-/******************************************************************************
- * audio_output.h : audio output thread interface
- * (c)1999 VideoLAN
- ******************************************************************************
- * Required headers:
- * - <sys/soundcard.h> ( audio_buf_info )
- * - "common.h" ( boolean_t )
- * - "mtime.h" ( mtime_t )
- * - "vlc_thread.h" ( vlc_thread_t )
- ******************************************************************************/
-
-/* TODO :
+/*****************************************************************************
+ * audio_output.h : audio output interface
+ *****************************************************************************
+ * Copyright (C) 2002 VideoLAN
+ * $Id: audio_output.h,v 1.63 2002/09/16 20:46:37 massiot Exp $
*
- * - Créer un flag destroy dans les fifos audio pour indiquer au thread audio
- * qu'il peut libérer la mémoire occupée par le buffer de la fifo lorsqu'il
- * le désire (fin du son ou fin du thread)
- * - Redéplacer les #define dans config.h
+ * Authors: Christophe Massiot <massiot@via.ecp.fr>
*
- */
-
-/*
- * Defines => "config.h"
- */
-
-/* Default output device. You probably should not change this. */
-#define AOUT_DEFAULT_DEVICE "/dev/dsp"
-
-/* Default audio output format (AFMT_S16_NE = Native Endianess) */
-#define AOUT_DEFAULT_FORMAT AFMT_S16_NE
-
-/* Default stereo mode (0 stands for mono, 1 for stereo) */
-#define AOUT_DEFAULT_STEREO 1
-
-/* Audio output rate, in Hz */
-#define AOUT_MIN_RATE 22050 /* ?? */
-#define AOUT_DEFAULT_RATE 44100
-#define AOUT_MAX_RATE 48000
-
-/* Number of audio samples (s16 integers) contained in an audio output frame...
- * - Layer I : a decoded frame contains 384 samples
- * - Layer II & III : a decoded frame contains 1152 = 3*384 samples */
-/*
-#define AOUT_FRAME_SIZE 384
-*/
-
-/* Number of audio output frames contained in an audio output fifo.
- * (AOUT_FIFO_SIZE + 1) must be a power of 2, in order to optimise the
- * %(AOUT_FIFO_SIZE + 1) operation with an &AOUT_FIFO_SIZE.
- * With 511 we have at least 511*384/2/48000=2 seconds of sound */
-#define AOUT_FIFO_SIZE 511
-
-/* Maximum number of audio fifos. The value of AOUT_MAX_FIFOS should be a power
- * of two, in order to optimize the '/AOUT_MAX_FIFOS' and '*AOUT_MAX_FIFOS'
- * operations with '>>' and '<<' (gcc changes this at compilation-time) */
-#define AOUT_MAX_FIFOS 2
-
-/* Duration (in microseconds) of an audio output buffer should be :
- * - short, in order to be able to play a new song very quickly (especially a
- * song from the interface)
- * - long, in order to perform the buffer calculations as few as possible */
-#define AOUT_BUFFER_DURATION 100000
-
-/*
- * Macros
- */
-#define AOUT_FIFO_ISEMPTY( fifo ) ( (fifo).l_end_frame == (fifo).i_start_frame )
-#define AOUT_FIFO_ISFULL( fifo ) ( ((((fifo).l_end_frame + 1) - (fifo).l_start_frame) & AOUT_FIFO_SIZE) == 0 )
-
-/******************************************************************************
- * aout_dsp_t
- ******************************************************************************/
-typedef struct
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ *****************************************************************************/
+
+/*****************************************************************************
+ * audio_sample_format_t
+ *****************************************************************************
+ * This structure defines a format for audio samples.
+ *****************************************************************************/
+struct audio_sample_format_t
{
- /* Path to the audio output device (default is set to "/dev/dsp") */
- char * psz_device;
- int i_fd;
-
- /* Format of the audio output samples (see <sys/soundcard.h>) */
int i_format;
- /* Following boolean is set to 0 if output sound is mono, 1 if stereo */
- boolean_t b_stereo;
- /* Rate of the audio output sound (in Hz) */
- long l_rate;
-
- /* Buffer information structure, used by aout_dspGetBufInfo() to store the
- * current state of the internal sound card buffer */
- audio_buf_info buf_info;
-
-} aout_dsp_t;
-
-/******************************************************************************
- * aout_increment_t
- ******************************************************************************
- * This structure is used to keep the progression of an index up-to-date, in
- * order to avoid rounding problems and heavy computations, as the function
- * that handles this structure only uses additions.
- ******************************************************************************/
-typedef struct
-{
- /* The remainder is used to keep track of the fractional part of the
- * index. */
- long l_remainder;
-
- /*
- * The increment structure is initialized with the result of an euclidean
- * division :
- *
- * l_euclidean_numerator l_euclidean_remainder
- * ----------------------- = l_euclidean_integer + -----------------------
- * l_euclidean_denominator l_euclidean_denominator
- *
- */
- long l_euclidean_integer;
- long l_euclidean_remainder;
- long l_euclidean_denominator;
-
-} aout_increment_t;
-
-/******************************************************************************
- * aout_frame_t
- ******************************************************************************/
-/*typedef s16 aout_frame_t[ AOUT_FRAME_SIZE ];*/
-
-/******************************************************************************
- * aout_fifo_t
- ******************************************************************************/
-typedef struct
+ int i_rate;
+ int i_channels;
+ /* Optional - for A52, SPDIF and DTS types */
+ int i_bytes_per_frame;
+ int i_frame_length;
+ /* Please note that it may be completely arbitrary - buffers are not
+ * obliged to contain a integral number of so-called "frames". It's
+ * just here for the division :
+ * i_nb_samples * i_bytes_per_frame / i_frame_length */
+};
+
+#define AOUT_FMT_MU_LAW 0x00000001
+#define AOUT_FMT_A_LAW 0x00000002
+#define AOUT_FMT_IMA_ADPCM 0x00000004
+#define AOUT_FMT_U8 0x00000008
+#define AOUT_FMT_S16_LE 0x00000010 /* Little endian signed 16 */
+#define AOUT_FMT_S16_BE 0x00000020 /* Big endian signed 16 */
+#define AOUT_FMT_S8 0x00000040
+#define AOUT_FMT_U16_LE 0x00000080 /* Little endian U16 */
+#define AOUT_FMT_U16_BE 0x00000100 /* Big endian U16 */
+#define AOUT_FMT_SPDIF 0x00000400 /* S/PDIF hardware support */
+#define AOUT_FMT_FLOAT32 0x00010000
+#define AOUT_FMT_FIXED32 0x00020000
+#define AOUT_FMT_A52 0x00100000
+#define AOUT_FMT_DTS 0x00200000
+
+#define AOUT_FMTS_IDENTICAL( p_first, p_second ) ( \
+ ((p_first)->i_format == (p_second)->i_format) \
+ && ((p_first)->i_rate == (p_second)->i_rate) \
+ && ((p_first)->i_channels == (p_second)->i_channels \
+ || (p_first)->i_channels == -1 || (p_second)->i_channels == -1) )
+
+/* Check if i_rate == i_rate and i_channels == i_channels */
+#define AOUT_FMTS_SIMILAR( p_first, p_second ) ( \
+ ((p_first)->i_rate == (p_second)->i_rate) \
+ && ((p_first)->i_channels == (p_second)->i_channels \
+ || (p_first)->i_channels == -1 || (p_second)->i_channels == -1) )
+
+#ifdef WORDS_BIGENDIAN
+# define AOUT_FMT_S16_NE AOUT_FMT_S16_BE
+# define AOUT_FMT_U16_NE AOUT_FMT_U16_BE
+#else
+# define AOUT_FMT_S16_NE AOUT_FMT_S16_LE
+# define AOUT_FMT_U16_NE AOUT_FMT_U16_LE
+#endif
+
+#define AOUT_FMT_NON_LINEAR( p_format ) \
+ ( ((p_format)->i_format == AOUT_FMT_SPDIF) \
+ || ((p_format)->i_format == AOUT_FMT_A52) \
+ || ((p_format)->i_format == AOUT_FMT_DTS) )
+
+/* This is heavily borrowed from libmad, by Robert Leslie <rob@mars.org> */
+/*
+ * Fixed-point format: 0xABBBBBBB
+ * A == whole part (sign + 3 bits)
+ * B == fractional part (28 bits)
+ *
+ * Values are signed two's complement, so the effective range is:
+ * 0x80000000 to 0x7fffffff
+ * -8.0 to +7.9999999962747097015380859375
+ *
+ * The smallest representable value is:
+ * 0x00000001 == 0.0000000037252902984619140625 (i.e. about 3.725e-9)
+ *
+ * 28 bits of fractional accuracy represent about
+ * 8.6 digits of decimal accuracy.
+ *
+ * Fixed-point numbers can be added or subtracted as normal
+ * integers, but multiplication requires shifting the 64-bit result
+ * from 56 fractional bits back to 28 (and rounding.)
+ */
+typedef s32 vlc_fixed_t;
+#define FIXED32_FRACBITS 28
+#define FIXED32_MIN ((vlc_fixed_t) -0x80000000L)
+#define FIXED32_MAX ((vlc_fixed_t) +0x7fffffffL)
+#define FIXED32_ONE ((vlc_fixed_t) 0x10000000)
+
+
+/* Dual mono. Two independant mono channels */
+#define AOUT_CHAN_CHANNEL 0x0000000B
+#define AOUT_CHAN_MONO 0x00000001
+#define AOUT_CHAN_STEREO 0x00000002
+/* 3 front channels (left, center, right) */
+#define AOUT_CHAN_3F 0x00000003
+/* 2 front, 1 rear surround channels (L, R, S) */
+#define AOUT_CHAN_2F1R 0x00000004
+/* 3 front, 1 rear surround channels (L, C, R, S) */
+#define AOUT_CHAN_3F1R 0x00000005
+/* 2 front, 2 rear surround channels (L, R, LS, RS) */
+#define AOUT_CHAN_2F2R 0x00000006
+/* 3 front, 2 rear surround channels (L, C, R, LS, RS) */
+#define AOUT_CHAN_3F2R 0x00000007
+/* First of two mono channels */
+#define AOUT_CHAN_CHANNEL1 0x00000008
+/* Second of two mono channels */
+#define AOUT_CHAN_CHANNEL2 0x00000009
+/* Dolby surround compatible stereo */
+#define AOUT_CHAN_DOLBY 0x0000000A
+
+#define AOUT_CHAN_MASK 0x0000000F
+
+/* Low frequency effects channel. Normally used to connect a subwoofer.
+ * Can be combined with any of the above channels. For example :
+ * AOUT_CHAN_3F2R | AOUT_CHAN_LFE -> 3 front, 2 rear, 1 LFE (5.1) */
+#define AOUT_CHAN_LFE 0x00000010
+
+
+/*****************************************************************************
+ * aout_buffer_t : audio output buffer
+ *****************************************************************************/
+struct aout_buffer_t
{
- /* See the fifo types below */
- int i_type;
- boolean_t b_die;
-
- boolean_t b_stereo;
- long l_rate;
-
- vlc_mutex_t data_lock;
- vlc_cond_t data_wait;
-
- long l_frame_size;
- void * buffer;
- mtime_t * date;
- /* The start frame is the first frame in the buffer that contains decoded
- * audio data. It it also the first frame in the current timestamped frame
- * area, ie the first dated frame in the decoded part of the buffer. :-p */
- long l_start_frame;
- boolean_t b_start_frame;
- /* The next frame is the end frame of the current timestamped frame area,
- * ie the first dated frame after the start frame. */
- long l_next_frame;
- boolean_t b_next_frame;
- /* The end frame is the first frame, after the start frame, that doesn't
- * contain decoded audio data. That's why the end frame is the first frame
- * where the audio decoder can store its decoded audio frames. */
- long l_end_frame;
-
- long l_unit;
- aout_increment_t unit_increment;
- /* The following variable is used to store the number of remaining audio
- * units in the current timestamped frame area. */
- long l_units;
-
-} aout_fifo_t;
-
-#define AOUT_EMPTY_FIFO 0
-#define AOUT_INTF_MONO_FIFO 1
-#define AOUT_INTF_STEREO_FIFO 2
-#define AOUT_ADEC_MONO_FIFO 3
-#define AOUT_ADEC_STEREO_FIFO 4
-
-/******************************************************************************
- * aout_thread_t
- ******************************************************************************/
-typedef struct aout_thread_s
+ byte_t * p_buffer;
+ int i_alloc_type;
+ /* i_size is the real size of the buffer (used for debug ONLY), i_nb_bytes
+ * is the number of significative bytes in it. */
+ size_t i_size, i_nb_bytes;
+ int i_nb_samples;
+ mtime_t start_date, end_date;
+
+ struct aout_buffer_t * p_next;
+};
+
+/* Size of a frame for S/PDIF output. */
+#define AOUT_SPDIF_SIZE 6144
+
+/*****************************************************************************
+ * audio_date_t : date incrementation without long-term rounding errors
+ *****************************************************************************/
+struct audio_date_t
{
- vlc_thread_t thread_id;
- boolean_t b_die;
-
- aout_dsp_t dsp;
-
- vlc_mutex_t fifos_lock;
- aout_fifo_t fifo[ AOUT_MAX_FIFOS ];
-
- void * buffer;
- /* The s32 buffer is used to mix all the audio fifos together before
- * converting them and storing them in the audio output buffer */
- s32 * s32_buffer;
+ mtime_t date;
+ u32 i_divider;
+ u32 i_remainder;
+};
- /* The size of the audio output buffer is kept in audio units, as this is
- * the only unit that is common with every audio decoder and audio fifo */
- long l_units;
+/*****************************************************************************
+ * audio_volume_t : integer value for the audio volume
+ *****************************************************************************/
+typedef unsigned int audio_volume_t;
- mtime_t date;
- /* date is the moment where the first audio unit of the output buffer
- * should be played and is kept up-to-date with the following incremental
- * structure */
- aout_increment_t date_increment;
-
-} aout_thread_t;
-
-/******************************************************************************
+/*****************************************************************************
* Prototypes
- ******************************************************************************/
-aout_thread_t * aout_CreateThread ( int *pi_status );
-void aout_DestroyThread ( aout_thread_t *p_aout, int *pi_status );
-
+ *****************************************************************************/
+/* From audio_output.c : */
+#define aout_NewInstance(a) __aout_NewInstance(VLC_OBJECT(a))
+VLC_EXPORT( aout_instance_t *, __aout_NewInstance, ( vlc_object_t * ) );
+VLC_EXPORT( void, aout_DeleteInstance, ( aout_instance_t * ) );
+VLC_EXPORT( aout_buffer_t *, aout_BufferNew, ( aout_instance_t *, aout_input_t *, size_t ) );
+VLC_EXPORT( void, aout_BufferDelete, ( aout_instance_t *, aout_input_t *, aout_buffer_t * ) );
+VLC_EXPORT( void, aout_BufferPlay, ( aout_instance_t *, aout_input_t *, aout_buffer_t * ) );
+VLC_EXPORT( void, aout_DateInit, ( audio_date_t *, u32 ) );
+VLC_EXPORT( void, aout_DateSet, ( audio_date_t *, mtime_t ) );
+VLC_EXPORT( void, aout_DateMove, ( audio_date_t *, mtime_t ) );
+VLC_EXPORT( mtime_t, aout_DateGet, ( const audio_date_t * ) );
+VLC_EXPORT( mtime_t, aout_DateIncrement, ( audio_date_t *, u32 ) );
+
+/* From input.c : */
+#define aout_InputNew(a,b,c) __aout_InputNew(VLC_OBJECT(a),b,c)
+VLC_EXPORT( aout_input_t *, __aout_InputNew, ( vlc_object_t *, aout_instance_t **, audio_sample_format_t * ) );
+VLC_EXPORT( void, aout_InputDelete, ( aout_instance_t *, aout_input_t * ) );
+
+/* From intf.c : */
-aout_fifo_t * aout_CreateFifo ( aout_thread_t *p_aout, aout_fifo_t *p_fifo );
-void aout_DestroyFifo ( aout_fifo_t *p_fifo );
projects / vlc / blobdiff