mirror of
https://code.videolan.org/videolan/vlc
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301 lines
11 KiB
C
301 lines
11 KiB
C
/*****************************************************************************
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* vlc_aout.h : audio output interface
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*****************************************************************************
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* Copyright (C) 2002-2011 VLC authors and VideoLAN
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*
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* Authors: Christophe Massiot <massiot@via.ecp.fr>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU Lesser General Public License as published by
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* the Free Software Foundation; either version 2.1 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
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*****************************************************************************/
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#ifndef VLC_AOUT_H
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#define VLC_AOUT_H 1
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/**
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* \file
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* This file defines functions, structures and macros for audio output object
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*/
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/* Buffers which arrive in advance of more than AOUT_MAX_ADVANCE_TIME
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* will be considered as bogus and be trashed */
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#define AOUT_MAX_ADVANCE_TIME (AOUT_MAX_PREPARE_TIME + CLOCK_FREQ)
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/* Buffers which arrive in advance of more than AOUT_MAX_PREPARE_TIME
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* will cause the calling thread to sleep */
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#define AOUT_MAX_PREPARE_TIME (2 * CLOCK_FREQ)
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/* Buffers which arrive after pts - AOUT_MIN_PREPARE_TIME will be trashed
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* to avoid too heavy resampling */
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#define AOUT_MIN_PREPARE_TIME AOUT_MAX_PTS_ADVANCE
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/* Tolerance values from EBU Recommendation 37 */
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/** Maximum advance of actual audio playback time to coded PTS,
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* above which downsampling will be performed */
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#define AOUT_MAX_PTS_ADVANCE (CLOCK_FREQ / 25)
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/** Maximum delay of actual audio playback time from coded PTS,
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* above which upsampling will be performed */
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#define AOUT_MAX_PTS_DELAY (3 * CLOCK_FREQ / 50)
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/* Max acceptable resampling (in %) */
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#define AOUT_MAX_RESAMPLING 10
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#include "vlc_es.h"
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#define AOUT_FMTS_IDENTICAL( p_first, p_second ) ( \
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((p_first)->i_format == (p_second)->i_format) \
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&& AOUT_FMTS_SIMILAR(p_first, p_second) )
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/* Check if i_rate == i_rate and i_channels == i_channels */
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#define AOUT_FMTS_SIMILAR( p_first, p_second ) ( \
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((p_first)->i_rate == (p_second)->i_rate) \
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&& ((p_first)->i_physical_channels == (p_second)->i_physical_channels)\
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&& ((p_first)->i_original_channels == (p_second)->i_original_channels) )
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#define AOUT_FMT_LINEAR( p_format ) \
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(aout_BitsPerSample((p_format)->i_format) != 0)
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#define VLC_CODEC_SPDIFL VLC_FOURCC('s','p','d','i')
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#define VLC_CODEC_SPDIFB VLC_FOURCC('s','p','d','b')
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#define AOUT_FMT_SPDIF( p_format ) \
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( ((p_format)->i_format == VLC_CODEC_SPDIFL) \
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|| ((p_format)->i_format == VLC_CODEC_SPDIFB) \
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|| ((p_format)->i_format == VLC_CODEC_A52) \
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|| ((p_format)->i_format == VLC_CODEC_DTS) )
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/* This is heavily borrowed from libmad, by Robert Leslie <rob@mars.org> */
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/*
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* Fixed-point format: 0xABBBBBBB
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* A == whole part (sign + 3 bits)
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* B == fractional part (28 bits)
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*
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* Values are signed two's complement, so the effective range is:
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* 0x80000000 to 0x7fffffff
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* -8.0 to +7.9999999962747097015380859375
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*
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* The smallest representable value is:
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* 0x00000001 == 0.0000000037252902984619140625 (i.e. about 3.725e-9)
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*
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* 28 bits of fractional accuracy represent about
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* 8.6 digits of decimal accuracy.
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*
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* Fixed-point numbers can be added or subtracted as normal
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* integers, but multiplication requires shifting the 64-bit result
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* from 56 fractional bits back to 28 (and rounding.)
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*/
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typedef int32_t vlc_fixed_t;
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#define FIXED32_FRACBITS 28
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#define FIXED32_MIN ((vlc_fixed_t) -0x80000000L)
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#define FIXED32_MAX ((vlc_fixed_t) +0x7fffffffL)
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#define FIXED32_ONE ((vlc_fixed_t) 0x10000000)
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/* Values used for the audio-device and audio-channels object variables */
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#define AOUT_VAR_MONO 1
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#define AOUT_VAR_STEREO 2
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#define AOUT_VAR_2F2R 4
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#define AOUT_VAR_3F2R 5
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#define AOUT_VAR_5_1 6
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#define AOUT_VAR_6_1 7
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#define AOUT_VAR_7_1 8
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#define AOUT_VAR_SPDIF 10
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#define AOUT_VAR_CHAN_UNSET 0 /* must be zero */
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#define AOUT_VAR_CHAN_STEREO 1
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#define AOUT_VAR_CHAN_RSTEREO 2
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#define AOUT_VAR_CHAN_LEFT 3
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#define AOUT_VAR_CHAN_RIGHT 4
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#define AOUT_VAR_CHAN_DOLBYS 5
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/*****************************************************************************
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* Main audio output structures
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*****************************************************************************/
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/* Size of a frame for S/PDIF output. */
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#define AOUT_SPDIF_SIZE 6144
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/* Number of samples in an A/52 frame. */
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#define A52_FRAME_NB 1536
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/* FIXME to remove once aout.h is cleaned a bit more */
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#include <vlc_block.h>
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/** Audio output object */
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struct audio_output
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{
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VLC_COMMON_MEMBERS
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struct aout_sys_t *sys; /**< Output plugin private data */
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int (*start) (audio_output_t *, audio_sample_format_t *);
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void (*stop) (audio_output_t *);
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void (*play)(audio_output_t *, block_t *, mtime_t *); /**< Play callback
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- queue a block for playback */
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void (*pause)( audio_output_t *, bool, mtime_t ); /**< Pause/resume
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callback (optional, may be NULL) */
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void (*flush)( audio_output_t *, bool ); /**< Flush/drain callback
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(optional, may be NULL) */
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int (*volume_set)(audio_output_t *, float); /**< Volume setter (or NULL) */
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int (*mute_set)(audio_output_t *, bool); /**< Mute setter (or NULL) */
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struct {
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void (*volume_report)(audio_output_t *, float);
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void (*mute_report)(audio_output_t *, bool);
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void (*policy_report)(audio_output_t *, bool);
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int (*gain_request)(audio_output_t *, float);
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} event;
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};
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/**
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* It describes the audio channel order VLC expect.
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*/
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static const uint32_t pi_vlc_chan_order_wg4[] =
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{
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AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT,
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AOUT_CHAN_MIDDLELEFT, AOUT_CHAN_MIDDLERIGHT,
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AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT, AOUT_CHAN_REARCENTER,
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AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0
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};
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/*****************************************************************************
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* Prototypes
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*****************************************************************************/
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/**
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* This function computes the reordering needed to go from pi_chan_order_in to
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* pi_chan_order_out.
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* If pi_chan_order_in or pi_chan_order_out is NULL, it will assume that vlc
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* internal (WG4) order is requested.
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*/
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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 );
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VLC_API void aout_ChannelReorder( void *, size_t, unsigned, const int *, unsigned );
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/**
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* This fonction will compute the extraction parameter into pi_selection to go
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* from i_channels with their type given by pi_order_src[] into the order
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* describe by pi_order_dst.
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* It will also set :
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* - *pi_channels as the number of channels that will be extracted which is
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* lower (in case of non understood channels type) or equal to i_channels.
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* - the layout of the channels (*pi_layout).
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*
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* It will return true if channel extraction is really needed, in which case
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* aout_ChannelExtract must be used
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*
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* XXX It must be used when the source may have channel type not understood
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* by VLC. In this case the channel type pi_order_src[] must be set to 0.
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* XXX It must also be used if multiple channels have the same type.
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*/
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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 );
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/**
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* Do the actual channels extraction using the parameters created by
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* aout_CheckChannelExtraction.
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*
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* XXX this function does not work in place (p_dst and p_src must not overlap).
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* XXX Only 8, 16, 24, 32, 64 bits per sample are supported.
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*/
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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 );
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/* */
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static inline unsigned aout_FormatNbChannels(const audio_sample_format_t *fmt)
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{
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return popcount(fmt->i_physical_channels);
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}
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VLC_API unsigned int aout_BitsPerSample( vlc_fourcc_t i_format ) VLC_USED;
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VLC_API void aout_FormatPrepare( audio_sample_format_t * p_format );
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VLC_API void aout_FormatPrint(vlc_object_t *, const char *,
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const audio_sample_format_t *);
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#define aout_FormatPrint(o, t, f) aout_FormatPrint(VLC_OBJECT(o), t, f)
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VLC_API const char * aout_FormatPrintChannels( const audio_sample_format_t * ) VLC_USED;
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VLC_API float aout_VolumeGet (audio_output_t *);
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VLC_API int aout_VolumeSet (audio_output_t *, float);
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VLC_API int aout_MuteGet (audio_output_t *);
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VLC_API int aout_MuteSet (audio_output_t *, bool);
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/**
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* Report change of configured audio volume to the core and UI.
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*/
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static inline void aout_VolumeReport(audio_output_t *aout, float volume)
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{
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aout->event.volume_report(aout, volume);
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}
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/**
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* Report change of muted flag to the core and UI.
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*/
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static inline void aout_MuteReport(audio_output_t *aout, bool mute)
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{
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aout->event.mute_report(aout, mute);
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}
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/**
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* Report audio policy status.
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* \parm cork true to request a cork, false to undo any pending cork.
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*/
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static inline void aout_PolicyReport(audio_output_t *aout, bool cork)
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{
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aout->event.policy_report(aout, cork);
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}
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/**
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* Request a change of software audio amplification.
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* \param gain linear amplitude gain (must be positive)
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* \warning Values in excess 1.0 may cause overflow and distorsion.
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*/
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static inline int aout_GainRequest(audio_output_t *aout, float gain)
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{
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return aout->event.gain_request(aout, gain);
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}
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VLC_API int aout_ChannelsRestart( vlc_object_t *, const char *, vlc_value_t, vlc_value_t, void * );
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/* */
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VLC_API vout_thread_t * aout_filter_RequestVout( filter_t *, vout_thread_t *p_vout, video_format_t *p_fmt ) VLC_USED;
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/** Audio output buffer FIFO */
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struct aout_fifo_t
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{
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block_t *p_first;
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block_t **pp_last;
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date_t end_date;
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};
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/* Legacy packet-oriented audio output helpers */
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typedef struct
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{
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vlc_mutex_t lock;
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audio_sample_format_t format;
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aout_fifo_t partial; /**< Audio blocks before packetization */
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aout_fifo_t fifo; /**< Packetized audio blocks */
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mtime_t pause_date; /**< Date when paused or VLC_TS_INVALID */
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mtime_t time_report; /**< Desynchronization estimate or VLC_TS_INVALID */
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unsigned samples; /**< Samples per packet */
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bool starving; /**< Whether currently starving (to limit error messages) */
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} aout_packet_t;
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VLC_DEPRECATED void aout_PacketInit(audio_output_t *, aout_packet_t *, unsigned, const audio_sample_format_t *);
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VLC_DEPRECATED void aout_PacketDestroy(audio_output_t *);
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VLC_DEPRECATED void aout_PacketPlay(audio_output_t *, block_t *, mtime_t *);
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VLC_DEPRECATED void aout_PacketPause(audio_output_t *, bool, mtime_t);
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VLC_DEPRECATED void aout_PacketFlush(audio_output_t *, bool);
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VLC_DEPRECATED block_t *aout_PacketNext(audio_output_t *, mtime_t) VLC_USED;
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#endif /* VLC_AOUT_H */
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