1 /*****************************************************************************
2 * vlc_aout.h : audio output interface
3 *****************************************************************************
4 * Copyright (C) 2002-2011 VLC authors and VideoLAN
6 * Authors: Christophe Massiot <massiot@via.ecp.fr>
8 * This program is free software; you can redistribute it and/or modify it
9 * under the terms of the GNU Lesser General Public License as published by
10 * the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public License
19 * along with this program; if not, write to the Free Software Foundation,
20 * 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 /* Buffers which arrive in advance of more than AOUT_MAX_ADVANCE_TIME
32 * will be considered as bogus and be trashed */
33 #define AOUT_MAX_ADVANCE_TIME (AOUT_MAX_PREPARE_TIME + CLOCK_FREQ)
35 /* Buffers which arrive in advance of more than AOUT_MAX_PREPARE_TIME
36 * will cause the calling thread to sleep */
37 #define AOUT_MAX_PREPARE_TIME (2 * CLOCK_FREQ)
39 /* Buffers which arrive after pts - AOUT_MIN_PREPARE_TIME will be trashed
40 * to avoid too heavy resampling */
41 #define AOUT_MIN_PREPARE_TIME AOUT_MAX_PTS_ADVANCE
43 /* Tolerance values from EBU Recommendation 37 */
44 /** Maximum advance of actual audio playback time to coded PTS,
45 * above which downsampling will be performed */
46 #define AOUT_MAX_PTS_ADVANCE (CLOCK_FREQ / 25)
48 /** Maximum delay of actual audio playback time from coded PTS,
49 * above which upsampling will be performed */
50 #define AOUT_MAX_PTS_DELAY (3 * CLOCK_FREQ / 50)
52 /* Max acceptable resampling (in %) */
53 #define AOUT_MAX_RESAMPLING 10
57 #define AOUT_FMTS_IDENTICAL( p_first, p_second ) ( \
58 ((p_first)->i_format == (p_second)->i_format) \
59 && AOUT_FMTS_SIMILAR(p_first, p_second) )
61 /* Check if i_rate == i_rate and i_channels == i_channels */
62 #define AOUT_FMTS_SIMILAR( p_first, p_second ) ( \
63 ((p_first)->i_rate == (p_second)->i_rate) \
64 && ((p_first)->i_physical_channels == (p_second)->i_physical_channels)\
65 && ((p_first)->i_original_channels == (p_second)->i_original_channels) )
67 #define AOUT_FMT_LINEAR( p_format ) \
68 (aout_BitsPerSample((p_format)->i_format) != 0)
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_SPDIF( 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)
105 /* Values used for the audio-device and audio-channels object variables */
106 #define AOUT_VAR_MONO 1
107 #define AOUT_VAR_STEREO 2
108 #define AOUT_VAR_2F2R 4
109 #define AOUT_VAR_3F2R 5
110 #define AOUT_VAR_5_1 6
111 #define AOUT_VAR_6_1 7
112 #define AOUT_VAR_7_1 8
113 #define AOUT_VAR_SPDIF 10
115 #define AOUT_VAR_CHAN_UNSET 0 /* must be zero */
116 #define AOUT_VAR_CHAN_STEREO 1
117 #define AOUT_VAR_CHAN_RSTEREO 2
118 #define AOUT_VAR_CHAN_LEFT 3
119 #define AOUT_VAR_CHAN_RIGHT 4
120 #define AOUT_VAR_CHAN_DOLBYS 5
122 /*****************************************************************************
123 * Main audio output structures
124 *****************************************************************************/
126 /* Size of a frame for S/PDIF output. */
127 #define AOUT_SPDIF_SIZE 6144
129 /* Number of samples in an A/52 frame. */
130 #define A52_FRAME_NB 1536
132 /* FIXME to remove once aout.h is cleaned a bit more */
133 #include <vlc_block.h>
135 /** Audio output object */
140 struct aout_sys_t *sys; /**< Private data for callbacks */
142 int (*start)(audio_output_t *, audio_sample_format_t *fmt);
143 /**< Starts a new stream (mandatory, cannot be NULL).
144 * \param fmt input stream sample format upon entry,
145 * output stream sample format upon return [IN/OUT]
146 * \return VLC_SUCCESS on success, non-zero on failure
147 * \note No other stream may be already started when called.
149 void (*stop)(audio_output_t *);
150 /**< Stops the existing stream (optional, may be NULL).
151 * \note A stream must have been started when called.
153 int (*time_get)(audio_output_t *, mtime_t *delay);
154 /**< Estimates playback buffer latency (optional, may be NULL).
155 * \param delay pointer to the delay until the next sample to be written
156 * to the playback buffer is rendered [OUT]
157 * \return 0 on success, non-zero on failure or lack of data
158 * \note A stream must have been started when called.
160 void (*play)(audio_output_t *, block_t *);
161 /**< Queues a block of samples for playback (mandatory, cannot be NULL).
162 * \note A stream must have been started when called.
164 void (*pause)( audio_output_t *, bool pause, mtime_t date);
165 /**< Pauses or resumes playback (optional, may be NULL).
166 * \param pause pause if true, resume from pause if false
167 * \param date timestamp when the pause or resume was requested
168 * \note A stream must have been started when called.
170 void (*flush)( audio_output_t *, bool wait);
171 /**< Flushes or drains the playback buffers (mandatoryl, cannot be NULL).
172 * \param wait true to wait for playback of pending buffers (drain),
173 * false to discard pending buffers (flush)
174 * \note A stream must have been started when called.
176 int (*volume_set)(audio_output_t *, float volume);
177 /**< Changes playback volume (optional, may be NULL).
178 * \param volume requested volume (0. = mute, 1. = nominal)
179 * \note The volume is always a positive number.
180 * \warning A stream may or may not have been started when called.
182 int (*mute_set)(audio_output_t *, bool mute);
183 /**< Changes muting (optinal, may be NULL).
184 * \param mute true to mute, false to unmute
185 * \warning A stream may or may not have been started when called.
187 int (*device_enum)(audio_output_t *, char ***ids, char ***names);
188 /**< Enumerates available audio output devices (optional, may be NULL).
189 * \param ids pointer to a heap-allocated table of heap-allocated
190 * nul-terminated device unique identifiers [OUT]
191 * \param names pointer to a heap-allocated table of heap-allocated
192 * nul-terminated device human-readable names [OUT]
193 * \return The number of entries, or -1 on error.
194 * \warning A stream may or may not have been started when called.
196 int (*device_select)(audio_output_t *, const char *id);
197 /**< Selects an audio output device (optional, may be NULL).
198 * \param id nul-terminated device unique identifier.
199 * \return 0 on success, non-zero on failure.
200 * \warning A stream may or may not have been started when called.
203 void (*volume_report)(audio_output_t *, float);
204 void (*mute_report)(audio_output_t *, bool);
205 void (*policy_report)(audio_output_t *, bool);
206 void (*device_report)(audio_output_t *, const char *);
207 int (*gain_request)(audio_output_t *, float);
212 * It describes the audio channel order VLC expect.
214 static const uint32_t pi_vlc_chan_order_wg4[] =
216 AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT,
217 AOUT_CHAN_MIDDLELEFT, AOUT_CHAN_MIDDLERIGHT,
218 AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT, AOUT_CHAN_REARCENTER,
219 AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0
222 /*****************************************************************************
224 *****************************************************************************/
227 * This function computes the reordering needed to go from pi_chan_order_in to
229 * If pi_chan_order_in or pi_chan_order_out is NULL, it will assume that vlc
230 * internal (WG4) order is requested.
232 VLC_API unsigned aout_CheckChannelReorder( const uint32_t *, const uint32_t *,
233 uint32_t mask, uint8_t *table );
234 VLC_API void aout_ChannelReorder( void *, size_t, unsigned, const uint8_t *, unsigned );
237 * This fonction will compute the extraction parameter into pi_selection to go
238 * from i_channels with their type given by pi_order_src[] into the order
239 * describe by pi_order_dst.
241 * - *pi_channels as the number of channels that will be extracted which is
242 * lower (in case of non understood channels type) or equal to i_channels.
243 * - the layout of the channels (*pi_layout).
245 * It will return true if channel extraction is really needed, in which case
246 * aout_ChannelExtract must be used
248 * XXX It must be used when the source may have channel type not understood
249 * by VLC. In this case the channel type pi_order_src[] must be set to 0.
250 * XXX It must also be used if multiple channels have the same type.
252 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 );
255 * Do the actual channels extraction using the parameters created by
256 * aout_CheckChannelExtraction.
258 * XXX this function does not work in place (p_dst and p_src must not overlap).
259 * XXX Only 8, 16, 24, 32, 64 bits per sample are supported.
261 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 );
264 static inline unsigned aout_FormatNbChannels(const audio_sample_format_t *fmt)
266 return popcount(fmt->i_physical_channels);
269 VLC_API unsigned int aout_BitsPerSample( vlc_fourcc_t i_format ) VLC_USED;
270 VLC_API void aout_FormatPrepare( audio_sample_format_t * p_format );
271 VLC_API void aout_FormatPrint(vlc_object_t *, const char *,
272 const audio_sample_format_t *);
273 #define aout_FormatPrint(o, t, f) aout_FormatPrint(VLC_OBJECT(o), t, f)
274 VLC_API const char * aout_FormatPrintChannels( const audio_sample_format_t * ) VLC_USED;
276 VLC_API float aout_VolumeGet (audio_output_t *);
277 VLC_API int aout_VolumeSet (audio_output_t *, float);
278 VLC_API int aout_MuteGet (audio_output_t *);
279 VLC_API int aout_MuteSet (audio_output_t *, bool);
280 VLC_API char *aout_DeviceGet (audio_output_t *);
281 VLC_API int aout_DeviceSet (audio_output_t *, const char *);
282 VLC_API int aout_DevicesList (audio_output_t *, char ***, char ***);
285 * Report change of configured audio volume to the core and UI.
287 static inline void aout_VolumeReport(audio_output_t *aout, float volume)
289 aout->event.volume_report(aout, volume);
293 * Report change of muted flag to the core and UI.
295 static inline void aout_MuteReport(audio_output_t *aout, bool mute)
297 aout->event.mute_report(aout, mute);
301 * Report audio policy status.
302 * \parm cork true to request a cork, false to undo any pending cork.
304 static inline void aout_PolicyReport(audio_output_t *aout, bool cork)
306 aout->event.policy_report(aout, cork);
310 * Report change of output device.
312 static inline void aout_DeviceReport(audio_output_t *aout, const char *id)
314 aout->event.device_report(aout, id);
318 * Request a change of software audio amplification.
319 * \param gain linear amplitude gain (must be positive)
320 * \warning Values in excess 1.0 may cause overflow and distorsion.
322 static inline int aout_GainRequest(audio_output_t *aout, float gain)
324 return aout->event.gain_request(aout, gain);
327 VLC_API int aout_ChannelsRestart( vlc_object_t *, const char *, vlc_value_t, vlc_value_t, void * );
330 VLC_API vout_thread_t * aout_filter_RequestVout( filter_t *, vout_thread_t *p_vout, video_format_t *p_fmt ) VLC_USED;
332 /** Audio output buffer FIFO */
340 /* Legacy packet-oriented audio output helpers */
344 audio_sample_format_t format;
345 aout_fifo_t partial; /**< Audio blocks before packetization */
346 aout_fifo_t fifo; /**< Packetized audio blocks */
347 mtime_t pause_date; /**< Date when paused or VLC_TS_INVALID */
348 unsigned samples; /**< Samples per packet */
349 bool starving; /**< Whether currently starving (to limit error messages) */
352 VLC_DEPRECATED void aout_PacketInit(audio_output_t *, aout_packet_t *, unsigned, const audio_sample_format_t *);
353 VLC_DEPRECATED void aout_PacketDestroy(audio_output_t *);
355 VLC_DEPRECATED int aout_PacketTimeGet(audio_output_t *, mtime_t *);
356 VLC_DEPRECATED void aout_PacketPlay(audio_output_t *, block_t *);
357 VLC_DEPRECATED void aout_PacketFlush(audio_output_t *, bool);
359 VLC_DEPRECATED block_t *aout_PacketNext(audio_output_t *, mtime_t) VLC_USED;
362 #endif /* VLC_AOUT_H */