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https://github.com/FFmpeg/FFmpeg.git
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3e1a7ae44a
* qatar/master: swfdec: Add support for sample_rate_code 0 (5512 Hz) dct-test: factor out some common code and do whas was likely intended doc: library versions need to be bumped in version.h Revert "ffmpeg: get rid of useless AVInputStream.nb_streams." Remove some forgotten AVCodecContext.palctrl usage. lavc/utils: move avcodec_init() higher in the file. lavc: replace some deprecated FF_*_TYPE with AV_PICTURE_TYPE_* ac3dec: actually use drc_scale private option lavc: undeprecate AVPALETTE_SIZE and AVPALETTE_COUNT macros alsa: add missing header msmpeg4: remove leftover unused debug variable declaration Fix assert() calls that need updates after FF_COMMON_FRAME macro elimination. Fix av_dlog invocations with wrong or missing logging context. vf_yadif: add support to yuva420p vf_yadif: correct documentation on the parity parameter vf_yadif: copy buffer properties like aspect for second frame as well oma: support for encrypted files id3v2: add support for non-text and GEOB type tag frames des: add possibility to calculate DES-CBC-MAC with small buffer Conflicts: ffmpeg.c libavcodec/dct-test.c libavformat/mpegts.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
365 lines
11 KiB
C
365 lines
11 KiB
C
/*
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* ALSA input and output
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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 GNU
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* 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
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* ALSA input and output: common code
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* @author Luca Abeni ( lucabe72 email it )
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* @author Benoit Fouet ( benoit fouet free fr )
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* @author Nicolas George ( nicolas george normalesup org )
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*/
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#include <alsa/asoundlib.h>
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#include "avdevice.h"
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#include "libavutil/avassert.h"
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#include "libavutil/audioconvert.h"
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#include "alsa-audio.h"
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static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
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{
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switch(codec_id) {
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case CODEC_ID_PCM_F64LE: return SND_PCM_FORMAT_FLOAT64_LE;
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case CODEC_ID_PCM_F64BE: return SND_PCM_FORMAT_FLOAT64_BE;
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case CODEC_ID_PCM_F32LE: return SND_PCM_FORMAT_FLOAT_LE;
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case CODEC_ID_PCM_F32BE: return SND_PCM_FORMAT_FLOAT_BE;
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case CODEC_ID_PCM_S32LE: return SND_PCM_FORMAT_S32_LE;
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case CODEC_ID_PCM_S32BE: return SND_PCM_FORMAT_S32_BE;
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case CODEC_ID_PCM_U32LE: return SND_PCM_FORMAT_U32_LE;
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case CODEC_ID_PCM_U32BE: return SND_PCM_FORMAT_U32_BE;
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case CODEC_ID_PCM_S24LE: return SND_PCM_FORMAT_S24_3LE;
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case CODEC_ID_PCM_S24BE: return SND_PCM_FORMAT_S24_3BE;
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case CODEC_ID_PCM_U24LE: return SND_PCM_FORMAT_U24_3LE;
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case CODEC_ID_PCM_U24BE: return SND_PCM_FORMAT_U24_3BE;
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case CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE;
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case CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE;
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case CODEC_ID_PCM_U16LE: return SND_PCM_FORMAT_U16_LE;
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case CODEC_ID_PCM_U16BE: return SND_PCM_FORMAT_U16_BE;
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case CODEC_ID_PCM_S8: return SND_PCM_FORMAT_S8;
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case CODEC_ID_PCM_U8: return SND_PCM_FORMAT_U8;
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case CODEC_ID_PCM_MULAW: return SND_PCM_FORMAT_MU_LAW;
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case CODEC_ID_PCM_ALAW: return SND_PCM_FORMAT_A_LAW;
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default: return SND_PCM_FORMAT_UNKNOWN;
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}
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}
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#define REORDER_OUT_50(NAME, TYPE) \
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static void alsa_reorder_ ## NAME ## _out_50(const void *in_v, void *out_v, int n) \
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{ \
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const TYPE *in = in_v; \
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TYPE *out = out_v; \
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\
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while (n-- > 0) { \
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out[0] = in[0]; \
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out[1] = in[1]; \
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out[2] = in[3]; \
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out[3] = in[4]; \
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out[4] = in[2]; \
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in += 5; \
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out += 5; \
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} \
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}
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#define REORDER_OUT_51(NAME, TYPE) \
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static void alsa_reorder_ ## NAME ## _out_51(const void *in_v, void *out_v, int n) \
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{ \
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const TYPE *in = in_v; \
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TYPE *out = out_v; \
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\
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while (n-- > 0) { \
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out[0] = in[0]; \
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out[1] = in[1]; \
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out[2] = in[4]; \
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out[3] = in[5]; \
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out[4] = in[2]; \
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out[5] = in[3]; \
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in += 6; \
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out += 6; \
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} \
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}
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#define REORDER_OUT_71(NAME, TYPE) \
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static void alsa_reorder_ ## NAME ## _out_71(const void *in_v, void *out_v, int n) \
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{ \
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const TYPE *in = in_v; \
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TYPE *out = out_v; \
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\
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while (n-- > 0) { \
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out[0] = in[0]; \
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out[1] = in[1]; \
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out[2] = in[4]; \
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out[3] = in[5]; \
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out[4] = in[2]; \
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out[5] = in[3]; \
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out[6] = in[6]; \
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out[7] = in[7]; \
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in += 8; \
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out += 8; \
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} \
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}
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REORDER_OUT_50(int8, int8_t)
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REORDER_OUT_51(int8, int8_t)
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REORDER_OUT_71(int8, int8_t)
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REORDER_OUT_50(int16, int16_t)
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REORDER_OUT_51(int16, int16_t)
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REORDER_OUT_71(int16, int16_t)
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REORDER_OUT_50(int32, int32_t)
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REORDER_OUT_51(int32, int32_t)
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REORDER_OUT_71(int32, int32_t)
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REORDER_OUT_50(f32, float)
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REORDER_OUT_51(f32, float)
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REORDER_OUT_71(f32, float)
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#define FORMAT_I8 0
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#define FORMAT_I16 1
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#define FORMAT_I32 2
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#define FORMAT_F32 3
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#define PICK_REORDER(layout)\
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switch(format) {\
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case FORMAT_I8: s->reorder_func = alsa_reorder_int8_out_ ##layout; break;\
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case FORMAT_I16: s->reorder_func = alsa_reorder_int16_out_ ##layout; break;\
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case FORMAT_I32: s->reorder_func = alsa_reorder_int32_out_ ##layout; break;\
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case FORMAT_F32: s->reorder_func = alsa_reorder_f32_out_ ##layout; break;\
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}
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static av_cold int find_reorder_func(AlsaData *s, int codec_id, int64_t layout, int out)
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{
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int format;
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/* reordering input is not currently supported */
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if (!out)
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return AVERROR(ENOSYS);
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/* reordering is not needed for QUAD or 2_2 layout */
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if (layout == AV_CH_LAYOUT_QUAD || layout == AV_CH_LAYOUT_2_2)
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return 0;
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switch (codec_id) {
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case CODEC_ID_PCM_S8:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_MULAW: format = FORMAT_I8; break;
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case CODEC_ID_PCM_S16LE:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_U16BE: format = FORMAT_I16; break;
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case CODEC_ID_PCM_S32LE:
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case CODEC_ID_PCM_S32BE:
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case CODEC_ID_PCM_U32LE:
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case CODEC_ID_PCM_U32BE: format = FORMAT_I32; break;
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case CODEC_ID_PCM_F32LE:
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case CODEC_ID_PCM_F32BE: format = FORMAT_F32; break;
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default: return AVERROR(ENOSYS);
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}
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if (layout == AV_CH_LAYOUT_5POINT0_BACK || layout == AV_CH_LAYOUT_5POINT0)
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PICK_REORDER(50)
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else if (layout == AV_CH_LAYOUT_5POINT1_BACK || layout == AV_CH_LAYOUT_5POINT1)
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PICK_REORDER(51)
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else if (layout == AV_CH_LAYOUT_7POINT1)
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PICK_REORDER(71)
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return s->reorder_func ? 0 : AVERROR(ENOSYS);
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}
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av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
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unsigned int *sample_rate,
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int channels, enum CodecID *codec_id)
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{
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AlsaData *s = ctx->priv_data;
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const char *audio_device;
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int res, flags = 0;
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snd_pcm_format_t format;
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snd_pcm_t *h;
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snd_pcm_hw_params_t *hw_params;
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snd_pcm_uframes_t buffer_size, period_size;
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int64_t layout = ctx->streams[0]->codec->channel_layout;
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if (ctx->filename[0] == 0) audio_device = "default";
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else audio_device = ctx->filename;
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if (*codec_id == CODEC_ID_NONE)
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*codec_id = DEFAULT_CODEC_ID;
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format = codec_id_to_pcm_format(*codec_id);
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if (format == SND_PCM_FORMAT_UNKNOWN) {
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av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
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return AVERROR(ENOSYS);
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}
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s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
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if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
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flags = SND_PCM_NONBLOCK;
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}
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res = snd_pcm_open(&h, audio_device, mode, flags);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
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audio_device, snd_strerror(res));
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return AVERROR(EIO);
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}
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res = snd_pcm_hw_params_malloc(&hw_params);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
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snd_strerror(res));
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goto fail1;
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}
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res = snd_pcm_hw_params_any(h, hw_params);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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res = snd_pcm_hw_params_set_format(h, hw_params, format);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
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*codec_id, format, snd_strerror(res));
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goto fail;
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}
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res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
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channels, snd_strerror(res));
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goto fail;
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}
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snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
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buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
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/* TODO: maybe use ctx->max_picture_buffer somehow */
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res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
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if (!period_size)
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period_size = buffer_size / 4;
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res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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s->period_size = period_size;
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res = snd_pcm_hw_params(h, hw_params);
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if (res < 0) {
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av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
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snd_strerror(res));
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goto fail;
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}
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snd_pcm_hw_params_free(hw_params);
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if (channels > 2 && layout) {
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if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
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char name[128];
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av_get_channel_layout_string(name, sizeof(name), channels, layout);
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av_log(ctx, AV_LOG_WARNING, "ALSA channel layout unknown or unimplemented for %s %s.\n",
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name, mode == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture");
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}
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if (s->reorder_func) {
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s->reorder_buf_size = buffer_size;
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s->reorder_buf = av_malloc(s->reorder_buf_size * s->frame_size);
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if (!s->reorder_buf)
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goto fail1;
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}
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}
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s->h = h;
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return 0;
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fail:
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snd_pcm_hw_params_free(hw_params);
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fail1:
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snd_pcm_close(h);
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return AVERROR(EIO);
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}
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av_cold int ff_alsa_close(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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av_freep(&s->reorder_buf);
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if (CONFIG_ALSA_INDEV)
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ff_timefilter_destroy(s->timefilter);
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snd_pcm_close(s->h);
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return 0;
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}
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int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
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{
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AlsaData *s = s1->priv_data;
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snd_pcm_t *handle = s->h;
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av_log(s1, AV_LOG_WARNING, "ALSA buffer xrun.\n");
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if (err == -EPIPE) {
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err = snd_pcm_prepare(handle);
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if (err < 0) {
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av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err));
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return AVERROR(EIO);
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}
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} else if (err == -ESTRPIPE) {
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av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n");
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return -1;
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}
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return err;
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}
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int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
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{
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int size = s->reorder_buf_size;
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void *r;
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av_assert0(size != 0);
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while (size < min_size)
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size *= 2;
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r = av_realloc(s->reorder_buf, size * s->frame_size);
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if (!r)
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return AVERROR(ENOMEM);
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s->reorder_buf = r;
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s->reorder_buf_size = size;
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return 0;
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}
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