutputLatency;
} AudioSystem;
struct {
jclass clazz;
jmethodID ctor;
jfieldID framePosition;
jfieldID nanoTime;
} AudioTimestamp;
} jfields;
/* init all jni fields.
* Done only one time during the first initialisation */
static bool
InitJNIFields( audio_output_t *p_aout, JNIEnv* env )
{
static vlc_mutex_t lock = VLC_STATIC_MUTEX;
static int i_init_state = -1;
bool ret;
jclass clazz;
jfieldID field;
vlc_mutex_lock( &lock );
if( i_init_state != -1 )
goto end;
#define CHECK_EXCEPTION( what, critical ) do { \
if( (*env)->ExceptionCheck( env ) ) \
{ \
msg_Err( p_aout, "%s failed", what ); \
(*env)->ExceptionClear( env ); \
if( (critical) ) \
{ \
i_init_state = 0; \
goto end; \
} \
} \
} while( 0 )
#define GET_CLASS( str, critical ) do { \
clazz = (*env)->FindClass( env, (str) ); \
CHECK_EXCEPTION( "FindClass(" str ")", critical ); \
} while( 0 )
#define GET_ID( get, id, str, args, critical ) do { \
jfields.id = (*env)->get( env, clazz, (str), (args) ); \
CHECK_EXCEPTION( #get "(" #id ")", critical ); \
} while( 0 )
#define GET_CONST_INT( id, str, critical ) do { \
field = NULL; \
field = (*env)->GetStaticFieldID( env, clazz, (str), "I" ); \
CHECK_EXCEPTION( "GetStaticFieldID(" #id ")", critical ); \
if( field ) \
{ \
jfields.id = (*env)->GetStaticIntField( env, clazz, field ); \
CHECK_EXCEPTION( #id, critical ); \
} \
} while( 0 )
/* AudioTrack class init */
GET_CLASS( "android/media/AudioTrack", true );
jfields.AudioTrack.clazz = (jclass) (*env)->NewGlobalRef( env, clazz );
CHECK_EXCEPTION( "NewGlobalRef", true );
GET_ID( GetMethodID, AudioTrack.ctor, "<init>", "(IIIIII)V", true );
GET_ID( GetMethodID, AudioTrack.release, "release", "()V", true );
GET_ID( GetMethodID, AudioTrack.getState, "getState", "()I", true );
GET_ID( GetMethodID, AudioTrack.play, "play", "()V", true );
GET_ID( GetMethodID, AudioTrack.stop, "stop", "()V", true );
GET_ID( GetMethodID, AudioTrack.flush, "flush", "()V", true );
GET_ID( GetMethodID, AudioTrack.pause, "pause", "()V", true );
GET_ID( GetMethodID, AudioTrack.writeV23, "write", "([BIII)I", false );
GET_ID( GetMethodID, AudioTrack.writeShortV23, "write", "([SIII)I", false );
if( !jfields.AudioTrack.writeV23 )
GET_ID( GetMethodID, AudioTrack.writeBufferV21, "write", "(Ljava/nio/ByteBuffer;II)I", false );
if( jfields.AudioTrack.writeV23 || jfields.AudioTrack.writeBufferV21 )
{
GET_CONST_INT( AudioTrack.WRITE_NON_BLOCKING, "WRITE_NON_BLOCKING", true );
#ifdef AUDIOTRACK_USE_FLOAT
GET_ID( GetMethodID, AudioTrack.writeFloat, "write", "([FIII)I", true );
#endif
} else
GET_ID( GetMethodID, AudioTrack.write, "write", "([BII)I", true );
GET_ID( GetMethodID, AudioTrack.getTimestamp,
"getTimestamp", "(Landroid/media/AudioTimestamp;)Z", false );
GET_ID( GetMethodID, AudioTrack.getPlaybackHeadPosition,
"getPlaybackHeadPosition", "()I", true );
GET_ID( GetStaticMethodID, AudioTrack.getMinBufferSize, "getMinBufferSize",
"(III)I", true );
#ifdef AUDIOTRACK_NATIVE_SAMPLERATE
GET_ID( GetStaticMethodID, AudioTrack.getNativeOutputSampleRate,
"getNativeOutputSampleRate", "(I)I", true );
#endif
GET_CONST_INT( AudioTrack.STATE_INITIALIZED, "STATE_INITIALIZED", true );
GET_CONST_INT( AudioTrack.MODE_STREAM, "MODE_STREAM", true );
GET_CONST_INT( AudioTrack.ERROR, "ERROR", true );
GET_CONST_INT( AudioTrack.ERROR_BAD_VALUE , "ERROR_BAD_VALUE", true );
GET_CONST_INT( AudioTrack.ERROR_INVALID_OPERATION,
"ERROR_INVALID_OPERATION", true );
/* AudioTimestamp class init (if any) */
if( jfields.AudioTrack.getTimestamp )
{
GET_CLASS( "android/media/AudioTimestamp", true );
jfields.AudioTimestamp.clazz = (jclass) (*env)->NewGlobalRef( env,
clazz );
CHECK_EXCEPTION( "NewGlobalRef", true );
GET_ID( GetMethodID, AudioTimestamp.ctor, "<init>", "()V", true );
GET_ID( GetFieldID, AudioTimestamp.framePosition,
"framePosition", "J", true );
GET_ID( GetFieldID, AudioTimestamp.nanoTime,
"nanoTime", "J", true );
}
#ifdef AUDIOTRACK_HW_LATENCY
/* AudioSystem class init */
GET_CLASS( "android/media/AudioSystem", false );
if( clazz )
{
jfields.AudioSystem.clazz = (jclass) (*env)->NewGlobalRef( env, clazz );
GET_ID( GetStaticMethodID, AudioSystem.getOutputLatency,
"getOutputLatency", "(I)I", false );
}
#endif
/* AudioFormat class init */
GET_CLASS( "android/media/AudioFormat", true );
GET_CONST_INT( AudioFormat.ENCODING_PCM_8BIT, "ENCODING_PCM_8BIT", true );
GET_CONST_INT( AudioFormat.ENCODING_PCM_16BIT, "ENCODING_PCM_16BIT", true );
#ifdef AUDIOTRACK_USE_FLOAT
GET_CONST_INT( AudioFormat.ENCODING_PCM_FLOAT, "ENCODING_PCM_FLOAT",
false );
jfields.AudioFormat.has_ENCODING_PCM_FLOAT = field != NULL &&
jfields.AudioTrack.writeFloat;
#else
jfields.AudioFormat.has_ENCODING_PCM_FLOAT = false;
#endif
if( jfields.AudioTrack.writeShortV23 )
{
GET_CONST_INT( AudioFormat.ENCODING_IEC61937, "ENCODING_IEC61937", false );
jfields.AudioFormat.has_ENCODING_IEC61937 = field != NULL;
}
else
jfields.AudioFormat.has_ENCODING_IEC61937 = false;
GET_CONST_INT( AudioFormat.ENCODING_AC3, "ENCODING_AC3", false );
jfields.AudioFormat.has_ENCODING_AC3 = field != NULL;
GET_CONST_INT( AudioFormat.ENCODING_E_AC3, "ENCODING_E_AC3", false );
jfields.AudioFormat.has_ENCODING_E_AC3 = field != NULL;
GET_CONST_INT( AudioFormat.ENCODING_DTS, "ENCODING_DTS", false );
jfields.AudioFormat.has_ENCODING_DTS = field != NULL;
GET_CONST_INT( AudioFormat.ENCODING_DTS_HD, "ENCODING_DTS_HD", false );
jfields.AudioFormat.has_ENCODING_DTS_HD = field != NULL;
GET_CONST_INT( AudioFormat.ENCODING_DOLBY_TRUEHD, "ENCODING_DOLBY_TRUEHD",
false );
jfields.AudioFormat.has_ENCODING_DOLBY_TRUEHD = field != NULL;
GET_CONST_INT( AudioFormat.CHANNEL_OUT_MONO, "CHANNEL_OUT_MONO", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_STEREO, "CHANNEL_OUT_STEREO", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_FRONT_LEFT, "CHANNEL_OUT_FRONT_LEFT", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_FRONT_RIGHT, "CHANNEL_OUT_FRONT_RIGHT", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_5POINT1, "CHANNEL_OUT_5POINT1", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_BACK_LEFT, "CHANNEL_OUT_BACK_LEFT", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_BACK_RIGHT, "CHANNEL_OUT_BACK_RIGHT", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_FRONT_CENTER, "CHANNEL_OUT_FRONT_CENTER", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_LOW_FREQUENCY, "CHANNEL_OUT_LOW_FREQUENCY", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_BACK_CENTER, "CHANNEL_OUT_BACK_CENTER", true );
GET_CONST_INT( AudioFormat.CHANNEL_OUT_SIDE_LEFT, "CHANNEL_OUT_SIDE_LEFT", false );
if( field != NULL )
{
GET_CONST_INT( AudioFormat.CHANNEL_OUT_SIDE_RIGHT, "CHANNEL_OUT_SIDE_RIGHT", true );
jfields.AudioFormat.has_CHANNEL_OUT_SIDE = true;
} else
jfields.AudioFormat.has_CHANNEL_OUT_SIDE = false;
/* AudioManager class init */
GET_CLASS( "android/media/AudioManager", true );
GET_CONST_INT( AudioManager.ERROR_DEAD_OBJECT, "ERROR_DEAD_OBJECT", false );
jfields.AudioManager.has_ERROR_DEAD_OBJECT = field != NULL;
GET_CONST_INT( AudioManager.STREAM_MUSIC, "STREAM_MUSIC", true );
#undef CHECK_EXCEPTION
#undef GET_CLASS
#undef GET_ID
#undef GET_CONST_INT
i_init_state = 1;
end:
ret = i_init_state == 1;
if( !ret )
msg_Err( p_aout, "AudioTrack jni init failed" );
vlc_mutex_unlock( &lock );
return ret;
}
static inline bool
check_exception( JNIEnv *env, audio_output_t *p_aout,
const char *method )
{
if( (*env)->ExceptionCheck( env ) )
{
aout_sys_t *p_sys = p_aout->sys;
p_sys->b_audiotrack_exception = true;
p_sys->b_error = true;
(*env)->ExceptionDescribe( env );
(*env)->ExceptionClear( env );
msg_Err( p_aout, "AudioTrack.%s triggered an exception !", method );
return true;
} else
return false;
}
#define CHECK_AT_EXCEPTION( method ) check_exception( env, p_aout, method )
#define JNI_CALL( what, obj, method, ... ) (*env)->what( env, obj, method, ##__VA_ARGS__ )
#define JNI_CALL_INT( obj, method, ... ) JNI_CALL( CallIntMethod, obj, method, ##__VA_ARGS__ )
#define JNI_CALL_BOOL( obj, method, ... ) JNI_CALL( CallBooleanMethod, obj, method, ##__VA_ARGS__ )
#define JNI_CALL_VOID( obj, method, ... ) JNI_CALL( CallVoidMethod, obj, method, ##__VA_ARGS__ )
#define JNI_CALL_STATIC_INT( clazz, method, ... ) JNI_CALL( CallStaticIntMethod, clazz, method, ##__VA_ARGS__ )
#define JNI_AT_NEW( ... ) JNI_CALL( NewObject, jfields.AudioTrack.clazz, jfields.AudioTrack.ctor, ##__VA_ARGS__ )
#define JNI_AT_CALL_INT( method, ... ) JNI_CALL_INT( p_sys->p_audiotrack, jfields.AudioTrack.method, ##__VA_ARGS__ )
#define JNI_AT_CALL_BOOL( method, ... ) JNI_CALL_BOOL( p_sys->p_audiotrack, jfields.AudioTrack.method, ##__VA_ARGS__ )
#define JNI_AT_CALL_VOID( method, ... ) JNI_CALL_VOID( p_sys->p_audiotrack, jfields.AudioTrack.method, ##__VA_ARGS__ )
#define JNI_AT_CALL_STATIC_INT( method, ... ) JNI_CALL( CallStaticIntMethod, jfields.AudioTrack.clazz, jfields.AudioTrack.method, ##__VA_ARGS__ )
#define JNI_AUDIOTIMESTAMP_GET_LONG( field ) JNI_CALL( GetLongField, p_sys->timestamp.p_obj, jfields.AudioTimestamp.field )
static inline mtime_t
frames_to_us( aout_sys_t *p_sys, uint64_t i_nb_frames )
{
return i_nb_frames * CLOCK_FREQ / p_sys->fmt.i_rate;
}
#define FRAMES_TO_US(x) frames_to_us( p_sys, (x) )
static inline uint64_t
bytes_to_frames( aout_sys_t *p_sys, size_t i_bytes )
{
return i_bytes * p_sys->fmt.i_frame_length / p_sys->fmt.i_bytes_per_frame;
}
#define BYTES_TO_FRAMES(x) bytes_to_frames( p_sys, (x) )
#define BYTES_TO_US(x) frames_to_us( p_sys, bytes_to_frames( p_sys, (x) ) )
static inline size_t
frames_to_bytes( aout_sys_t *p_sys, uint64_t i_frames )
{
return i_frames * p_sys->fmt.i_bytes_per_frame / p_sys->fmt.i_frame_length;
}
#define FRAMES_TO_BYTES(x) frames_to_bytes( p_sys, (x) )
/**
* Get the AudioTrack position
*
* The doc says that the position is reset to zero after flush but it's not
* true for all devices or Android versions.
*/
static uint64_t
AudioTrack_getPlaybackHeadPosition( JNIEnv *env, audio_output_t *p_aout )
{
/* Android doc:
* getPlaybackHeadPosition: Returns the playback head position expressed in
* frames. Though the "int" type is signed 32-bits, the value should be
* reinterpreted as if it is unsigned 32-bits. That is, the next position
* after 0x7FFFFFFF is (int) 0x80000000. This is a continuously advancing
* counter. It will wrap (overflow) periodically, for example approximately
* once every 27:03:11 hours:minutes:seconds at 44.1 kHz. It is reset to
* zero by flush(), reload(), and stop().
*/
aout_sys_t *p_sys = p_aout->sys;
uint32_t i_pos;
/* int32_t to uint32_t */
i_pos = 0xFFFFFFFFL & JNI_AT_CALL_INT( getPlaybackHeadPosition );
/* uint32_t to uint64_t */
if( p_sys->headpos.i_last > i_pos )
p_sys->headpos.i_wrap_count++;
p_sys->headpos.i_last = i_pos;
return p_sys->headpos.i_last + ((uint64_t)p_sys->headpos.i_wrap_count << 32);
}
/**
* Reset AudioTrack position
*
* Called after flush, or start
*/
static void
AudioTrack_ResetPlaybackHeadPosition( JNIEnv *env, audio_output_t *p_aout )
{
(void) env;
aout_sys_t *p_sys = p_aout->sys;
p_sys->headpos.i_last = 0;
p_sys->headpos.i_wrap_count = 0;
}
/**
* Reset AudioTrack SmoothPosition and TimestampPosition
*/
static void
AudioTrack_ResetPositions( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
VLC_UNUSED( env );
p_sys->timestamp.i_play_time = mdate();
p_sys->timestamp.i_last_time = 0;
p_sys->timestamp.i_frame_us = 0;
p_sys->timestamp.i_frame_pos = 0;
p_sys->smoothpos.i_count = 0;
p_sys->smoothpos.i_idx = 0;
p_sys->smoothpos.i_last_time = 0;
p_sys->smoothpos.i_us = 0;
p_sys->smoothpos.i_latency_us = 0;
}
/**
* Reset all AudioTrack positions and internal state
*/
static void
AudioTrack_Reset( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
AudioTrack_ResetPositions( env, p_aout );
AudioTrack_ResetPlaybackHeadPosition( env, p_aout );
p_sys->i_samples_written = 0;
}
/**
* Get a smooth AudioTrack position
*
* This function smooth out the AudioTrack position since it has a very bad
* precision (+/- 20ms on old devices).
*/
static mtime_t
AudioTrack_GetSmoothPositionUs( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
uint64_t i_audiotrack_us;
mtime_t i_now = mdate();
/* Fetch an AudioTrack position every SMOOTHPOS_INTERVAL_US (30ms) */
if( i_now - p_sys->smoothpos.i_last_time >= SMOOTHPOS_INTERVAL_US )
{
i_audiotrack_us = FRAMES_TO_US( AudioTrack_getPlaybackHeadPosition( env, p_aout ) );
p_sys->smoothpos.i_last_time = i_now;
/* Base the position off the current time */
p_sys->smoothpos.p_us[p_sys->smoothpos.i_idx] = i_audiotrack_us - i_now;
p_sys->smoothpos.i_idx = (p_sys->smoothpos.i_idx + 1)
% SMOOTHPOS_SAMPLE_COUNT;
if( p_sys->smoothpos.i_count < SMOOTHPOS_SAMPLE_COUNT )
p_sys->smoothpos.i_count++;
/* Calculate the average position based off the current time */
p_sys->smoothpos.i_us = 0;
for( uint32_t i = 0; i < p_sys->smoothpos.i_count; ++i )
p_sys->smoothpos.i_us += p_sys->smoothpos.p_us[i];
p_sys->smoothpos.i_us /= p_sys->smoothpos.i_count;
if( jfields.AudioSystem.getOutputLatency )
{
int i_latency_ms = JNI_CALL( CallStaticIntMethod,
jfields.AudioSystem.clazz,
jfields.AudioSystem.getOutputLatency,
jfields.AudioManager.STREAM_MUSIC );
p_sys->smoothpos.i_latency_us = i_latency_ms > 0 ?
i_latency_ms * 1000L : 0;
}
}
if( p_sys->smoothpos.i_us != 0 )
return p_sys->smoothpos.i_us + i_now - p_sys->smoothpos.i_latency_us;
else
return 0;
}
static mtime_t
AudioTrack_GetTimestampPositionUs( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
mtime_t i_now;
if( !p_sys->timestamp.p_obj )
return 0;
i_now = mdate();
/* Android doc:
* getTimestamp: Poll for a timestamp on demand.
*
* If you need to track timestamps during initial warmup or after a
* routing or mode change, you should request a new timestamp once per
* second until the reported timestamps show that the audio clock is
* stable. Thereafter, query for a new timestamp approximately once
* every 10 seconds to once per minute. Calling this method more often
* is inefficient. It is also counter-productive to call this method
* more often than recommended, because the short-term differences
* between successive timestamp reports are not meaningful. If you need
* a high-resolution mapping between frame position and presentation
* time, consider implementing that at application level, based on
* low-resolution timestamps.
*/
/* Fetch an AudioTrack timestamp every AUDIOTIMESTAMP_INTERVAL_US (500ms) */
if( i_now - p_sys->timestamp.i_last_time >= AUDIOTIMESTAMP_INTERVAL_US )
{
p_sys->timestamp.i_last_time = i_now;
if( JNI_AT_CALL_BOOL( getTimestamp, p_sys->timestamp.p_obj ) )
{
p_sys->timestamp.i_frame_us = JNI_AUDIOTIMESTAMP_GET_LONG( nanoTime ) / 1000;
p_sys->timestamp.i_frame_pos = JNI_AUDIOTIMESTAMP_GET_LONG( framePosition );
}
else
{
p_sys->timestamp.i_frame_us = 0;
p_sys->timestamp.i_frame_pos = 0;
}
}
/* frame time should be after last play time
* frame time shouldn't be in the future
* frame time should be less than 10 seconds old */
if( p_sys->timestamp.i_frame_us != 0 && p_sys->timestamp.i_frame_pos != 0
&& p_sys->timestamp.i_frame_us > p_sys->timestamp.i_play_time
&& i_now > p_sys->timestamp.i_frame_us
&& ( i_now - p_sys->timestamp.i_frame_us ) <= INT64_C(10000000) )
{
jlong i_time_diff = i_now - p_sys->timestamp.i_frame_us;
jlong i_frames_diff = i_time_diff * p_sys->fmt.i_rate / CLOCK_FREQ;
return FRAMES_TO_US( p_sys->timestamp.i_frame_pos + i_frames_diff );
} else
return 0;
}
static int
TimeGet( audio_output_t *p_aout, mtime_t *restrict p_delay )
{
aout_sys_t *p_sys = p_aout->sys;
mtime_t i_audiotrack_us;
JNIEnv *env;
if( p_sys->b_passthrough )
return -1;
vlc_mutex_lock( &p_sys->lock );
if( p_sys->b_error || !p_sys->i_samples_written || !( env = GET_ENV() ) )
goto bailout;
i_audiotrack_us = AudioTrack_GetTimestampPositionUs( env, p_aout );
if( i_audiotrack_us <= 0 )
i_audiotrack_us = AudioTrack_GetSmoothPositionUs(env, p_aout );
/* Debug log for both delays */
#if 0
{
mtime_t i_written_us = FRAMES_TO_US( p_sys->i_samples_written );
mtime_t i_ts_us = AudioTrack_GetTimestampPositionUs( env, p_aout );
mtime_t i_smooth_us = 0;
if( i_ts_us > 0 )
i_smooth_us = AudioTrack_GetSmoothPositionUs(env, p_aout );
else if ( p_sys->smoothpos.i_us != 0 )
i_smooth_us = p_sys->smoothpos.i_us + mdate()
- p_sys->smoothpos.i_latency_us;
msg_Err( p_aout, "TimeGet: TimeStamp: %lld, Smooth: %lld (latency: %lld)",
i_ts_us ? i_written_us - i_ts_us : 0,
i_smooth_us ? i_written_us - i_smooth_us : 0,
p_sys->smoothpos.i_latency_us );
}
#endif
if( i_audiotrack_us > 0 )
{
/* AudioTrack delay */
mtime_t i_delay = FRAMES_TO_US( p_sys->i_samples_written )
- i_audiotrack_us;
if( i_delay >= 0 )
{
/* Circular buffer delay */
i_delay += BYTES_TO_US( p_sys->circular.i_write
- p_sys->circular.i_read );
*p_delay = i_delay;
vlc_mutex_unlock( &p_sys->lock );
return 0;
}
else
{
msg_Warn( p_aout, "timing screwed, reset positions" );
AudioTrack_ResetPositions( env, p_aout );
}
}
bailout:
vlc_mutex_unlock( &p_sys->lock );
return -1;
}
static void
AudioTrack_GetChanOrder( uint16_t i_physical_channels, uint32_t p_chans_out[] )
{
#define HAS_CHAN( x ) ( ( i_physical_channels & (x) ) == (x) )
/* samples will be in the following order: FL FR FC LFE BL BR BC SL SR */
int i = 0;
if( HAS_CHAN( AOUT_CHAN_LEFT ) )
p_chans_out[i++] = AOUT_CHAN_LEFT;
if( HAS_CHAN( AOUT_CHAN_RIGHT ) )
p_chans_out[i++] = AOUT_CHAN_RIGHT;
if( HAS_CHAN( AOUT_CHAN_CENTER ) )
p_chans_out[i++] = AOUT_CHAN_CENTER;
if( HAS_CHAN( AOUT_CHAN_LFE ) )
p_chans_out[i++] = AOUT_CHAN_LFE;
if( HAS_CHAN( AOUT_CHAN_REARLEFT ) )
p_chans_out[i++] = AOUT_CHAN_REARLEFT;
if( HAS_CHAN( AOUT_CHAN_REARRIGHT ) )
p_chans_out[i++] = AOUT_CHAN_REARRIGHT;
if( HAS_CHAN( AOUT_CHAN_REARCENTER ) )
p_chans_out[i++] = AOUT_CHAN_REARCENTER;
if( HAS_CHAN( AOUT_CHAN_MIDDLELEFT ) )
p_chans_out[i++] = AOUT_CHAN_MIDDLELEFT;
if( HAS_CHAN( AOUT_CHAN_MIDDLERIGHT ) )
p_chans_out[i++] = AOUT_CHAN_MIDDLERIGHT;
assert( i <= AOUT_CHAN_MAX );
#undef HAS_CHAN
}
/**
* Create an Android AudioTrack.
* returns -1 on error, 0 on success.
*/
static int
AudioTrack_New( JNIEnv *env, audio_output_t *p_aout, unsigned int i_rate,
int i_channel_config, int i_format, int i_size )
{
aout_sys_t *p_sys = p_aout->sys;
jobject p_audiotrack = JNI_AT_NEW( jfields.AudioManager.STREAM_MUSIC,
i_rate, i_channel_config, i_format,
i_size, jfields.AudioTrack.MODE_STREAM );
if( CHECK_AT_EXCEPTION( "AudioTrack<init>" ) || !p_audiotrack )
{
msg_Warn( p_aout, "AudioTrack Init failed" ) ;
return -1;
}
if( JNI_CALL_INT( p_audiotrack, jfields.AudioTrack.getState )
!= jfields.AudioTrack.STATE_INITIALIZED )
{
JNI_CALL_VOID( p_audiotrack, jfields.AudioTrack.release );
(*env)->DeleteLocalRef( env, p_audiotrack );
msg_Err( p_aout, "AudioTrack getState failed" );
return -1;
}
p_sys->p_audiotrack = (*env)->NewGlobalRef( env, p_audiotrack );
(*env)->DeleteLocalRef( env, p_audiotrack );
if( !p_sys->p_audiotrack )
return -1;
return 0;
}
/**
* Destroy and recreate an Android AudioTrack using the same arguments.
* returns -1 on error, 0 on success.
*/
static int
AudioTrack_Recreate( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
JNI_AT_CALL_VOID( release );
(*env)->DeleteGlobalRef( env, p_sys->p_audiotrack );
p_sys->p_audiotrack = NULL;
return AudioTrack_New( env, p_aout, p_sys->audiotrack_args.i_rate,
p_sys->audiotrack_args.i_channel_config,
p_sys->audiotrack_args.i_format,
p_sys->audiotrack_args.i_size );
}
/**
* Configure and create an Android AudioTrack.
* returns -1 on configuration error, 0 on success.
*/
static int
AudioTrack_Create( JNIEnv *env, audio_output_t *p_aout,
unsigned int i_rate,
int i_format,
uint16_t i_physical_channels )
{
aout_sys_t *p_sys = p_aout->sys;
int i_size, i_min_buffer_size, i_channel_config;
switch( i_physical_channels )
{
case AOUT_CHANS_7_1:
/* bitmask of CHANNEL_OUT_7POINT1 doesn't correspond to 5POINT1 and
* SIDES */
i_channel_config = jfields.AudioFormat.CHANNEL_OUT_5POINT1 |
jfields.AudioFormat.CHANNEL_OUT_SIDE_LEFT |
jfields.AudioFormat.CHANNEL_OUT_SIDE_RIGHT;
break;
case AOUT_CHANS_5_1:
i_channel_config = jfields.AudioFormat.CHANNEL_OUT_5POINT1;
break;
case AOUT_CHAN_LEFT:
i_channel_config = jfields.AudioFormat.CHANNEL_OUT_MONO;
break;
case AOUT_CHANS_STEREO:
i_channel_config = jfields.AudioFormat.CHANNEL_OUT_STEREO;
break;
default:
vlc_assert_unreachable();
}
i_min_buffer_size = JNI_AT_CALL_STATIC_INT( getMinBufferSize, i_rate,
i_channel_config, i_format );
if( i_min_buffer_size <= 0 )
{
msg_Warn( p_aout, "getMinBufferSize returned an invalid size" ) ;
return -1;
}
i_size = i_min_buffer_size * 2;
/* create AudioTrack object */
if( AudioTrack_New( env, p_aout, i_rate, i_channel_config,
i_format , i_size ) != 0 )
return -1;
p_sys->audiotrack_args.i_rate = i_rate;
p_sys->audiotrack_args.i_channel_config = i_channel_config;
p_sys->audiotrack_args.i_format = i_format;
p_sys->audiotrack_args.i_size = i_size;
return 0;
}
static bool
AudioTrack_HasEncoding( audio_output_t *p_aout, vlc_fourcc_t i_format,
bool *p_dtshd )
{
aout_sys_t *p_sys = p_aout->sys;
#define MATCH_ENCODING_FLAG(x) jfields.AudioFormat.has_##x && \
( p_sys->i_encoding_flags == 0 || p_sys->i_encoding_flags & (1 << jfields.AudioFormat.x) )
*p_dtshd = false;
switch( i_format )
{
case VLC_CODEC_DTS:
if( MATCH_ENCODING_FLAG( ENCODING_DTS_HD ) )
{
*p_dtshd = true;
return true;
}
return MATCH_ENCODING_FLAG( ENCODING_DTS );
case VLC_CODEC_A52:
return MATCH_ENCODING_FLAG( ENCODING_AC3 );
case VLC_CODEC_EAC3:
return MATCH_ENCODING_FLAG( ENCODING_E_AC3 );
case VLC_CODEC_TRUEHD:
case VLC_CODEC_MLP:
return MATCH_ENCODING_FLAG( ENCODING_DOLBY_TRUEHD );
default:
return false;
}
}
static int
StartPassthrough( JNIEnv *env, audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
int i_at_format;
if( jfields.AudioFormat.has_ENCODING_IEC61937 )
{
bool b_dtshd;
if( !AudioTrack_HasEncoding( p_aout, p_sys->fmt.i_format, &b_dtshd ) )
return VLC_EGENERIC;
i_at_format = jfields.AudioFormat.ENCODING_IEC61937;
switch( p_sys->fmt.i_format )
{
case VLC_CODEC_TRUEHD:
case VLC_CODEC_MLP:
p_sys->fmt.i_rate = 192000;
p_sys->fmt.i_bytes_per_frame = 16;
/* AudioFormat.ENCODING_IEC61937 documentation says that the
* channel layout must be stereo. Well, not for TrueHD
* apparently */
p_sys->fmt.i_physical_channels = AOUT_CHANS_7_1;
break;
case VLC_CODEC_DTS:
if( b_dtshd && p_sys->fmt.i_rate >= 48000 )
{
p_sys->fmt.i_rate = 192000;
p_sys->fmt.i_bytes_per_frame = 16;
}
else
p_sys->fmt.i_bytes_per_frame = 4;
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
break;
case VLC_CODEC_EAC3:
p_sys->fmt.i_rate = 192000;
case VLC_CODEC_A52:
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
p_sys->fmt.i_bytes_per_frame = 4;
break;
default:
return VLC_EGENERIC;
}
p_sys->fmt.i_frame_length = 1;
p_sys->fmt.i_channels = aout_FormatNbChannels( &p_sys->fmt );
p_sys->fmt.i_format = VLC_CODEC_SPDIFL;
}
else
{
switch( p_sys->fmt.i_format )
{
case VLC_CODEC_A52:
if( !jfields.AudioFormat.has_ENCODING_AC3 )
return VLC_EGENERIC;
i_at_format = jfields.AudioFormat.ENCODING_AC3;
break;
case VLC_CODEC_DTS:
if( !jfields.AudioFormat.has_ENCODING_DTS )
return VLC_EGENERIC;
i_at_format = jfields.AudioFormat.ENCODING_DTS;
break;
default:
return VLC_EGENERIC;
}
p_sys->fmt.i_bytes_per_frame = 4;
p_sys->fmt.i_frame_length = 1;
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
p_sys->fmt.i_channels = 2;
p_sys->fmt.i_format = VLC_CODEC_SPDIFB;
}
int i_ret = AudioTrack_Create( env, p_aout, p_sys->fmt.i_rate, i_at_format,
p_sys->fmt.i_physical_channels );
if( i_ret != VLC_SUCCESS )
msg_Warn( p_aout, "SPDIF configuration failed" );
else
{
p_sys->b_passthrough = true;
p_sys->i_chans_to_reorder = 0;
}
return i_ret;
}
static int
StartPCM( JNIEnv *env, audio_output_t *p_aout, unsigned i_max_channels )
{
aout_sys_t *p_sys = p_aout->sys;
unsigned i_nb_channels;
int i_at_format, i_ret;
if (jfields.AudioTrack.getNativeOutputSampleRate)
p_sys->fmt.i_rate =
JNI_AT_CALL_STATIC_INT( getNativeOutputSampleRate,
jfields.AudioManager.STREAM_MUSIC );
else
p_sys->fmt.i_rate = VLC_CLIP( p_sys->fmt.i_rate, 4000, 48000 );
do
{
/* We can only accept U8, S16N, FL32, and AC3 */
switch( p_sys->fmt.i_format )
{
case VLC_CODEC_U8:
i_at_format = jfields.AudioFormat.ENCODING_PCM_8BIT;
break;
case VLC_CODEC_S16N:
i_at_format = jfields.AudioFormat.ENCODING_PCM_16BIT;
break;
case VLC_CODEC_FL32:
if( jfields.AudioFormat.has_ENCODING_PCM_FLOAT )
i_at_format = jfields.AudioFormat.ENCODING_PCM_FLOAT;
else
{
p_sys->fmt.i_format = VLC_CODEC_S16N;
i_at_format = jfields.AudioFormat.ENCODING_PCM_16BIT;
}
break;
default:
p_sys->fmt.i_format = VLC_CODEC_S16N;
i_at_format = jfields.AudioFormat.ENCODING_PCM_16BIT;
break;
}
/* Android AudioTrack supports only mono, stereo, 5.1 and 7.1.
* Android will downmix to stereo if audio output doesn't handle 5.1 or 7.1
*/
i_nb_channels = aout_FormatNbChannels( &p_sys->fmt );
if( i_nb_channels == 0 )
return VLC_EGENERIC;
if( AOUT_FMT_LINEAR( &p_sys->fmt ) )
i_nb_channels = __MIN( i_max_channels, i_nb_channels );
if( i_nb_channels > 5 )
{
if( i_nb_channels > 7 && jfields.AudioFormat.has_CHANNEL_OUT_SIDE )
p_sys->fmt.i_physical_channels = AOUT_CHANS_7_1;
else
p_sys->fmt.i_physical_channels = AOUT_CHANS_5_1;
} else
{
if( i_nb_channels == 1 )
p_sys->fmt.i_physical_channels = AOUT_CHAN_LEFT;
else
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
}
/* Try to create an AudioTrack with the most advanced channel and
* format configuration. If AudioTrack_Create fails, try again with a
* less advanced format (PCM S16N). If it fails again, try again with
* Stereo channels. */
i_ret = AudioTrack_Create( env, p_aout, p_sys->fmt.i_rate, i_at_format,
p_sys->fmt.i_physical_channels );
if( i_ret != 0 )
{
if( p_sys->fmt.i_format == VLC_CODEC_FL32 )
{
msg_Warn( p_aout, "FL32 configuration failed, "
"fallback to S16N PCM" );
p_sys->fmt.i_format = VLC_CODEC_S16N;
}
else if( p_sys->fmt.i_physical_channels & AOUT_CHANS_5_1 )
{
msg_Warn( p_aout, "5.1 or 7.1 configuration failed, "
"fallback to Stereo" );
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
}
else
break;
}
} while( i_ret != 0 );
if( i_ret != VLC_SUCCESS )
return i_ret;
uint32_t p_chans_out[AOUT_CHAN_MAX];
memset( p_chans_out, 0, sizeof(p_chans_out) );
AudioTrack_GetChanOrder( p_sys->fmt.i_physical_channels, p_chans_out );
p_sys->i_chans_to_reorder =
aout_CheckChannelReorder( NULL, p_chans_out,
p_sys->fmt.i_physical_channels,
p_sys->p_chan_table );
aout_FormatPrepare( &p_sys->fmt );
return VLC_SUCCESS;
}
static int
Start( audio_output_t *p_aout, audio_sample_format_t *restrict p_fmt )
{
aout_sys_t *p_sys = p_aout->sys;
JNIEnv *env;
int i_ret;
bool b_try_passthrough;
unsigned i_max_channels;
if( p_sys->at_dev == AT_DEV_ENCODED )
{
b_try_passthrough = true;
i_max_channels = AT_DEV_MAX_CHANNELS;
}
else
{
b_try_passthrough = var_InheritBool( p_aout, "spdif" );
i_max_channels = p_sys->at_dev == AT_DEV_STEREO ? 2 : AT_DEV_MAX_CHANNELS;
}
if( !( env = GET_ENV() ) )
return VLC_EGENERIC;
p_sys->fmt = *p_fmt;
aout_FormatPrint( p_aout, "VLC is looking for:", &p_sys->fmt );
bool low_latency = false;
if (p_sys->fmt.channel_type == AUDIO_CHANNEL_TYPE_AMBISONICS)
{
p_sys->fmt.channel_type = AUDIO_CHANNEL_TYPE_BITMAP;
/* TODO: detect sink channel layout */
p_sys->fmt.i_physical_channels = AOUT_CHANS_STEREO;
aout_FormatPrepare(&p_sys->fmt);
low_latency = true;
}
if( AOUT_FMT_LINEAR( &p_sys->fmt ) )
i_ret = StartPCM( env, p_aout, i_max_channels );
else if( b_try_passthrough )
i_ret = StartPassthrough( env, p_aout );
else
return VLC_EGENERIC;
if( i_ret != 0 )
return VLC_EGENERIC;
p_sys->i_max_audiotrack_samples = BYTES_TO_FRAMES( p_sys->audiotrack_args.i_size );
#ifdef AUDIOTRACK_HW_LATENCY
if( jfields.AudioTimestamp.clazz )
{
/* create AudioTimestamp object */
jobject p_obj = JNI_CALL( NewObject, jfields.AudioTimestamp.clazz,
jfields.AudioTimestamp.ctor );
if( p_obj )
{
p_sys->timestamp.p_obj = (*env)->NewGlobalRef( env, p_obj );
(*env)->DeleteLocalRef( env, p_obj );
}
if( !p_sys->timestamp.p_obj )
goto error;
}
#endif
AudioTrack_Reset( env, p_aout );
if( p_sys->fmt.i_format == VLC_CODEC_FL32 )
{
msg_Dbg( p_aout, "using WRITE_FLOATARRAY");
p_sys->i_write_type = WRITE_FLOATARRAY;
}
else if( p_sys->fmt.i_format == VLC_CODEC_SPDIFL )
{
assert( jfields.AudioFormat.has_ENCODING_IEC61937 );
msg_Dbg( p_aout, "using WRITE_SHORTARRAYV23");
p_sys->i_write_type = WRITE_SHORTARRAYV23;
}
else if( jfields.AudioTrack.writeV23 )
{
msg_Dbg( p_aout, "using WRITE_BYTEARRAYV23");
p_sys->i_write_type = WRITE_BYTEARRAYV23;
}
else if( jfields.AudioTrack.writeBufferV21 )
{
msg_Dbg( p_aout, "using WRITE_BYTEBUFFER");
p_sys->i_write_type = WRITE_BYTEBUFFER;
}
else
{
msg_Dbg( p_aout, "using WRITE_BYTEARRAY");
p_sys->i_write_type = WRITE_BYTEARRAY;
}
p_sys->circular.i_read = p_sys->circular.i_write = 0;
p_sys->circular.i_size = (int)p_sys->fmt.i_rate
* p_sys->fmt.i_bytes_per_frame
/ p_sys->fmt.i_frame_length;
if (low_latency)
{
/* 40 ms of buffering */
p_sys->circular.i_size = p_sys->circular.i_size / 25;
}
else
{
/* 2 seconds of buffering */
p_sys->circular.i_size = p_sys->circular.i_size * AOUT_MAX_PREPARE_TIME
/ CLOCK_FREQ;
}
/* Allocate circular buffer */
switch( p_sys->i_write_type )
{
case WRITE_BYTEARRAY:
case WRITE_BYTEARRAYV23:
{
jbyteArray p_bytearray;
p_bytearray = (*env)->NewByteArray( env, p_sys->circular.i_size );
if( p_bytearray )
{
p_sys->circular.u.p_bytearray = (*env)->NewGlobalRef( env, p_bytearray );
(*env)->DeleteLocalRef( env, p_bytearray );
}
if( !p_sys->circular.u.p_bytearray )
{
msg_Err(p_aout, "byte array allocation failed");
goto error;
}
break;
}
case WRITE_SHORTARRAYV23:
{
jshortArray p_shortarray;
p_shortarray = (*env)->NewShortArray( env,
p_sys->circular.i_size / 2 );
if( p_shortarray )
{
p_sys->circular.u.p_shortarray = (*env)->NewGlobalRef( env, p_shortarray );
(*env)->DeleteLocalRef( env, p_shortarray );
}
if( !p_sys->circular.u.p_shortarray )
{
msg_Err(p_aout, "short array allocation failed");
goto error;
}
break;
}
case WRITE_FLOATARRAY:
{
jfloatArray p_floatarray;
p_floatarray = (*env)->NewFloatArray( env,
p_sys->circular.i_size / 4 );
if( p_floatarray )
{
p_sys->circular.u.p_floatarray = (*env)->NewGlobalRef( env, p_floatarray );
(*env)->DeleteLocalRef( env, p_floatarray );
}
if( !p_sys->circular.u.p_floatarray )
{
msg_Err(p_aout, "float array allocation failed");
goto error;
}
break;
}
case WRITE_BYTEBUFFER:
p_sys->circular.u.bytebuffer.p_data = malloc( p_sys->circular.i_size );
if( !p_sys->circular.u.bytebuffer.p_data )
{
msg_Err(p_aout, "bytebuffer allocation failed");
goto error;
}
break;
}
/* Run AudioTrack_Thread */
p_sys->b_thread_running = true;
p_sys->b_thread_paused = false;
if ( vlc_clone( &p_sys->thread, AudioTrack_Thread, p_aout,
VLC_THREAD_PRIORITY_LOW ) )
{
msg_Err(p_aout, "vlc clone failed");
goto error;
}
JNI_AT_CALL_VOID( play );
CHECK_AT_EXCEPTION( "play" );
*p_fmt = p_sys->fmt;
aout_SoftVolumeStart( p_aout );
aout_FormatPrint( p_aout, "VLC will output:", &p_sys->fmt );
return VLC_SUCCESS;
error:
Stop( p_aout );
return VLC_EGENERIC;
}
static void
Stop( audio_output_t *p_aout )
{
aout_sys_t *p_sys = p_aout->sys;
JNIEnv *env;
if( !( env = GET_ENV() ) )
return;
/* Stop the AudioTrack thread */
vlc_mutex_lock( &p_sys->lock );
if( p_sys->b_thread_running )
{
p_sys->b_thread_running = false;
vlc_cond_signal( &p_sys->thread_cond );
vlc_mutex_unlock( &p_sys->lock );
vlc_join( p_sys->thread, NULL );
}
else
vlc_mutex_unlock( &p_sys->lock );
/* Release the AudioTrack object */
if( p_sys->p_audiotrack )
{
if( !p_sys->b_audiotrack_exception )
{
JNI_AT_CALL_VOID( stop );
if( !CHECK_AT_EXCEPTION( "stop" ) )
JNI_AT_CALL_VOID( release );
}
(*env)->DeleteGlobalRef( env, p_sys->p_audiotrack );
p_sys->p_audiotrack = NULL;
}
/* Release the timestamp object */
if( p_sys->timestamp.p_obj )
{
(*env)->DeleteGlobalRef( env, p_sys->timestamp.p_obj );
p_sys->timestamp.p_obj = NULL;
}
/* Release the Circular buffer data */
switch( p_sys->i_write_type )
{
case WRITE_BYTEARRAY:
case WRITE_BYTEARRAYV23:
if( p_sys->circular.u.p_bytearray )
{
(*env)->DeleteGlobalRef( env, p_sys->circular.u.p_bytearray );
p_sys->circular.u.p_bytearray = NULL;
}
break;
case WRITE_SHORTARRAYV23:
if( p_sys->circular.u.p_shortarray )
{
(*env)->DeleteGlobalRef( env, p_sys->circular.u.p_shortarray );
p_sys->circular.u.p_shortarray = NULL;
}
break;
case WRITE_FLOATARRAY:
if( p_sys->circular.u.p_floatarray )
{
(*env)->DeleteGlobalRef( env, p_sys->circular.u.p_floatarray );
p_sys->circular.u.p_floatarray = NULL;
}
break;
case WRITE_BYTEBUFFER:
free( p_sys->circular.u.bytebuffer.p_data );
p_sys->circular.u.bytebuffer.p_data = NULL;
break;
}
p_sys->b_audiotrack_exception = false;
p_sys->b_error = false;
p_sys->b_passthrough = false;
}
/**
* Non blocking write function, run from AudioTrack_Thread.
* Do a calculation between current position and audiotrack position and assure
* that we won't wait in AudioTrack.write() method.
*/
static int
AudioTrack_PlayByteArray( JNIEnv *env, audio_output_t *p_aout,
size_t i_data_size, size_t i_data_offset,
bool b_force )
{
aout_sys_t *p_sys = p_aout->sys;
uint64_t i_samples;
uint64_t i_audiotrack_pos;
uint64_t i_samples_pending;
i_audiotrack_pos = AudioTrack_getPlaybackHeadPosition( env, p_aout );
assert( i_audiotrack_pos <= p_sys->i_samples_written );
if( i_audiotrack_pos > p_sys->i_samples_written )
{
msg_Err( p_aout, "audiotrack position is ahead. Should NOT happen" );
p_sys->i_samples_written = 0;
p_sys->b_error = true;
return 0;
}
i_samples_pending = p_sys->i_samples_written - i_audiotrack_pos;
/* check if audiotrack buffer is not full before writing on it. */
if( b_force )
{
msg_Warn( p_aout, "Force write. It may block..." );
i_samples_pending = 0;
} else if( i_samples_pending >= p_sys->i_max_audiotrack_samples )
return 0;
i_samples = __MIN( p_sys->i_max_audiotrack_samples - i_samples_pending,
BYTES_TO_FRAMES( i_data_size ) );
i_data_size = FRAMES_TO_BYTES( i_samples );
return JNI_AT_CALL_INT( write, p_sys->circular.u.p_bytearray,
i_data_offset, i_data_size );
}
/**
* Non blocking write function for Android M and after, run from
* AudioTrack_Thread. It calls a new write method with WRITE_NON_BLOCKING
* flags.
*/
static int
AudioTrack_PlayByteArrayV23( JNIEnv *env, audio_output_t *p_aout,
size_t i_data_size, size_t i_data_offset )
{
aout_sys_t *p_sys = p_aout->sys;
return JNI_AT_CALL_INT( writeV23, p_sys->circular.u.p_bytearray,
i_data_offset, i_data_size,
jfields.AudioTrack.WRITE_NON_BLOCKING );
}
/**
* Non blocking play function for Lollipop and after, run from
* AudioTrack_Thread. It calls a new write method with WRITE_NON_BLOCKING
* flags.
*/
static int
AudioTrack_PlayByteBuffer( JNIEnv *env, audio_output_t *p_aout,
size_t i_data_size, size_t i_data_offset )
{
aout_sys_t *p_sys = p_aout->sys;
/* The same DirectByteBuffer will be used until the data_offset reaches 0.
* The internal position of this buffer is moved by the writeBufferV21
* wall. */
if( i_data_offset == 0 )
{
/* No need to get a global ref, this object will be only used from the
* same Thread */
if( p_sys->circular.u.bytebuffer.p_obj )
(*env)->DeleteLocalRef( env, p_sys->circular.u.bytebuffer.p_obj );
p_sys->circular.u.bytebuffer.p_obj = (*env)->NewDirectByteBuffer( env,
p_sys->circular.u.bytebuffer.p_data,
p_sys->circular.i_size );
if( !p_sys->circular.u.bytebuffer.p_obj )
{
if( (*env)->ExceptionCheck( env ) )
(*env)->ExceptionClear( env );
return jfields.AudioTrack.ERROR;
}
}
return JNI_AT_CALL_INT( writeBufferV21, p_sys->circular.u.bytebuffer.p_obj,
i_data_size,
jfields.AudioTrack.WRITE_NON_BLOCKING );
}
/**
* Non blocking short write function for Android M and after, run from
* AudioTrack_Thread. It calls a new write method with WRITE_NON_BLOCKING
* flags.
*/
static int
AudioTrack_PlayShortArrayV23( JNIEnv *env, audio_output_t *p_aout,
size_t i_data_size, size_t i_data_offset )
{
aout_sys_t *p_sys = p_aout->sys;
int i_ret;
i_ret = JNI_AT_CALL_INT( writeShortV23, p_sys->circular.u.p_shortarray,
i_data_offset / 2, i_data_size / 2,
jfields.AudioTrack.WRITE_NON_BLOCKING );
if( i_ret < 0 )
return i_ret;
else
return i_ret * 2;
}
/**
* Non blocking play float function for Lollipop and after, run from
* AudioTrack_Thread. It calls a new write method with WRITE_NON_BLOCKING
* flags.
*/
static int
AudioTrack_PlayFloatArray( JNIEnv *env, audio_output_t *p_aout,
size_t i_data_size, size_t i_data_offset )
{
aout_sys_t *p_sys = p_aout->sys;
int i_ret;
i_ret = JNI_AT_CALL_INT( writeFloat, p_sys->circular.u.p_floatarray,
i_data_offset / 4, i_data_size / 4,
jfields.AudioTrack.WRITE_NON_BLOCKING );
if( i_ret < 0 )
return i_ret;
else
return i_ret * 4;
}
static int
AudioTrack_Play( JNIEnv *env, audio_output_t *p_aout, size_t i_data_size,
size_t i_data_offset, bool b_force )
{
aout_sys_t *p_sys = p_aout->sys;
int i_ret;
switch( p_sys->i_write_type )
{
case WRITE_BYTEARRAYV23:
i_ret = AudioTrack_PlayByteArrayV23( env, p_aout, i_data_size,
i_data_offset );
break;
case WRITE_BYTEBUFFER:
i_ret = AudioTrack_PlayByteBuffer( env, p_aout, i_data_size,
i_data_offset );
break;
case WRITE_SHORTARRAYV23:
i_ret = AudioTrack_PlayShortArrayV23( env, p_aout, i_data_size,
i_data_offset );
break;
case WRITE_BYTEARRAY:
i_ret = AudioTrack_PlayByteArray( env, p_aout, i_data_size,
i_data_offset, b_force );
break;
case WRITE_FLOATARRAY:
i_ret = AudioTrack_PlayFloatArray( env, p_aout, i_data_size,
i_data_offset );
break;
default:
vlc_assert_unreachable();
}
if( i_ret < 0 ) {
if( jfields.AudioManager.has_ERROR_DEAD_OBJECT
&& i_ret == jfields.AudioManager.ERROR_DEAD_OBJECT )
{
msg_Warn( p_aout, "ERROR_DEAD_OBJECT: "
"try recreating AudioTrack" );
if( ( i_ret = AudioTrack_Recreate( env, p_aout ) ) == 0 )
{
AudioTrack_Reset( env, p_aout );
JNI_AT_CALL_VOID( play );
CHECK_AT_EXCEPTION( "play" );
}
} else
{
const char *str;
if( i_ret == jfields.AudioTrack.ERROR_INVALID_OPERATION )
str = "ERROR_INVALID_OPERATION";
else if( i_ret == jfields.AudioTrack.ERROR_BAD_VALUE )
str = "ERROR_BAD_VALUE";
else
str = "ERROR";
msg_Err( p_aout, "Write failed: %s", str );
p_sys->b_error = true;
}
} else
p_sys->i_samples_written += BYTES_TO_FRAMES( i_ret );
return i_ret;
}
/**
* This thread will play the data coming from the circular buffer.
*/
static void *
AudioTrack_Thread( void *p_data )
{
audio_output_t *p_aout = p_data;
aout_sys_t *p_sys = p_aout->sys;
JNIEnv *env = GET_ENV();
mtime_t i_play_deadline = 0;
mtime_t i_last_time_blocked = 0;
if( !env )
return NULL;
for( ;; )
{
int i_ret = 0;
bool b_forced;
size_t i_data_offset;
size_t i_data_size;
vlc_mutex_lock( &p_sys->lock );
/* Wait for free space in Audiotrack internal buffer */
if( i_play_deadline != 0 && mdate() < i_play_deadline )
{
/* Don't wake up the thread when there is new data since we are
* waiting for more space */
p_sys->b_thread_waiting = true;
while( p_sys->b_thread_running && i_ret != ETIMEDOUT )
i_ret = vlc_cond_timedwait( &p_sys->thread_cond,
&p_sys->lock,
i_play_deadline );
i_play_deadline = 0;
p_sys->b_thread_waiting = false;
}
/* Wait for not paused state */
while( p_sys->b_thread_running && p_sys->b_thread_paused )
{
i_last_time_blocked = 0;
vlc_cond_wait( &p_sys->thread_cond, &p_sys->lock );
}
/* Wait for more data in the circular buffer */
while( p_sys->b_thread_running
&& p_sys->circular.i_read >= p_sys->circular.i_write )
vlc_cond_wait( &p_sys->thread_cond, &p_sys->lock );
if( !p_sys->b_thread_running || p_sys->b_error )
{
vlc_mutex_unlock( &p_sys->lock );
break;
}
/* HACK: AudioFlinger can drop frames without notifying us and there is
* no way to know it. If it happens, i_audiotrack_pos won't move and
* the current code will be stuck because it'll assume that audiotrack
* internal buffer is full when it's not. It may happen only after
* Android 4.4.2 if we send frames too quickly. To fix this issue,
* force the writing of the buffer after a certain delay. */
if( i_last_time_blocked != 0 )
b_forced = mdate() - i_last_time_blocked >
FRAMES_TO_US( p_sys->i_max_audiotrack_samples ) * 2;
else
b_forced = false;
i_data_offset = p_sys->circular.i_read % p_sys->circular.i_size;
i_data_size = __MIN( p_sys->circular.i_size - i_data_offset,
p_sys->circular.i_write - p_sys->circular.i_read );
i_ret = AudioTrack_Play( env, p_aout, i_data_size, i_data_offset,
b_forced );
if( i_ret >= 0 )
{
if( p_sys->i_write_type == WRITE_BYTEARRAY )
{
if( i_ret != 0 )
i_last_time_blocked = 0;
else if( i_last_time_blocked == 0 )
i_last_time_blocked = mdate();
}
if( i_ret == 0 )
i_play_deadline = mdate() + __MAX( 10000, FRAMES_TO_US(
p_sys->i_max_audiotrack_samples / 5 ) );
else
p_sys->circular.i_read += i_ret;
}
vlc_cond_signal( &p_sys->aout_cond );
vlc_mutex_unlock( &p_sys->lock );
}
if( p_sys->circular.u.bytebuffer.p_obj )
{
(*env)->DeleteLocalRef( env, p_sys->circular.u.bytebuffer.p_obj );
p_sys->circular.u.bytebuffer.p_obj = NULL;
}
return NULL;
}
static void
Play( audio_output_t *p_aout, block_t *p_buffer )
{
JNIEnv *env = NULL;
size_t i_buffer_offset = 0;
aout_sys_t *p_sys = p_aout->sys;
vlc_mutex_lock( &p_sys->lock );
if( p_sys->b_error || !( env = GET_ENV() ) )
goto bailout;
if( p_sys->i_chans_to_reorder )
aout_ChannelReorder( p_buffer->p_buffer, p_buffer->i_buffer,
p_sys->i_chans_to_reorder, p_sys->p_chan_table,
p_sys->fmt.i_format );
while( i_buffer_offset < p_buffer->i_buffer && !p_sys->b_error )
{
size_t i_circular_free;
size_t i_data_offset;
size_t i_data_size;
/* Wait for enough room in circular buffer */
while( !p_sys->b_error && ( i_circular_free = p_sys->circular.i_size -
( p_sys->circular.i_write - p_sys->circular.i_read ) ) == 0 )
vlc_cond_wait( &p_sys->aout_cond, &p_sys->lock );
if( p_sys->b_error )
goto bailout;
i_data_offset = p_sys->circular.i_write % p_sys->circular.i_size;
i_data_size = __MIN( p_buffer->i_buffer - i_buffer_offset,
p_sys->circular.i_size - i_data_offset );
i_data_size = __MIN( i_data_size, i_circular_free );
switch( p_sys->i_write_type )
{
case WRITE_BYTEARRAY:
case WRITE_BYTEARRAYV23:
(*env)->SetByteArrayRegion( env, p_sys->circular.u.p_bytearray,
i_data_offset, i_data_size,
(jbyte *)p_buffer->p_buffer
+ i_buffer_offset);
break;
case WRITE_SHORTARRAYV23:
i_data_offset &= ~1;
i_data_size &= ~1;
(*env)->SetShortArrayRegion( env, p_sys->circular.u.p_shortarray,
i_data_offset / 2, i_data_size / 2,
(jshort *)p_buffer->p_buffer
+ i_buffer_offset / 2);
break;
case WRITE_FLOATARRAY:
i_data_offset &= ~3;
i_data_size &= ~3;
(*env)->SetFloatArrayRegion( env, p_sys->circular.u.p_floatarray,
i_data_offset / 4, i_data_size / 4,
(jfloat *)p_buffer->p_buffer
+ i_buffer_offset / 4);
break;
case WRITE_BYTEBUFFER:
memcpy( p_sys->circular.u.bytebuffer.p_data + i_data_offset,
p_buffer->p_buffer + i_buffer_offset, i_data_size );
break;
}
i_buffer_offset += i_data_size;
p_sys->circular.i_write += i_data_size;
if( !p_sys->b_thread_waiting )
vlc_cond_signal( &p_sys->thread_cond );
}
bailout:
vlc_mutex_unlock( &p_sys->lock );
block_Release( p_buffer );
}
static void
Pause( audio_output_t *p_aout, bool b_pause, mtime_t i_date )
{
aout_sys_t *p_sys = p_aout->sys;
JNIEnv *env;
VLC_UNUSED( i_date );
vlc_mutex_lock( &p_sys->lock );
if( p_sys->b_error || !( env = GET_ENV() ) )
goto bailout;
if( b_pause )
{
p_sys->b_thread_paused = true;
JNI_AT_CALL_VOID( pause );
CHECK_AT_EXCEPTION( "pause" );
} else
{
p_sys->b_thread_paused = false;
AudioTrack_ResetPositions( env, p_aout );
JNI_AT_CALL_VOID( play );
CHECK_AT_EXCEPTION( "play" );
}
bailout:
vlc_mutex_unlock( &p_sys->lock );
}
static void
Flush( audio_output_t *p_aout, bool b_wait )
{
aout_sys_t *p_sys = p_aout->sys;
JNIEnv *env;
vlc_mutex_lock( &p_sys->lock );
if( p_sys->b_error || !( env = GET_ENV() ) )
goto bailout;
/* Android doc:
* stop(): Stops playing the audio data. When used on an instance created
* in MODE_STREAM mode, audio will stop playing after the last buffer that
* was written has been played. For an immediate stop, use pause(),
* followed by flush() to discard audio data that hasn't been played back
* yet.
*
* flush(): Flushes the audio data currently queued for playback. Any data
* that has not been played back will be discarded. No-op if not stopped
* or paused, or if the track's creation mode is not MODE_STREAM.
*/
if( b_wait )
{
/* Wait for the thread to process the circular buffer */
while( !p_sys->b_error
&& p_sys->circular.i_read != p_sys->circular.i_write )
vlc_cond_wait( &p_sys->aout_cond, &p_sys->lock );
if( p_sys->b_error )
goto bailout;
JNI_AT_CALL_VOID( stop );
if( CHECK_AT_EXCEPTION( "stop" ) )
goto bailout;
} else
{
JNI_AT_CALL_VOID( pause );
if( CHECK_AT_EXCEPTION( "pause" ) )
goto bailout;
JNI_AT_CALL_VOID( flush );
}
p_sys->circular.i_read = p_sys->circular.i_write = 0;
/* HACK: Before Android 4.4, the head position is not reset to zero and is
* still moving after a flush or a stop. This prevents to get a precise
* head position and there is no way to know when it stabilizes. Therefore
* recreate an AudioTrack object in that case. The AudioTimestamp class was
* added in API Level 19, so if this class is not found, the Android
* Version is 4.3 or before */
if( !jfields.AudioTimestamp.clazz && p_sys->i_samples_written > 0 )
{
if( AudioTrack_Recreate( env, p_aout ) != 0 )
{
p_sys->b_error = true;
goto bailout;
}
}
AudioTrack_Reset( env, p_aout );
JNI_AT_CALL_VOID( play );
CHECK_AT_EXCEPTION( "play" );
bailout:
vlc_mutex_unlock( &p_sys->lock );
}
static int DeviceSelect(audio_output_t *p_aout, const char *p_id)
{
aout_sys_t *p_sys = p_aout->sys;
enum at_dev at_dev = AT_DEV_DEFAULT;
if( p_id )
{
for( unsigned int i = 0; at_devs[i].id; ++i )
{
if( strncmp( p_id, at_devs[i].id, strlen( at_devs[i].id ) ) == 0 )
{
at_dev = at_devs[i].at_dev;
break;
}
}
}
long long i_encoding_flags = 0;
if( at_dev == AT_DEV_ENCODED )
{
const size_t i_prefix_size = strlen( "encoded:" );
if( strncmp( p_id, "encoded:", i_prefix_size ) == 0 )
i_encoding_flags = atoll( p_id + i_prefix_size );
}
if( at_dev != p_sys->at_dev || i_encoding_flags != p_sys->i_encoding_flags )
{
p_sys->at_dev = at_dev;
p_sys->i_encoding_flags = i_encoding_flags;
aout_RestartRequest( p_aout, AOUT_RESTART_OUTPUT );
msg_Dbg( p_aout, "selected device: %s", p_id );
if( at_dev == AT_DEV_ENCODED )
{
static const vlc_fourcc_t enc_fourccs[] = {
VLC_CODEC_DTS, VLC_CODEC_A52, VLC_CODEC_EAC3, VLC_CODEC_TRUEHD,
};
for( size_t i = 0;
i < sizeof( enc_fourccs ) / sizeof( enc_fourccs[0] ); ++i )
{
bool b_dtshd;
if( AudioTrack_HasEncoding( p_aout, enc_fourccs[i], &b_dtshd ) )
msg_Dbg( p_aout, "device has %4.4s passthrough support",
b_dtshd ? "dtsh" : (const char *)&enc_fourccs[i] );
}
}
}
aout_DeviceReport( p_aout, p_id );
return VLC_SUCCESS;
}
static int
Open( vlc_object_t *obj )
{
audio_output_t *p_aout = (audio_output_t *) obj;
aout_sys_t *p_sys;
JNIEnv *env = GET_ENV();
if( !env || !InitJNIFields( p_aout, env ) )
return VLC_EGENERIC;
p_sys = calloc( 1, sizeof (aout_sys_t) );
if( unlikely( p_sys == NULL ) )
return VLC_ENOMEM;
p_sys->at_dev = AT_DEV_DEFAULT;
vlc_mutex_init(&p_sys->lock);
vlc_cond_init(&p_sys->aout_cond);
vlc_cond_init(&p_sys->thread_cond);
p_aout->sys = p_sys;
p_aout->start = Start;
p_aout->stop = Stop;
p_aout->play = Play;
p_aout->pause = Pause;
p_aout->flush = Flush;
p_aout->time_get = TimeGet;
p_aout->device_select = DeviceSelect;
for( unsigned int i = 0; at_devs[i].id; ++i )
aout_HotplugReport(p_aout, at_devs[i].id, at_devs[i].name);
aout_SoftVolumeInit( p_aout );
return VLC_SUCCESS;
}
static void
Close( vlc_object_t *obj )
{
audio_output_t *p_aout = (audio_output_t *) obj;
aout_sys_t *p_sys = p_aout->sys;
vlc_mutex_destroy(&p_sys->lock);
vlc_cond_destroy(&p_sys->aout_cond);
vlc_cond_destroy(&p_sys->thread_cond);
free( p_sys );
}