view libmpcodecs/ae_pcm.c @ 33109:01b19cf2649c

Fix segfault in lavcac3enc audio filter. The FFmpeg ac3 encoder would fail to open if the correct sample format is not set. As the opening is done in control() the audio filter would not fail back, but would instead continue and call encoding functions that dereference NULL pointer.

/*
 * This file is part of MPlayer.
 *
 * MPlayer 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.
 *
 * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <stdio.h>
#include <stdlib.h>
#include <inttypes.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include "m_option.h"
#include "mp_msg.h"
#include "libmpdemux/aviheader.h"
#include "libaf/af_format.h"
#include "libaf/reorder_ch.h"
#include "libmpdemux/ms_hdr.h"
#include "stream/stream.h"
#include "libmpdemux/muxer.h"
#include "ae_pcm.h"


static int bind_pcm(audio_encoder_t *encoder, muxer_stream_t *mux_a)
{
	mux_a->h.dwScale=1;
	mux_a->h.dwRate=encoder->params.sample_rate;
	mux_a->wf=malloc(sizeof(*mux_a->wf));
	mux_a->wf->wFormatTag=0x1; // PCM
	mux_a->wf->nChannels=encoder->params.channels;
	mux_a->h.dwSampleSize=2*mux_a->wf->nChannels;
	mux_a->wf->nBlockAlign=mux_a->h.dwSampleSize;
	mux_a->wf->nSamplesPerSec=mux_a->h.dwRate;
	mux_a->wf->nAvgBytesPerSec=mux_a->h.dwSampleSize*mux_a->wf->nSamplesPerSec;
	mux_a->wf->wBitsPerSample=16;
	mux_a->wf->cbSize=0; // FIXME for l3codeca.acm

	encoder->input_format = (mux_a->wf->wBitsPerSample==8) ? AF_FORMAT_U8 : AF_FORMAT_S16_LE;
	encoder->min_buffer_size = 16384;
	encoder->max_buffer_size = mux_a->wf->nAvgBytesPerSec;

	return 1;
}

static int encode_pcm(audio_encoder_t *encoder, uint8_t *dest, void *src, int nsamples, int max_size)
{
	max_size = FFMIN(nsamples, max_size);
	if (encoder->params.channels == 5 || encoder->params.channels == 6 ||
		    encoder->params.channels == 8) {
		max_size -= max_size % (encoder->params.channels * 2);
		reorder_channel_copy_nch(src, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
		                         dest, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
		                         encoder->params.channels,
		                         max_size / 2, 2);
	}
	else
	memcpy(dest, src, max_size);
	return max_size;
}

static int set_decoded_len(audio_encoder_t *encoder, int len)
{
	return len;
}

static int close_pcm(audio_encoder_t *encoder)
{
	return 1;
}

static int get_frame_size(audio_encoder_t *encoder)
{
	return 0;
}

int mpae_init_pcm(audio_encoder_t *encoder)
{
	encoder->params.samples_per_frame = encoder->params.sample_rate;
	encoder->params.bitrate = encoder->params.sample_rate * encoder->params.channels * 2 * 8;

	encoder->decode_buffer_size = encoder->params.bitrate / 8;
	encoder->bind = bind_pcm;
	encoder->get_frame_size = get_frame_size;
	encoder->set_decoded_len = set_decoded_len;
	encoder->encode = encode_pcm;
	encoder->close = close_pcm;

	return 1;
}