# include "config.h"
#endif
+#include <assert.h>
+
#include <vlc_common.h>
#include <vlc_aout.h>
#include <vlc_codec.h>
{
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;
#define BLOCK_FLAG_PRIVATE_REALLOCATED (1 << BLOCK_FLAG_PRIVATE_SHIFT)
static void SetupOutputFormat( decoder_t *p_dec, bool b_trust );
+static int GetAudioBuf( struct AVCodecContext *, AVFrame * );
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 )
+ int i_offset = 0;
+ int i_size = p_dec->fmt_in.i_extra;
+
+ 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 */
if( i_size < 36 )
i_size = 0;
}
- else
- {
- i_offset = 0;
- i_size = p_dec->fmt_in.i_extra;
- }
if( i_size > 0 )
{
p_codec->type = AVMEDIA_TYPE_AUDIO;
p_context->codec_type = AVMEDIA_TYPE_AUDIO;
p_context->codec_id = i_codec_id;
+ p_context->get_buffer = GetAudioBuf;
p_sys->p_context = p_context;
p_sys->p_codec = p_codec;
p_sys->i_codec_id = i_codec_id;
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;
- case CODEC_ID_WMAPRO:
- p_sys->i_output_max = 8 * sizeof(float) * 6144; /* (1 << 12) * 3/2 */
- break;
- 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;
return VLC_SUCCESS;
}
-/*****************************************************************************
- * SplitBuffer: Needed because aout really doesn't like big audio chunk and
- * wma produces easily > 30000 samples...
- *****************************************************************************/
-static block_t *SplitBuffer( decoder_t *p_dec )
+/**
+ * Allocates decoded audio buffer for libavcodec to use.
+ */
+static int GetAudioBuf( AVCodecContext *ctx, AVFrame *buf )
{
- decoder_sys_t *p_sys = p_dec->p_sys;
- int i_samples = __MIN( p_sys->i_samples, 4096 );
- int sample_planar=0;
- block_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;
-
- sample_planar = av_sample_fmt_is_planar( p_sys->p_context->sample_fmt );
- if( sample_planar )
- aout_Interleave( p_buffer->p_buffer, p_sys->p_samples, i_samples, p_sys->p_context->channels, p_dec->fmt_out.audio.i_format );
-
- if( p_sys->b_extract )
+ block_t *block;
+ bool planar = av_sample_fmt_is_planar( ctx->sample_fmt );
+ unsigned channels = planar ? 1 : ctx->channels;
+ unsigned planes = planar ? ctx->channels : 1;
+
+ int bytes = av_samples_get_buffer_size( &buf->linesize[0], channels,
+ buf->nb_samples, ctx->sample_fmt,
+ 16 );
+ assert( bytes >= 0 );
+ block = block_Alloc( bytes * planes );
+ if( unlikely(block == NULL) )
+ return AVERROR(ENOMEM);
+
+ block->i_nb_samples = buf->nb_samples;
+ buf->opaque = block;
+
+ if( planes > AV_NUM_DATA_POINTERS )
{
- if( sample_planar )
- memcpy( p_sys->p_samples, p_buffer->p_buffer, p_buffer->i_buffer );
-
- 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 );
+ uint8_t **ext = malloc( sizeof( *ext ) * planes );
+ if( unlikely(ext == NULL) )
+ {
+ block_Release( block );
+ return AVERROR(ENOMEM);
+ }
+ buf->extended_data = ext;
}
- else if (!sample_planar)
- 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;
+ else
+ buf->extended_data = buf->data;
+ uint8_t *buffer = block->p_buffer;
+ for( unsigned i = 0; i < planes; i++ )
+ {
+ buf->linesize[i] = buf->linesize[0];
+ buf->extended_data[i] = buffer;
+ buffer += bytes;
+ }
- return p_buffer;
+ return 0;
}
/*****************************************************************************
block_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
- int i_used, i_output;
- block_t *p_buffer;
- block_t *p_block;
- AVPacket pkt;
+ AVCodecContext *ctx = p_sys->p_context;
- if( !pp_block || !*pp_block ) return NULL;
+ if( !pp_block || !*pp_block )
+ return NULL;
- p_block = *pp_block;
+ block_t *p_block = *pp_block;
- if( !p_sys->p_context->extradata_size && p_dec->fmt_in.i_extra &&
- p_sys->b_delayed_open)
+ if( !ctx->extradata_size && p_dec->fmt_in.i_extra && p_sys->b_delayed_open)
{
- InitDecoderConfig( p_dec, p_sys->p_context);
+ InitDecoderConfig( p_dec, ctx );
if( ffmpeg_OpenCodec( p_dec ) )
msg_Err( p_dec, "Cannot open decoder %s", p_sys->psz_namecodec );
}
if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
- avcodec_flush_buffers( p_sys->p_context );
- p_sys->i_samples = 0;
+ avcodec_flush_buffers( ctx );
date_Set( &p_sys->end_date, 0 );
- if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 )
+ if( p_sys->i_codec_id == AV_CODEC_ID_MP2 || p_sys->i_codec_id == AV_CODEC_ID_MP3 )
p_sys->i_reject_count = 3;
goto end;
}
- if( p_sys->i_samples > 0 )
- {
- /* More data */
- p_buffer = SplitBuffer( p_dec );
- if( !p_buffer ) block_Release( p_block );
- return p_buffer;
- }
-
/* We've just started the stream, wait for the first PTS. */
- if( !date_Get( &p_sys->end_date ) && !p_block->i_pts )
+ if( !date_Get( &p_sys->end_date ) && p_block->i_pts <= VLC_TS_INVALID )
goto end;
if( p_block->i_buffer <= 0 )
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 );
+ 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;
p_block->i_flags |= BLOCK_FLAG_PRIVATE_REALLOCATED;
}
- do
+ AVFrame frame;
+ memset( &frame, 0, sizeof( frame ) );
+
+ for( int got_frame = 0; !got_frame; )
{
- 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 );
- }
+ if( p_block->i_buffer == 0 )
+ goto end;
+ AVPacket pkt;
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 )
+ int used = avcodec_decode_audio4( ctx, &frame, &got_frame, &pkt );
+ if( used < 0 )
{
- if( i_used < 0 )
- msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
- p_block->i_buffer );
-
+ msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
+ p_block->i_buffer );
goto end;
}
- 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 );
+ assert( p_block->i_buffer >= (unsigned)used );
+ p_block->p_buffer += used;
+ p_block->i_buffer -= used;
+ }
- if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 8 ||
- p_sys->p_context->sample_rate <= 0 )
+ if( ctx->channels <= 0 || ctx->channels > 8 || ctx->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 );
+ ctx->channels, ctx->sample_rate );
goto end;
}
- if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate )
+ if( p_dec->fmt_out.audio.i_rate != (unsigned int)ctx->sample_rate )
+ date_Init( &p_sys->end_date, ctx->sample_rate, 1 );
+
+ if( p_block->i_pts > VLC_TS_INVALID &&
+ p_block->i_pts > date_Get( &p_sys->end_date ) )
{
- 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 );
+ if( p_block->i_buffer == 0 )
+ { /* Done with this buffer */
+ block_Release( p_block );
+ *pp_block = NULL;
}
- 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;
+ /* NOTE WELL: Beyond this point, p_block now refers to the DECODED block */
+ p_block = frame.opaque;
+ SetupOutputFormat( p_dec, true );
/* Silent unwanted samples */
if( p_sys->i_reject_count > 0 )
{
- memset( p_sys->p_output, 0, i_output );
+ memset( p_block->p_buffer, 0, p_block->i_buffer );
p_sys->i_reject_count--;
}
- p_buffer = SplitBuffer( p_dec );
- if( !p_buffer ) block_Release( p_block );
- return p_buffer;
+ block_t *p_buffer = decoder_NewAudioBuffer( p_dec, p_block->i_nb_samples );
+ if (!p_buffer)
+ return NULL;
+ assert( p_block->i_nb_samples >= (unsigned)frame.nb_samples );
+ assert( p_block->i_nb_samples == p_buffer->i_nb_samples );
+ p_block->i_buffer = p_buffer->i_buffer; /* drop buffer padding */
+
+ /* Interleave audio if required */
+ if( av_sample_fmt_is_planar( ctx->sample_fmt ) )
+ {
+ aout_Interleave( p_buffer->p_buffer, p_block->p_buffer,
+ p_block->i_nb_samples, ctx->channels,
+ p_dec->fmt_out.audio.i_format );
+ if( ctx->channels > AV_NUM_DATA_POINTERS )
+ free( frame.extended_data );
+ block_Release( p_block );
+ p_block = p_buffer;
+ }
+ else /* FIXME: improve decoder_NewAudioBuffer(), avoid useless buffer... */
+ block_Release( p_buffer );
+
+ if (p_sys->b_extract)
+ { /* TODO: do not drop channels... at least not here */
+ p_buffer = block_Alloc( p_dec->fmt_out.audio.i_bytes_per_frame
+ * frame.nb_samples );
+ if( unlikely(p_buffer == NULL) )
+ {
+ block_Release( p_block );
+ return NULL;
+ }
+ aout_ChannelExtract( p_buffer->p_buffer,
+ p_dec->fmt_out.audio.i_channels,
+ p_block->p_buffer, ctx->channels,
+ frame.nb_samples, p_sys->pi_extraction,
+ p_dec->fmt_out.audio.i_bitspersample );
+ block_Release( p_block );
+ p_block = p_buffer;
+ }
+
+ p_block->i_nb_samples = frame.nb_samples;
+ p_block->i_buffer = frame.nb_samples
+ * p_dec->fmt_out.audio.i_bytes_per_frame;
+ p_block->i_pts = date_Get( &p_sys->end_date );
+ p_block->i_length = date_Increment( &p_sys->end_date, frame.nb_samples )
+ - p_block->i_pts;
+ return p_block;
end:
block_Release(p_block);
+ *pp_block = NULL;
return NULL;
}
-/*****************************************************************************
- * 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 );
-}
-
/*****************************************************************************
*
*****************************************************************************/
projects / vlc / blobdiff