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libstb-hal/libeplayer3/output/writer/pcm.cpp
martii 97f1aff64f libeplayer3: rename to c++
2014-04-05 16:01:12 +02:00

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/*
* linuxdvb output/writer handling.
*
* konfetti 2010 based on linuxdvb.c code from libeplayer2
*
* This program 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.
*
* This program 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 this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/* ***************************** */
/* Includes */
/* ***************************** */
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>
#include <sys/uio.h>
#include <linux/dvb/video.h>
#include <linux/dvb/audio.h>
#include <linux/dvb/stm_ioctls.h>
#include <memory.h>
#include <asm/types.h>
#include <pthread.h>
#include <errno.h>
#include "common.h"
#include "output.h"
#include "debug.h"
#include "misc.h"
#include "pes.h"
#include "writer.h"
/* ***************************** */
/* Makros/Constants */
/* ***************************** */
#define PCM_DEBUG
#ifdef PCM_DEBUG
static short debug_level = 0;
#define pcm_printf(level, fmt, x...) do { \
if (debug_level >= level) printf("[%s:%s] " fmt, __FILE__, __FUNCTION__, ## x); } while (0)
#else
#define pcm_printf(level, fmt, x...)
#endif
#ifndef PCM_SILENT
#define pcm_err(fmt, x...) do { printf("[%s:%s] " fmt, __FILE__, __FUNCTION__, ## x); } while (0)
#else
#define pcm_err(fmt, x...)
#endif
/* ***************************** */
/* Types */
/* ***************************** */
/* ***************************** */
/* Varaibles */
/* ***************************** */
static int initialHeader = 1;
static unsigned int SubFrameLen = 0;
static unsigned int SubFramesPerPES = 0;
// reference: search for TypeLpcmDVDAudio in player/frame_parser/frame_parser_audio_lpcm.cpp
static const unsigned char clpcm_prv[14] = { 0xA0, //sub_stream_id
0, 0, //resvd and UPC_EAN_ISRC stuff, unused
0x0A, //private header length
0, 9, //first_access_unit_pointer
0x00, //emph,rsvd,stereo,downmix
0x0F, //quantisation word length 1,2
0x0F, //audio sampling freqency 1,2
0, //resvd, multi channel type
0, //bit shift on channel GR2, assignment
0x80, //dynamic range control
0, 0 //resvd for copyright management
};
static unsigned char lpcm_prv[14];
static unsigned char breakBuffer[8192];
static unsigned int breakBufferFillSize = 0;
/* ***************************** */
/* Prototypes */
/* ***************************** */
/* ***************************** */
/* MISC Functions */
/* ***************************** */
static int prepareClipPlay(int uNoOfChannels, int uSampleRate,
int uBitsPerSample, int bLittleEndian
__attribute__ ((unused)))
{
printf("rate: %d ch: %d bits: %d (%d bps)\n",
uSampleRate /*Format->dwSamplesPerSec */ ,
uNoOfChannels /*Format->wChannels */ ,
uBitsPerSample /*Format->wBitsPerSample */ ,
(uBitsPerSample /*Format->wBitsPerSample */ / 8)
);
SubFrameLen = 0;
SubFramesPerPES = 0;
breakBufferFillSize = 0;
memcpy(lpcm_prv, clpcm_prv, sizeof(lpcm_prv));
// figure out size of subframe and set up sample rate
switch (uSampleRate) {
case 48000:
SubFrameLen = 40;
break;
case 96000:
lpcm_prv[8] |= 0x10;
SubFrameLen = 80;
break;
case 192000:
lpcm_prv[8] |= 0x20;
SubFrameLen = 160;
break;
case 44100:
lpcm_prv[8] |= 0x80;
SubFrameLen = 40;
break;
case 88200:
lpcm_prv[8] |= 0x90;
SubFrameLen = 80;
break;
case 176400:
lpcm_prv[8] |= 0xA0;
SubFrameLen = 160;
break;
default:
break;
}
SubFrameLen *= uNoOfChannels;
SubFrameLen *= (uBitsPerSample / 8);
//rewrite PES size to have as many complete subframes per PES as we can
// FIXME: PES header size was hardcoded to 18 in previous code. Actual size returned by InsertPesHeader is 14.
SubFramesPerPES = ((2048 - 18) - sizeof(lpcm_prv)) / SubFrameLen;
SubFrameLen *= SubFramesPerPES;
//set number of channels
lpcm_prv[10] = uNoOfChannels - 1;
switch (uBitsPerSample) {
case 24:
lpcm_prv[7] |= 0x20;
case 16:
break;
default:
printf("inappropriate bits per sample (%d) - must be 16 or 24\n", uBitsPerSample);
return 1;
}
return 0;
}
static int reset()
{
initialHeader = 1;
return 0;
}
static int writeData(WriterAVCallData_t *call)
{
unsigned char PesHeader[PES_MAX_HEADER_SIZE];
pcm_printf(10, "\n");
if (!call) {
pcm_err("call data is NULL...\n");
return 0;
}
pcm_printf(10, "AudioPts %lld\n", call->Pts);
if (!call->data || (call->len <= 0)) {
pcm_err("parsing NULL Data. ignoring...\n");
return 0;
}
if (call->fd < 0) {
pcm_err("file pointer < 0. ignoring ...\n");
return 0;
}
if (initialHeader) {
initialHeader = 0;
prepareClipPlay(call->uNoOfChannels,
call->uSampleRate,
call->uBitsPerSample,
call->bLittleEndian);
}
unsigned char *buffer = call->data;
unsigned int size = call->len;
unsigned int n;
unsigned char *injectBuffer = (unsigned char *) malloc(SubFrameLen);
unsigned int pos;
for (pos = 0; pos < size;) {
//printf("PCM %s - Position=%d\n", __FUNCTION__, pos);
if ((size - pos) < SubFrameLen) {
breakBufferFillSize = size - pos;
memcpy(breakBuffer, &buffer[pos],
sizeof(unsigned char) * breakBufferFillSize);
//printf("PCM %s - Unplayed=%d\n", __FUNCTION__, breakBufferFillSize);
break;
}
//get first PES's worth
if (breakBufferFillSize > 0) {
memcpy(injectBuffer, breakBuffer, sizeof(unsigned char) * breakBufferFillSize);
memcpy(&injectBuffer[breakBufferFillSize], &buffer[pos], sizeof(unsigned char) * (SubFrameLen - breakBufferFillSize));
pos += (SubFrameLen - breakBufferFillSize);
breakBufferFillSize = 0;
} else {
memcpy(injectBuffer, &buffer[pos], sizeof(unsigned char) * SubFrameLen);
pos += SubFrameLen;
}
struct iovec iov[3];
iov[0].iov_base = PesHeader;
iov[1].iov_base = lpcm_prv;
iov[1].iov_len = sizeof(lpcm_prv);
iov[2].iov_base = injectBuffer;
iov[2].iov_len = SubFrameLen;
//write the PCM data
if (call->uBitsPerSample == 16) {
for (n = 0; n < SubFrameLen; n += 2) {
unsigned char tmp;
tmp = injectBuffer[n];
injectBuffer[n] = injectBuffer[n + 1];
injectBuffer[n + 1] = tmp;
}
} else {
// 0 1 2 3 4 5 6 7 8 9 10 11
// A1c A1b A1a-B1c B1b B1a-A2c A2b A2a-B2c B2b B2a
// to A1a A1b B1a B1b.A2a A2b B2a B2b-A1c B1c A2c B2c
for (n = 0; n < SubFrameLen; n += 12) {
unsigned char t, *p = &injectBuffer[n];
t = p[0];
p[0] = p[2];
p[2] = p[5];
p[5] = p[7];
p[7] = p[11];
p[11] = p[9];
p[9] = p[3];
p[3] = p[4];
p[4] = p[8];
p[8] = t;
}
}
//increment err... subframe count?
lpcm_prv[1] = ((lpcm_prv[1] + SubFramesPerPES) & 0x1F);
iov[0].iov_len = InsertPesHeader(PesHeader, iov[1].iov_len + iov[2].iov_len, PCM_PES_START_CODE, call->Pts, 0);
int len = writev(call->fd, iov, 3);
if (len < 0)
break;
}
free(injectBuffer);
return size;
}
SwrContext *swr = NULL;
AVFrame *decoded_frame = NULL;
int out_sample_rate = 44100;
int out_channels = 2;
uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
int restart_audio_resampling = 0;
static int resetIpcm()
{
if (swr)
swr_free(&swr);
if (decoded_frame)
av_frame_free(&decoded_frame);
return 0;
}
int64_t calcPts(AVFormatContext *, AVStream *, int64_t);
static int writeDataIpcm(WriterAVCallData_t *call)
{
AVCodecContext *c = call->stream->codec;
AVPacket *packet = call->packet;
uint8_t *packet_data = packet->data;
unsigned int packet_size = packet->size;
if (call->restart_audio_resampling)
call->restart_audio_resampling = 1;
if (restart_audio_resampling) {
restart_audio_resampling = 0;
if (swr) {
swr_free(&swr);
swr = NULL;
}
if (decoded_frame) {
av_frame_free(&decoded_frame);
decoded_frame = NULL;
}
call->context->output->Command(call->context, OUTPUT_CLEAR, NULL);
call->context->output->Command(call->context, OUTPUT_PLAY, NULL);
AVCodec *codec = avcodec_find_decoder(c->codec_id);
if (!codec || avcodec_open2(c, codec, NULL))
fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
}
while (packet_size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "out of memory\n");
exit(1);
}
} else
av_frame_unref(decoded_frame);
int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
if (len < 0) {
restart_audio_resampling = 1;
break;
}
packet_data += len;
packet_size -= len;
if (!got_frame)
continue;
int e;
if (!swr) {
int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
int *rate = rates;
int in_rate = c->sample_rate;
while (*rate && ((*rate / in_rate) * in_rate != *rate) && (in_rate / *rate) * *rate != in_rate)
rate++;
out_sample_rate = *rate ? *rate : 44100;
swr = swr_alloc();
out_channels = c->channels;
if (c->channel_layout == 0) {
// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
out_channel_layout = c->channel_layout;
// player2 won't play mono
if (out_channel_layout == AV_CH_LAYOUT_MONO) {
out_channel_layout = AV_CH_LAYOUT_STEREO;
out_channels = 2;
}
av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
av_opt_set_int(swr, "in_sample_fmt", c->sample_fmt, 0);
av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
e = swr_init(swr);
if (e < 0) {
fprintf(stderr,
"swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d\n",
-e, (int) c->channel_layout,
(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
swr_free(&swr);
swr = NULL;
}
}
uint8_t *output = NULL;
int in_samples = decoded_frame->nb_samples;
int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
if (e < 0) {
fprintf(stderr, "av_samples_alloc: %d\n", -e);
continue;
}
// FIXME. PTS calculation is probably broken.
int64_t pts;
int64_t next_in_pts = av_rescale(av_frame_get_best_effort_timestamp(decoded_frame),
call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate,
call->stream->time_base.den);
int64_t next_out_pts = av_rescale(swr_next_pts(swr, next_in_pts),
call->stream->time_base.den,
call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate);
*(call->context->currentAudioPtsP) = /* audioTrack->pts = */ pts = calcPts(call->avfc, call->stream, next_out_pts);
out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)
&decoded_frame->data[0], in_samples);
WriterAVCallData_t pcmOut;
pcmOut.fd = call->fd;
pcmOut.uSampleRate = out_sample_rate;
pcmOut.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
pcmOut.uBitsPerSample = 16;
pcmOut.bLittleEndian = 1;
pcmOut.data = output;
pcmOut.len = out_samples * sizeof(short) * out_channels;
pcmOut.Pts = pts; // FIXME videoTrack ? pts : 0;
pcmOut.stream = call->stream;
pcmOut.avfc = call->avfc;
pcmOut.packet = NULL;
writeData(&pcmOut);
av_freep(&output);
}
return packet->size;
}
/* ***************************** */
/* Writer Definition */
/* ***************************** */
static WriterCaps_t caps_pcm = {
"pcm",
eAudio,
"A_PCM",
AUDIO_ENCODING_LPCMA
};
struct Writer_s WriterAudioPCM = {
&reset,
&writeData,
NULL,
&caps_pcm
};
static WriterCaps_t caps_ipcm = {
"ipcm",
eAudio,
"A_IPCM",
AUDIO_ENCODING_LPCMA
};
struct Writer_s WriterAudioIPCM = {
&resetIpcm,
&writeDataIpcm,
NULL,
&caps_ipcm
};
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