/***************************************************************************** * audio.c: audio decoder using ffmpeg library ***************************************************************************** * Copyright (C) 1999-2003 the VideoLAN team * $Id$ * * Authors: Laurent Aimar * Gildas Bazin * * 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., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include /* ffmpeg header */ #ifdef HAVE_LIBAVCODEC_AVCODEC_H # include #else # include #endif #include "avcodec.h" /***************************************************************************** * decoder_sys_t : decoder descriptor *****************************************************************************/ struct decoder_sys_t { AVCODEC_COMMON_MEMBERS /* Temporary buffer for libavcodec */ int i_output_max; uint8_t *p_output; /* * Output properties */ audio_sample_format_t aout_format; date_t end_date; /* * */ uint8_t *p_samples; int i_samples; /* */ int i_reject_count; /* */ bool b_extract; int pi_extraction[AOUT_CHAN_MAX]; int i_previous_channels; int64_t i_previous_layout; }; #define BLOCK_FLAG_PRIVATE_REALLOCATED (1 << BLOCK_FLAG_PRIVATE_SHIFT) static void SetupOutputFormat( decoder_t *p_dec, bool b_trust ); static void InitDecoderConfig( decoder_t *p_dec, AVCodecContext *p_context ) { if( p_dec->fmt_in.i_extra > 0 ) { const uint8_t * const p_src = p_dec->fmt_in.p_extra; int i_offset; int i_size; if( p_dec->fmt_in.i_codec == VLC_CODEC_FLAC ) { i_offset = 8; i_size = p_dec->fmt_in.i_extra - 8; } else if( p_dec->fmt_in.i_codec == VLC_CODEC_ALAC ) { static const uint8_t p_pattern[] = { 0, 0, 0, 36, 'a', 'l', 'a', 'c' }; /* Find alac atom XXX it is a bit ugly */ for( i_offset = 0; i_offset < p_dec->fmt_in.i_extra - sizeof(p_pattern); i_offset++ ) { if( !memcmp( &p_src[i_offset], p_pattern, sizeof(p_pattern) ) ) break; } i_size = __MIN( p_dec->fmt_in.i_extra - i_offset, 36 ); if( i_size < 36 ) i_size = 0; } else { i_offset = 0; i_size = p_dec->fmt_in.i_extra; } if( i_size > 0 ) { p_context->extradata = malloc( i_size + FF_INPUT_BUFFER_PADDING_SIZE ); if( p_context->extradata ) { uint8_t *p_dst = p_context->extradata; p_context->extradata_size = i_size; memcpy( &p_dst[0], &p_src[i_offset], i_size ); memset( &p_dst[i_size], 0, FF_INPUT_BUFFER_PADDING_SIZE ); } } } else { p_context->extradata_size = 0; p_context->extradata = NULL; } } /***************************************************************************** * InitAudioDec: initialize audio decoder ***************************************************************************** * The ffmpeg codec will be opened, some memory allocated. *****************************************************************************/ int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context, AVCodec *p_codec, int i_codec_id, const char *psz_namecodec ) { decoder_sys_t *p_sys; /* Allocate the memory needed to store the decoder's structure */ if( ( p_dec->p_sys = p_sys = malloc(sizeof(*p_sys)) ) == NULL ) { return VLC_ENOMEM; } p_codec->type = AVMEDIA_TYPE_AUDIO; p_context->codec_type = AVMEDIA_TYPE_AUDIO; p_context->codec_id = i_codec_id; p_sys->p_context = p_context; p_sys->p_codec = p_codec; p_sys->i_codec_id = i_codec_id; p_sys->psz_namecodec = psz_namecodec; p_sys->b_delayed_open = true; // Initialize decoder extradata InitDecoderConfig( p_dec, p_context); /* ***** Open the codec ***** */ if( ffmpeg_OpenCodec( p_dec ) < 0 ) { msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec ); free( p_sys->p_context->extradata ); free( p_sys ); return VLC_EGENERIC; } switch( i_codec_id ) { case CODEC_ID_WAVPACK: p_sys->i_output_max = 8 * sizeof(int32_t) * 131072; break; case CODEC_ID_TTA: p_sys->i_output_max = p_sys->p_context->channels * sizeof(int32_t) * p_sys->p_context->sample_rate * 2; break; case CODEC_ID_FLAC: p_sys->i_output_max = 8 * sizeof(int32_t) * 65535; break; #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 52, 35, 0 ) case CODEC_ID_WMAPRO: p_sys->i_output_max = 8 * sizeof(float) * 6144; /* (1 << 12) * 3/2 */ break; #endif default: p_sys->i_output_max = 0; break; } if( p_sys->i_output_max < AVCODEC_MAX_AUDIO_FRAME_SIZE ) p_sys->i_output_max = AVCODEC_MAX_AUDIO_FRAME_SIZE; msg_Dbg( p_dec, "Using %d bytes output buffer", p_sys->i_output_max ); p_sys->p_output = av_malloc( p_sys->i_output_max ); p_sys->p_samples = NULL; p_sys->i_samples = 0; p_sys->i_reject_count = 0; p_sys->b_extract = false; p_sys->i_previous_channels = 0; p_sys->i_previous_layout = 0; /* */ p_dec->fmt_out.i_cat = AUDIO_ES; /* Try to set as much information as possible but do not trust it */ SetupOutputFormat( p_dec, false ); date_Set( &p_sys->end_date, 0 ); if( p_dec->fmt_out.audio.i_rate ) date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate, 1 ); else if( p_dec->fmt_in.audio.i_rate ) date_Init( &p_sys->end_date, p_dec->fmt_in.audio.i_rate, 1 ); return VLC_SUCCESS; } /***************************************************************************** * SplitBuffer: Needed because aout really doesn't like big audio chunk and * wma produces easily > 30000 samples... *****************************************************************************/ static aout_buffer_t *SplitBuffer( decoder_t *p_dec ) { decoder_sys_t *p_sys = p_dec->p_sys; int i_samples = __MIN( p_sys->i_samples, 4096 ); aout_buffer_t *p_buffer; if( i_samples == 0 ) return NULL; if( ( p_buffer = decoder_NewAudioBuffer( p_dec, i_samples ) ) == NULL ) return NULL; p_buffer->i_pts = date_Get( &p_sys->end_date ); p_buffer->i_length = date_Increment( &p_sys->end_date, i_samples ) - p_buffer->i_pts; if( p_sys->b_extract ) aout_ChannelExtract( p_buffer->p_buffer, p_dec->fmt_out.audio.i_channels, p_sys->p_samples, p_sys->p_context->channels, i_samples, p_sys->pi_extraction, p_dec->fmt_out.audio.i_bitspersample ); else memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_buffer ); p_sys->p_samples += i_samples * p_sys->p_context->channels * ( p_dec->fmt_out.audio.i_bitspersample / 8 ); p_sys->i_samples -= i_samples; return p_buffer; } /***************************************************************************** * DecodeAudio: Called to decode one frame *****************************************************************************/ aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block ) { decoder_sys_t *p_sys = p_dec->p_sys; int i_used, i_output; aout_buffer_t *p_buffer; block_t *p_block; AVPacket pkt; if( !pp_block || !*pp_block ) return NULL; p_block = *pp_block; if( !p_sys->p_context->extradata_size && p_dec->fmt_in.i_extra && p_sys->b_delayed_open) { InitDecoderConfig( p_dec, p_sys->p_context); if( ffmpeg_OpenCodec( p_dec ) ) msg_Err( p_dec, "Cannot open decoder %s", p_sys->psz_namecodec ); } if( p_sys->b_delayed_open ) { block_Release( p_block ); return NULL; } if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) ) { block_Release( p_block ); avcodec_flush_buffers( p_sys->p_context ); p_sys->i_samples = 0; date_Set( &p_sys->end_date, 0 ); if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 ) p_sys->i_reject_count = 3; return NULL; } if( p_sys->i_samples > 0 ) { /* More data */ p_buffer = SplitBuffer( p_dec ); if( !p_buffer ) block_Release( p_block ); return p_buffer; } if( !date_Get( &p_sys->end_date ) && !p_block->i_pts ) { /* We've just started the stream, wait for the first PTS. */ block_Release( p_block ); return NULL; } if( p_block->i_buffer <= 0 ) { block_Release( p_block ); return NULL; } if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 ) { *pp_block = p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE ); if( !p_block ) return NULL; p_block->i_buffer -= FF_INPUT_BUFFER_PADDING_SIZE; memset( &p_block->p_buffer[p_block->i_buffer], 0, FF_INPUT_BUFFER_PADDING_SIZE ); p_block->i_flags |= BLOCK_FLAG_PRIVATE_REALLOCATED; } do { i_output = __MAX( p_block->i_buffer, p_sys->i_output_max ); if( i_output > p_sys->i_output_max ) { /* Grow output buffer if necessary (eg. for PCM data) */ p_sys->p_output = av_realloc( p_sys->p_output, i_output ); } av_init_packet( &pkt ); pkt.data = p_block->p_buffer; pkt.size = p_block->i_buffer; i_used = avcodec_decode_audio3( p_sys->p_context, (int16_t*)p_sys->p_output, &i_output, &pkt ); if( i_used < 0 || i_output < 0 ) { if( i_used < 0 ) msg_Warn( p_dec, "cannot decode one frame (%zu bytes)", p_block->i_buffer ); block_Release( p_block ); return NULL; } else if( (size_t)i_used > p_block->i_buffer ) { i_used = p_block->i_buffer; } p_block->i_buffer -= i_used; p_block->p_buffer += i_used; } while( p_block->i_buffer > 0 && i_output <= 0 ); if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 8 || p_sys->p_context->sample_rate <= 0 ) { msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d", p_sys->p_context->channels, p_sys->p_context->sample_rate ); block_Release( p_block ); return NULL; } if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate ) { date_Init( &p_sys->end_date, p_sys->p_context->sample_rate, 1 ); date_Set( &p_sys->end_date, p_block->i_pts ); } /* **** Set audio output parameters **** */ SetupOutputFormat( p_dec, true ); if( p_block->i_pts != 0 && p_block->i_pts != date_Get( &p_sys->end_date ) ) { date_Set( &p_sys->end_date, p_block->i_pts ); } p_block->i_pts = 0; /* **** Now we can output these samples **** */ p_sys->i_samples = i_output / (p_dec->fmt_out.audio.i_bitspersample / 8) / p_sys->p_context->channels; p_sys->p_samples = p_sys->p_output; /* Silent unwanted samples */ if( p_sys->i_reject_count > 0 ) { memset( p_sys->p_output, 0, i_output ); p_sys->i_reject_count--; } p_buffer = SplitBuffer( p_dec ); if( !p_buffer ) block_Release( p_block ); return p_buffer; } /***************************************************************************** * EndAudioDec: audio decoder destruction *****************************************************************************/ void EndAudioDec( decoder_t *p_dec ) { decoder_sys_t *p_sys = p_dec->p_sys; av_free( p_sys->p_output ); } /***************************************************************************** * *****************************************************************************/ void GetVlcAudioFormat( vlc_fourcc_t *pi_codec, unsigned *pi_bits, int i_sample_fmt ) { switch( i_sample_fmt ) { case SAMPLE_FMT_U8: *pi_codec = VLC_CODEC_U8; *pi_bits = 8; break; case SAMPLE_FMT_S32: *pi_codec = VLC_CODEC_S32N; *pi_bits = 32; break; case SAMPLE_FMT_FLT: *pi_codec = VLC_CODEC_FL32; *pi_bits = 32; break; case SAMPLE_FMT_DBL: *pi_codec = VLC_CODEC_FL64; *pi_bits = 64; break; case SAMPLE_FMT_S16: default: *pi_codec = VLC_CODEC_S16N; *pi_bits = 16; break; } } static const uint64_t pi_channels_map[][2] = { { CH_FRONT_LEFT, AOUT_CHAN_LEFT }, { CH_FRONT_RIGHT, AOUT_CHAN_RIGHT }, { CH_FRONT_CENTER, AOUT_CHAN_CENTER }, { CH_LOW_FREQUENCY, AOUT_CHAN_LFE }, { CH_BACK_LEFT, AOUT_CHAN_REARLEFT }, { CH_BACK_RIGHT, AOUT_CHAN_REARRIGHT }, { CH_FRONT_LEFT_OF_CENTER, 0 }, { CH_FRONT_RIGHT_OF_CENTER, 0 }, { CH_BACK_CENTER, AOUT_CHAN_REARCENTER }, { CH_SIDE_LEFT, AOUT_CHAN_MIDDLELEFT }, { CH_SIDE_RIGHT, AOUT_CHAN_MIDDLERIGHT }, { CH_TOP_CENTER, 0 }, { CH_TOP_FRONT_LEFT, 0 }, { CH_TOP_FRONT_CENTER, 0 }, { CH_TOP_FRONT_RIGHT, 0 }, { CH_TOP_BACK_LEFT, 0 }, { CH_TOP_BACK_CENTER, 0 }, { CH_TOP_BACK_RIGHT, 0 }, { CH_STEREO_LEFT, 0 }, { CH_STEREO_RIGHT, 0 }, }; static void SetupOutputFormat( decoder_t *p_dec, bool b_trust ) { decoder_sys_t *p_sys = p_dec->p_sys; GetVlcAudioFormat( &p_dec->fmt_out.i_codec, &p_dec->fmt_out.audio.i_bitspersample, p_sys->p_context->sample_fmt ); p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate; /* */ if( p_sys->i_previous_channels == p_sys->p_context->channels && p_sys->i_previous_layout == p_sys->p_context->channel_layout ) return; if( b_trust ) { p_sys->i_previous_channels = p_sys->p_context->channels; p_sys->i_previous_layout = p_sys->p_context->channel_layout; } /* Specified order * FIXME should we use fmt_in.audio.i_physical_channels or not ? */ const unsigned i_order_max = 8 * sizeof(p_sys->p_context->channel_layout); uint32_t pi_order_src[i_order_max]; int i_channels_src = 0; if( p_sys->p_context->channel_layout ) { for( unsigned i = 0; i < sizeof(pi_channels_map)/sizeof(*pi_channels_map); i++ ) { if( p_sys->p_context->channel_layout & pi_channels_map[i][0] ) pi_order_src[i_channels_src++] = pi_channels_map[i][1]; } } else { /* Create default order */ if( b_trust ) msg_Warn( p_dec, "Physical channel configuration not set : guessing" ); for( unsigned int i = 0; i < __MIN( i_order_max, (unsigned)p_sys->p_context->channels ); i++ ) { if( i < sizeof(pi_channels_map)/sizeof(*pi_channels_map) ) pi_order_src[i_channels_src++] = pi_channels_map[i][1]; } } if( i_channels_src != p_sys->p_context->channels && b_trust ) msg_Err( p_dec, "Channel layout not understood" ); uint32_t i_layout_dst; int i_channels_dst; p_sys->b_extract = aout_CheckChannelExtraction( p_sys->pi_extraction, &i_layout_dst, &i_channels_dst, NULL, pi_order_src, i_channels_src ); if( i_channels_dst != i_channels_src && b_trust ) msg_Warn( p_dec, "%d channels are dropped", i_channels_src - i_channels_dst ); p_dec->fmt_out.audio.i_physical_channels = p_dec->fmt_out.audio.i_original_channels = i_layout_dst; p_dec->fmt_out.audio.i_channels = i_channels_dst; }