1 /*****************************************************************************
2 * vlc_aout.h : audio output interface
3 *****************************************************************************
4 * Copyright (C) 2002-2011 the VideoLAN team
6 * Authors: Christophe Massiot <massiot@via.ecp.fr>
8 * This program is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU General Public License as published by
10 * the Free Software Foundation; either version 2 of the License, or
11 * (at your option) any later version.
13 * This program is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU General Public License
19 * along with this program; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
21 *****************************************************************************/
28 * This file defines functions, structures and macros for audio output object
31 /* Max number of pre-filters per input, and max number of post-filters */
32 #define AOUT_MAX_FILTERS 10
34 /* Buffers which arrive in advance of more than AOUT_MAX_ADVANCE_TIME
35 * will be considered as bogus and be trashed */
36 #define AOUT_MAX_ADVANCE_TIME (AOUT_MAX_PREPARE_TIME + CLOCK_FREQ)
38 /* Buffers which arrive in advance of more than AOUT_MAX_PREPARE_TIME
39 * will cause the calling thread to sleep */
40 #define AOUT_MAX_PREPARE_TIME (2 * CLOCK_FREQ)
42 /* Buffers which arrive after pts - AOUT_MIN_PREPARE_TIME will be trashed
43 * to avoid too heavy resampling */
44 #define AOUT_MIN_PREPARE_TIME AOUT_MAX_PTS_ADVANCE
46 /* Tolerance values from EBU Recommendation 37 */
47 /** Maximum advance of actual audio playback time to coded PTS,
48 * above which downsampling will be performed */
49 #define AOUT_MAX_PTS_ADVANCE (CLOCK_FREQ / 25)
51 /** Maximum delay of actual audio playback time from coded PTS,
52 * above which upsampling will be performed */
53 #define AOUT_MAX_PTS_DELAY (3 * CLOCK_FREQ / 50)
55 /* Max acceptable resampling (in %) */
56 #define AOUT_MAX_RESAMPLING 10
60 #define AOUT_FMTS_IDENTICAL( p_first, p_second ) ( \
61 ((p_first)->i_format == (p_second)->i_format) \
62 && AOUT_FMTS_SIMILAR(p_first, p_second) )
64 /* Check if i_rate == i_rate and i_channels == i_channels */
65 #define AOUT_FMTS_SIMILAR( p_first, p_second ) ( \
66 ((p_first)->i_rate == (p_second)->i_rate) \
67 && ((p_first)->i_physical_channels == (p_second)->i_physical_channels)\
68 && ((p_first)->i_original_channels == (p_second)->i_original_channels) )
70 #define VLC_CODEC_SPDIFL VLC_FOURCC('s','p','d','i')
71 #define VLC_CODEC_SPDIFB VLC_FOURCC('s','p','d','b')
73 #define AOUT_FMT_NON_LINEAR( p_format ) \
74 ( ((p_format)->i_format == VLC_CODEC_SPDIFL) \
75 || ((p_format)->i_format == VLC_CODEC_SPDIFB) \
76 || ((p_format)->i_format == VLC_CODEC_A52) \
77 || ((p_format)->i_format == VLC_CODEC_DTS) )
79 /* This is heavily borrowed from libmad, by Robert Leslie <rob@mars.org> */
81 * Fixed-point format: 0xABBBBBBB
82 * A == whole part (sign + 3 bits)
83 * B == fractional part (28 bits)
85 * Values are signed two's complement, so the effective range is:
86 * 0x80000000 to 0x7fffffff
87 * -8.0 to +7.9999999962747097015380859375
89 * The smallest representable value is:
90 * 0x00000001 == 0.0000000037252902984619140625 (i.e. about 3.725e-9)
92 * 28 bits of fractional accuracy represent about
93 * 8.6 digits of decimal accuracy.
95 * Fixed-point numbers can be added or subtracted as normal
96 * integers, but multiplication requires shifting the 64-bit result
97 * from 56 fractional bits back to 28 (and rounding.)
99 typedef int32_t vlc_fixed_t;
100 #define FIXED32_FRACBITS 28
101 #define FIXED32_MIN ((vlc_fixed_t) -0x80000000L)
102 #define FIXED32_MAX ((vlc_fixed_t) +0x7fffffffL)
103 #define FIXED32_ONE ((vlc_fixed_t) 0x10000000)
106 * Channels descriptions
109 /* Values available for physical and original channels */
110 #define AOUT_CHAN_CENTER 0x1
111 #define AOUT_CHAN_LEFT 0x2
112 #define AOUT_CHAN_RIGHT 0x4
113 #define AOUT_CHAN_REARCENTER 0x10
114 #define AOUT_CHAN_REARLEFT 0x20
115 #define AOUT_CHAN_REARRIGHT 0x40
116 #define AOUT_CHAN_MIDDLELEFT 0x100
117 #define AOUT_CHAN_MIDDLERIGHT 0x200
118 #define AOUT_CHAN_LFE 0x1000
120 /* Values available for original channels only */
121 #define AOUT_CHAN_DOLBYSTEREO 0x10000
122 #define AOUT_CHAN_DUALMONO 0x20000
123 #define AOUT_CHAN_REVERSESTEREO 0x40000
125 #define AOUT_CHAN_PHYSMASK 0xFFFF
126 #define AOUT_CHAN_MAX 9
128 /* Values used for the audio-device and audio-channels object variables */
129 #define AOUT_VAR_MONO 1
130 #define AOUT_VAR_STEREO 2
131 #define AOUT_VAR_2F2R 4
132 #define AOUT_VAR_3F2R 5
133 #define AOUT_VAR_5_1 6
134 #define AOUT_VAR_6_1 7
135 #define AOUT_VAR_7_1 8
136 #define AOUT_VAR_SPDIF 10
138 #define AOUT_VAR_CHAN_STEREO 1
139 #define AOUT_VAR_CHAN_RSTEREO 2
140 #define AOUT_VAR_CHAN_LEFT 3
141 #define AOUT_VAR_CHAN_RIGHT 4
142 #define AOUT_VAR_CHAN_DOLBYS 5
144 /*****************************************************************************
145 * Main audio output structures
146 *****************************************************************************/
148 #define aout_BufferFree( buffer ) block_Release( buffer )
150 /* Size of a frame for S/PDIF output. */
151 #define AOUT_SPDIF_SIZE 6144
153 /* Number of samples in an A/52 frame. */
154 #define A52_FRAME_NB 1536
156 /** audio output buffer FIFO */
159 aout_buffer_t * p_first;
160 aout_buffer_t ** pp_last;
164 /* FIXME to remove once aout.h is cleaned a bit more */
165 #include <vlc_block.h>
167 typedef int (*aout_volume_cb) (audio_output_t *, float, bool);
169 /** Audio output object */
176 audio_sample_format_t format; /**< Output format (plugin can modify it
177 only when succesfully probed and not afterward) */
181 struct aout_sys_t *sys; /**< Output plugin private data */
182 void (*pf_play)(audio_output_t *, block_t *); /**< Audio buffer callback */
183 void (* pf_pause)( audio_output_t *, bool, mtime_t ); /**< Pause/resume
184 callback (optional, may be NULL) */
185 void (* pf_flush)( audio_output_t *, bool ); /**< Flush/drain callback
186 (optional, may be NULL) */
187 aout_volume_cb pf_volume_set; /**< Volume setter (or NULL) */
192 * It describes the audio channel order VLC expect.
194 static const uint32_t pi_vlc_chan_order_wg4[] =
196 AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT,
197 AOUT_CHAN_MIDDLELEFT, AOUT_CHAN_MIDDLERIGHT,
198 AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT, AOUT_CHAN_REARCENTER,
199 AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0
202 /*****************************************************************************
204 *****************************************************************************/
206 VLC_API aout_buffer_t * aout_OutputNextBuffer( audio_output_t *, mtime_t, bool ) VLC_USED;
209 * This function computes the reordering needed to go from pi_chan_order_in to
211 * If pi_chan_order_in or pi_chan_order_out is NULL, it will assume that vlc
212 * internal (WG4) order is requested.
214 VLC_API int aout_CheckChannelReorder( const uint32_t *pi_chan_order_in, const uint32_t *pi_chan_order_out, uint32_t i_channel_mask, int i_channels, int *pi_chan_table );
215 VLC_API void aout_ChannelReorder( uint8_t *, int, int, const int *, int );
218 * This fonction will compute the extraction parameter into pi_selection to go
219 * from i_channels with their type given by pi_order_src[] into the order
220 * describe by pi_order_dst.
222 * - *pi_channels as the number of channels that will be extracted which is
223 * lower (in case of non understood channels type) or equal to i_channels.
224 * - the layout of the channels (*pi_layout).
226 * It will return true if channel extraction is really needed, in which case
227 * aout_ChannelExtract must be used
229 * XXX It must be used when the source may have channel type not understood
230 * by VLC. In this case the channel type pi_order_src[] must be set to 0.
231 * XXX It must also be used if multiple channels have the same type.
233 VLC_API bool aout_CheckChannelExtraction( int *pi_selection, uint32_t *pi_layout, int *pi_channels, const uint32_t pi_order_dst[AOUT_CHAN_MAX], const uint32_t *pi_order_src, int i_channels );
236 * Do the actual channels extraction using the parameters created by
237 * aout_CheckChannelExtraction.
239 * XXX this function does not work in place (p_dst and p_src must not overlap).
240 * XXX Only 8, 16, 24, 32, 64 bits per sample are supported.
242 VLC_API void aout_ChannelExtract( void *p_dst, int i_dst_channels, const void *p_src, int i_src_channels, int i_sample_count, const int *pi_selection, int i_bits_per_sample );
245 static inline unsigned aout_FormatNbChannels(const audio_sample_format_t *fmt)
247 return popcount(fmt->i_physical_channels & AOUT_CHAN_PHYSMASK);
250 VLC_API unsigned int aout_BitsPerSample( vlc_fourcc_t i_format ) VLC_USED;
251 VLC_API void aout_FormatPrepare( audio_sample_format_t * p_format );
252 VLC_API void aout_FormatPrint( audio_output_t * p_aout, const char * psz_text, const audio_sample_format_t * p_format );
253 VLC_API const char * aout_FormatPrintChannels( const audio_sample_format_t * ) VLC_USED;
255 VLC_API mtime_t aout_FifoFirstDate( const aout_fifo_t * ) VLC_USED;
256 VLC_API aout_buffer_t *aout_FifoPop( aout_fifo_t * p_fifo ) VLC_USED;
257 VLC_API void aout_FifoPush( aout_fifo_t *, block_t * );
259 VLC_API void aout_VolumeNoneInit( audio_output_t * );
260 VLC_API void aout_VolumeSoftInit( audio_output_t * );
261 VLC_API void aout_VolumeHardInit( audio_output_t *, aout_volume_cb );
262 VLC_API void aout_VolumeHardSet( audio_output_t *, float, bool );
264 VLC_API int aout_ChannelsRestart( vlc_object_t *, const char *, vlc_value_t, vlc_value_t, void * );
267 VLC_API vout_thread_t * aout_filter_RequestVout( filter_t *, vout_thread_t *p_vout, video_format_t *p_fmt ) VLC_USED;
269 #endif /* VLC_AOUT_H */