view libmpcodecs/ae_pcm.c @ 33672:e576232a39d5

Prevent balance from hopping. Only recalculate the balance if the balance has changed, not if just the volume has changed. Because (at least with my soundcard) not all volume values can be stored, but seem to be mapped onto a discrete value set, recalculation the balance from the volume isn't accurate enough.
author ib
date Tue, 28 Jun 2011 18:16:06 +0000
parents c08363dc5320
children
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/*
 * 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;
}