view libmpcodecs/ae_pcm.c @ 29507:fc8416cffdcd

Use a buffer of about half a second, instead of sizing it to have a constant number of frames. This improves the behaviour at very small or large sample rates, and gets rid of lots of obsolete code.
author cladisch
date Mon, 24 Aug 2009 07:32:25 +0000
parents 0f1b5b68af32
children 4eae69f3f4f4
line wrap: on
line source

#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(WAVEFORMATEX));
	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 == 6 || encoder->params.channels == 5) {
		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;
}