/*****************************************************************************
* 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;
}
static block_t *Extract( filter_t *p_filter, block_t *p_in_buf )
{
size_t i_out_channels = aout_FormatNbChannels( &p_filter->fmt_out.audio );
size_t i_out_size = p_in_buf->i_nb_samples
* p_filter->fmt_out.audio.i_bitspersample
* i_out_channels / 8;
block_t *p_out_buf = block_Alloc( i_out_size );
if( unlikely(p_out_buf == NULL) )
{
block_Release( p_in_buf );
return NULL;
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_dts = p_in_buf->i_dts;
p_out_buf->i_pts = p_in_buf->i_pts;
p_out_buf->i_length = p_in_buf->i_length;
static const int pi_selections[] = {
0, 1, 2, 3, 4, 5, 6, 7, 8,
};
static_assert(sizeof(pi_selections)/sizeof(int) == AOUT_CHAN_MAX,
"channel max size mismatch!");
aout_ChannelExtract( p_out_buf->p_buffer, i_out_channels,
p_in_buf->p_buffer, p_filter->fmt_in.audio.i_channels,
p_in_buf->i_nb_samples, pi_selections,
p_filter->fmt_out.audio.i_bitspersample );
return p_out_buf;
}
static block_t * ConvertAVFrame( decoder_t *p_dec, AVFrame *frame )
{
decoder_sys_t *p_sys = p_dec->p_sys;
AVCodecContext *ctx = p_sys->p_context;
block_t *p_block;
/* Interleave audio if required */
if( av_sample_fmt_is_planar( ctx->sample_fmt ) )
{
p_block = block_Alloc(frame->linesize[0] * ctx->channels);
if ( likely(p_block) )
{
const void *planes[ctx->channels];
for (int i = 0; i < ctx->channels; i++)
planes[i] = frame->extended_data[i];
aout_Interleave(p_block->p_buffer, planes, frame->nb_samples,
ctx->channels, p_dec->fmt_out.audio.i_format);
p_block->i_nb_samples = frame->nb_samples;
}
av_frame_free(&frame);
}
else
{
p_block = vlc_av_frame_Wrap(frame);
frame = NULL;
}
if (p_sys->b_extract && p_block)
{ /* TODO: do not drop channels... at least not here */
block_t *p_buffer = block_Alloc( p_dec->fmt_out.audio.i_bytes_per_frame
* p_block->i_nb_samples );
if( likely(p_buffer) )
{
aout_ChannelExtract( p_buffer->p_buffer,
p_dec->fmt_out.audio.i_channels,
p_block->p_buffer, ctx->channels,
p_block->i_nb_samples, p_sys->pi_extraction,
p_dec->fmt_out.audio.i_bitspersample );
p_buffer->i_nb_samples = p_block->i_nb_samples;
}
block_Release( p_block );
p_block = p_buffer;
}
return p_block;
}
/*****************************************************************************
* DecodeAudio: Called to decode one frame
*****************************************************************************/
block_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
AVCodecContext *ctx = p_sys->p_context;
if( !pp_block || !*pp_block )
return NULL;
block_t *p_block = *pp_block;
if( !ctx->extradata_size && p_dec->fmt_in.i_extra && p_sys->b_delayed_open)
{
InitDecoderConfig( p_dec, ctx );
if( ffmpeg_OpenCodec( p_dec ) )
msg_Err( p_dec, "Cannot open decoder %s", p_sys->psz_namecodec );
}
if( p_sys->b_delayed_open )
goto end;
if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
avcodec_flush_buffers( ctx );
date_Set( &p_sys->end_date, 0 );
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;
}
/* We've just started the stream, wait for the first PTS. */
if( !date_Get( &p_sys->end_date ) && p_block->i_pts <= VLC_TS_INVALID )
goto end;
if( p_block->i_buffer <= 0 )
goto end;
if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 )
{
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;
}
AVFrame frame;
memset( &frame, 0, sizeof( frame ) );
for( int got_frame = 0; !got_frame; )
{
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;
int used = avcodec_decode_audio4( ctx, &frame, &got_frame, &pkt );
if( used < 0 )
{
msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
p_block->i_buffer );
goto end;
}
assert( p_block->i_buffer >= (unsigned)used );
p_block->p_buffer += used;
p_block->i_buffer -= used;
}
if( ctx->channels <= 0 || ctx->channels > 8 || ctx->sample_rate <= 0 )
{
msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
ctx->channels, ctx->sample_rate );
goto end;
}
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_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;
}
/* 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_block->p_buffer, 0, p_block->i_buffer );
p_sys->i_reject_count--;
}
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;
}
static int Audio_GetOutput(decoder_t *p_dec, picture_t **pp_out_pic,
block_t **pp_out_block, bool *p_abort,
mtime_t i_timeout)
{
decoder_sys_t *p_sys = p_dec->p_sys;
mc_api_out out;
int i_ret;
(void) p_abort;
assert(!pp_out_pic && pp_out_block);
i_ret = p_sys->api->get_out(p_sys->api, &out, i_timeout);
if (i_ret != 1)
return i_ret;
if (out.type == MC_OUT_TYPE_BUF)
{
block_t *p_block = NULL;
if (!p_sys->b_has_format) {
msg_Warn(p_dec, "Buffers returned before output format is set, dropping frame");
return p_sys->api->release_out(p_sys->api, out.u.buf.i_index, false);
}
p_block = block_Alloc(out.u.buf.i_size);
if (!p_block)
return -1;
p_block->i_nb_samples = out.u.buf.i_size
/ p_dec->fmt_out.audio.i_bytes_per_frame;
if (p_sys->u.audio.b_extract)
{
aout_ChannelExtract(p_block->p_buffer,
p_dec->fmt_out.audio.i_channels,
out.u.buf.p_ptr, p_sys->u.audio.i_channels,
p_block->i_nb_samples, p_sys->u.audio.pi_extraction,
p_dec->fmt_out.audio.i_bitspersample);
}
else
memcpy(p_block->p_buffer, out.u.buf.p_ptr, out.u.buf.i_size);
if (out.u.buf.i_ts != 0 && out.u.buf.i_ts != date_Get(&p_sys->u.audio.i_end_date))
date_Set(&p_sys->u.audio.i_end_date, out.u.buf.i_ts);
p_block->i_pts = date_Get(&p_sys->u.audio.i_end_date);
p_block->i_length = date_Increment(&p_sys->u.audio.i_end_date,
p_block->i_nb_samples)
- p_block->i_pts;
if (p_sys->api->release_out(p_sys->api, out.u.buf.i_index, false))
{
block_Release(p_block);
return -1;
}
*pp_out_block = p_block;
return 1;
} else {
uint32_t i_layout_dst;
int i_channels_dst;
assert(out.type == MC_OUT_TYPE_CONF);
if (out.u.conf.audio.channel_count <= 0
|| out.u.conf.audio.channel_count > 8
|| out.u.conf.audio.sample_rate <= 0)
{
msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
out.u.conf.audio.channel_count,
out.u.conf.audio.sample_rate);
return -1;
}
msg_Err(p_dec, "output: channel_count: %d, channel_mask: 0x%X, rate: %d",
out.u.conf.audio.channel_count, out.u.conf.audio.channel_mask,
out.u.conf.audio.sample_rate);
p_dec->fmt_out.i_codec = VLC_CODEC_S16N;
p_dec->fmt_out.audio.i_format = p_dec->fmt_out.i_codec;
p_dec->fmt_out.audio.i_rate = out.u.conf.audio.sample_rate;
date_Init(&p_sys->u.audio.i_end_date, out.u.conf.audio.sample_rate, 1 );
p_sys->u.audio.i_channels = out.u.conf.audio.channel_count;
p_sys->u.audio.b_extract =
aout_CheckChannelExtraction(p_sys->u.audio.pi_extraction,
&i_layout_dst, &i_channels_dst,
NULL, pi_audio_order_src,
p_sys->u.audio.i_channels);
if (p_sys->u.audio.b_extract)
msg_Warn(p_dec, "need channel extraction: %d -> %d",
p_sys->u.audio.i_channels, i_channels_dst);
p_dec->fmt_out.audio.i_original_channels =
p_dec->fmt_out.audio.i_physical_channels = i_layout_dst;
aout_FormatPrepare(&p_dec->fmt_out.audio);
if (decoder_UpdateAudioFormat(p_dec))
return -1;
p_sys->b_has_format = true;
return 0;
}
}
/*****************************************************************************
* DecodeAudio: Called to decode one frame
*****************************************************************************/
static block_t *DecodeAudio( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
AVCodecContext *ctx = p_sys->p_context;
AVFrame *frame = NULL;
if( !pp_block || !*pp_block )
return NULL;
block_t *p_block = *pp_block;
if( !ctx->extradata_size && p_dec->fmt_in.i_extra && p_sys->b_delayed_open)
{
InitDecoderConfig( p_dec, ctx );
OpenAudioCodec( p_dec );
}
if( p_sys->b_delayed_open )
goto end;
if( p_block->i_flags & BLOCK_FLAG_CORRUPTED )
{
Flush( p_dec );
goto end;
}
if( p_block->i_flags & BLOCK_FLAG_DISCONTINUITY )
{
date_Set( &p_sys->end_date, VLC_TS_INVALID );
}
/* We've just started the stream, wait for the first PTS. */
if( !date_Get( &p_sys->end_date ) && p_block->i_pts <= VLC_TS_INVALID )
goto end;
if( p_block->i_buffer <= 0 )
goto end;
if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 )
{
p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE );
if( !p_block )
return NULL;
*pp_block = p_block;
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;
}
frame = av_frame_alloc();
if (unlikely(frame == NULL))
goto end;
for( int got_frame = 0; !got_frame; )
{
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;
int ret = avcodec_send_packet( ctx, &pkt );
if( ret != 0 && ret != AVERROR(EAGAIN) )
{
msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
p_block->i_buffer );
goto end;
}
int used = ret != AVERROR(EAGAIN) ? pkt.size : 0;
ret = avcodec_receive_frame( ctx, frame );
if( ret != 0 && ret != AVERROR(EAGAIN) )
{
msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
p_block->i_buffer );
goto end;
}
got_frame = ret == 0;
p_block->p_buffer += used;
p_block->i_buffer -= used;
}
if( ctx->channels <= 0 || ctx->channels > 8 || ctx->sample_rate <= 0 )
{
msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
ctx->channels, ctx->sample_rate );
goto end;
}
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 > 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 );
p_block = NULL;
*pp_block = NULL;
}
/* NOTE WELL: Beyond this point, p_block refers to the DECODED block! */
SetupOutputFormat( p_dec, true );
if( decoder_UpdateAudioFormat( p_dec ) )
goto drop;
/* Interleave audio if required */
if( av_sample_fmt_is_planar( ctx->sample_fmt ) )
{
p_block = block_Alloc(frame->linesize[0] * ctx->channels);
if (unlikely(p_block == NULL))
goto drop;
const void *planes[ctx->channels];
for (int i = 0; i < ctx->channels; i++)
planes[i] = frame->extended_data[i];
aout_Interleave(p_block->p_buffer, planes, frame->nb_samples,
ctx->channels, p_dec->fmt_out.audio.i_format);
p_block->i_nb_samples = frame->nb_samples;
av_frame_free(&frame);
}
else
{
p_block = vlc_av_frame_Wrap(frame);
if (unlikely(p_block == NULL))
goto drop;
frame = NULL;
}
if (p_sys->b_extract)
{ /* TODO: do not drop channels... at least not here */
block_t *p_buffer = block_Alloc( p_dec->fmt_out.audio.i_bytes_per_frame
* p_block->i_nb_samples );
if( unlikely(p_buffer == NULL) )
goto drop;
aout_ChannelExtract( p_buffer->p_buffer,
p_dec->fmt_out.audio.i_channels,
p_block->p_buffer, ctx->channels,
p_block->i_nb_samples, p_sys->pi_extraction,
p_dec->fmt_out.audio.i_bitspersample );
p_buffer->i_nb_samples = p_block->i_nb_samples;
block_Release( p_block );
p_block = p_buffer;
}
/* Silent unwanted samples */
if( p_sys->i_reject_count > 0 )
{
memset( p_block->p_buffer, 0, p_block->i_buffer );
p_sys->i_reject_count--;
}
p_block->i_buffer = p_block->i_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,
p_block->i_nb_samples ) - p_block->i_pts;
return p_block;
end:
*pp_block = NULL;
drop:
av_frame_free(&frame);
if( p_block != NULL )
block_Release(p_block);
return NULL;
}
