コード例 #1
0
ファイル: hda.c プロジェクト: RafaelRMachado/Coreboot
static void codecs_init(void *base, u32 codec_mask)
{
	int i;
	for (i = 2; i >= 0; i--) {
		if (codec_mask & (1 << i))
			codec_init(base, i);
	}
}
コード例 #2
0
ファイル: rtp.c プロジェクト: CSRedRat/vlc
/* PT=32
 * MPV: MPEG Video (RFC2250, §3.5)
 */
static void *mpv_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, VIDEO_ES, VLC_CODEC_MPGV);
    fmt.b_packetized = false;
    return codec_init (demux, &fmt);
}
コード例 #3
0
ファイル: azalia.c プロジェクト: canistation/coreboot
static void codecs_init(struct device *dev, u8 *base, u32 codec_mask)
{
	int i;
	for (i = 2; i >= 0; i--) {
		if (codec_mask & (1 << i))
			codec_init(dev, base, i);
	}
}
コード例 #4
0
ファイル: a52.c プロジェクト: ifroz/rockbox
/* this is called for each file to process */
enum codec_status codec_run(void)
{
    size_t n;
    unsigned char *filebuf;
    int sample_loc;
    intptr_t param;

    if (codec_init())
        return CODEC_ERROR;

    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);
    
    /* Intialise the A52 decoder and check for success */
    state = a52_init(0);

    samplesdone = 0;

    /* The main decoding loop */
    if (ci->id3->offset) {
        if (ci->seek_buffer(ci->id3->offset)) {
            samplesdone = (ci->id3->offset / ci->id3->bytesperframe) *
                A52_SAMPLESPERFRAME;
            ci->set_elapsed(samplesdone/(ci->id3->frequency / 1000));
        }
    }
    else {
        ci->seek_buffer(ci->id3->first_frame_offset);
        ci->set_elapsed(0);
    }

    while (1) {
        enum codec_command_action action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        if (action == CODEC_ACTION_SEEK_TIME) {
            sample_loc = param/1000 * ci->id3->frequency;

            if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
                samplesdone = sample_loc;
                ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
            }
            ci->seek_complete();
        }

        filebuf = ci->request_buffer(&n, BUFFER_SIZE);

        if (n == 0) /* End of Stream */
            break;
  
        a52_decode_data(filebuf, filebuf + n);
        ci->advance_buffer(n);
    }

    return CODEC_OK;
}
コード例 #5
0
ファイル: rtp.c プロジェクト: shanewfx/vlc-arib
/* PT=14
 * MPA: MPEG Audio (RFC2250, §3.4)
 */
static void *mpa_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_MPGA);
    fmt.audio.i_channels = 2;
    fmt.b_packetized = false;
    return codec_init (demux, &fmt);
}
コード例 #6
0
ファイル: rtp.c プロジェクト: shanewfx/vlc-arib
/* PT=12
 * QCELP
 */
static void *qcelp_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_QCELP);
    fmt.audio.i_rate = 8000;
    fmt.audio.i_channels = 1;
    return codec_init (demux, &fmt);
}
コード例 #7
0
ファイル: rtp.c プロジェクト: shanewfx/vlc-arib
static void *l16m_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_S16B);
    fmt.audio.i_rate = 44100;
    fmt.audio.i_channels = 1;
    return codec_init (demux, &fmt);
}
コード例 #8
0
ファイル: rtp.c プロジェクト: Geal/vlc
/* PT=10,11
 * L16: 16-bits (network byte order) PCM
 */
static void *l16s_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_S16B);
    fmt.audio.i_rate = 44100;
    fmt.audio.i_original_channels =
    fmt.audio.i_physical_channels = AOUT_CHANS_STEREO;
    return codec_init (demux, &fmt);
}
コード例 #9
0
ファイル: rtp.c プロジェクト: Geal/vlc
/* PT=12
 * QCELP
 */
static void *qcelp_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_QCELP);
    fmt.audio.i_rate = 8000;
    fmt.audio.i_original_channels =
    fmt.audio.i_physical_channels = AOUT_CHAN_CENTER;
    return codec_init (demux, &fmt);
}
コード例 #10
0
ファイル: rtp.c プロジェクト: Geal/vlc
/* PT=14
 * MPA: MPEG Audio (RFC2250, §3.4)
 */
static void *mpa_init (demux_t *demux)
{
    es_format_t fmt;

    es_format_Init (&fmt, AUDIO_ES, VLC_CODEC_MPGA);
    fmt.audio.i_original_channels =
    fmt.audio.i_physical_channels = AOUT_CHANS_STEREO;
    fmt.b_packetized = false;
    return codec_init (demux, &fmt);
}
コード例 #11
0
ファイル: vorbis.c プロジェクト: 4nykey/rockbox
/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
    if (reason == CODEC_LOAD) {
        if (codec_init())
            return CODEC_ERROR;
        ci->configure(DSP_SET_SAMPLE_DEPTH, 24);
    }

    return CODEC_OK;
}
コード例 #12
0
ファイル: init.c プロジェクト: mauzybroadway/mycrosinth
void init_all(void)
{
	GPIO_init();
	DAC_DMA_init();
	P24_init();
	vdisplay_init();
	codec_init();
	codec_ctrl_init();
	TIM6_init();			/* Initialize timer */
	TIM3_init();			/* interrupts last... */
	TIM4_init();
}
コード例 #13
0
ファイル: drv_toshiba.c プロジェクト: d-torrance/dockapps
/*
 * Initialize the driver.
 */
static int
tosh_init( struct wm_drive *d )
{
	extern int	min_volume;

/* Sun use Toshiba drives as well */
#if defined(SYSV) && defined(CODEC)
        codec_init();
#endif
	min_volume = 0;
	return ( 0 );
}
コード例 #14
0
ファイル: snd_main.c プロジェクト: Wattos/Unvanquished
void S_Init( void )
{
	cvar_t *cv;
	qboolean  started = qfalse;

	Com_Printf("%s", _( "------ Initializing Sound -----\n" ));

	cv = Cvar_Get( "s_initsound", "1", 0 );
	if ( !cv->integer )
	{
		Com_Printf( "Sound Disabled\n" );
		return;
	}
	else
	{
		codec_init();

		s_volume = Cvar_Get( "s_volume", "0.8", CVAR_ARCHIVE );
		s_musicVolume = Cvar_Get( "s_musicvolume", "0.25", CVAR_ARCHIVE );
		s_separation = Cvar_Get( "s_separation", "0.5", CVAR_ARCHIVE );
		s_doppler = Cvar_Get( "s_doppler", "1", CVAR_ARCHIVE );
		s_khz = Cvar_Get( "s_khz", "22", CVAR_ARCHIVE );
		s_mixahead = Cvar_Get( "s_mixahead", "0.2", CVAR_ARCHIVE );

		s_mixPreStep = Cvar_Get( "s_mixPreStep", "0.05", CVAR_ARCHIVE );
		s_show = Cvar_Get( "s_show", "0", CVAR_CHEAT );
		s_testsound = Cvar_Get( "s_testsound", "0", CVAR_CHEAT );

		cv = Cvar_Get( "s_usemodule", "1", CVAR_ARCHIVE );
		if( cv->integer )
		{
			started = S_AL_Init( &si );
		}

		if( !started )
		{
			Com_Printf( "Using builtin sound backend\n" );
			S_Base_Init( &si );
		}
		else
		{
			Com_Printf( "Using OpenAL sound backend\n" );
		}

		Cmd_AddCommand( "play", S_Play_f );
		Cmd_AddCommand( "music", S_Music_f );
		Cmd_AddCommand( "s_list", S_SoundList_f );
		Cmd_AddCommand( "s_info", S_SoundInfo_f );
		Cmd_AddCommand( "s_stop", S_StopAllSounds );
	}
	Com_Printf( "------------------------------------\n" );
}
コード例 #15
0
ファイル: aml.c プロジェクト: AreaScout/moonlight-embedded
int aml_setup(int videoFormat, int width, int height, int redrawRate, void* context, int drFlags) {
  codecParam.stream_type = STREAM_TYPE_ES_VIDEO;
  codecParam.has_video = 1;
  codecParam.noblock = 0;
  codecParam.am_sysinfo.param = 0;

  switch (videoFormat) {
    case VIDEO_FORMAT_H264:
      if (width > 1920 || height > 1080) {
        codecParam.video_type = VFORMAT_H264_4K2K;
        codecParam.am_sysinfo.format = VIDEO_DEC_FORMAT_H264_4K2K;
      } else {
        codecParam.video_type = VFORMAT_H264;
        codecParam.am_sysinfo.format = VIDEO_DEC_FORMAT_H264;

        // Workaround for decoding special case of C1, 1080p, H264
        int major, minor;
        struct utsname name;
        uname(&name);
        int ret = sscanf(name.release, "%d.%d", &major, &minor);
        if (!(major > 3 || (major == 3 && minor >= 14)) && width == 1920 && height == 1080)
            codecParam.am_sysinfo.param = (void*) UCODE_IP_ONLY_PARAM;
      }
      break;
    case VIDEO_FORMAT_H265:
      codecParam.video_type = VFORMAT_HEVC;
      codecParam.am_sysinfo.format = VIDEO_DEC_FORMAT_HEVC;
      break;
    default:
      printf("Video format not supported\n");
      return -1;
  }

  codecParam.am_sysinfo.width = width;
  codecParam.am_sysinfo.height = height;
  codecParam.am_sysinfo.rate = 96000 / redrawRate;
  codecParam.am_sysinfo.param = (void*) ((size_t) codecParam.am_sysinfo.param | SYNC_OUTSIDE);

  int ret;
  if ((ret = codec_init(&codecParam)) != 0) {
    fprintf(stderr, "codec_init error: %x\n", ret);
    return -2;
  }

  if ((ret = codec_set_freerun_mode(&codecParam, 1)) != 0) {
    fprintf(stderr, "Can't set Freerun mode: %x\n", ret);
    return -2;
  }

  return 0;
}
コード例 #16
0
ファイル: azalia.c プロジェクト: 0ida/coreboot
static void codecs_init(struct device *dev, u32 base, u32 codec_mask)
{
	int i;
	for (i = 3; i >= 0; i--) {
		if (codec_mask & (1 << i))
			codec_init(dev, base, i);
	}

	for (i = 0; i < pc_beep_verbs_size; i++) {
		if (wait_for_ready(base) == -1)
			return;

		write32(base + 0x60, pc_beep_verbs[i]);

		if (wait_for_valid(base) == -1)
			return;
	}
}
コード例 #17
0
ファイル: gstamlvdec.c プロジェクト: vitmod/buildroot-aml
static gboolean
gst_set_vstream_info(GstAmlVdec *amlvdec, GstCaps * caps)
{
	GstStructure *structure;
	const char *name;
	gint32 ret = CODEC_ERROR_NONE;
	AmlStreamInfo *videoinfo = NULL;
	int coordinate[4] = {0, 0, 0, 0};
	structure = gst_caps_get_structure(caps, 0);
	name = gst_structure_get_name(structure);

	videoinfo = amlVstreamInfoInterface(name);
	if (NULL == name) {
		return FALSE;
	}
	amlvdec->info = videoinfo;
	videoinfo->init(videoinfo, amlvdec->pcodec, structure);
	if (amlvdec->pcodec && amlvdec->pcodec->stream_type == STREAM_TYPE_ES_VIDEO) {
		if (!amlvdec->codec_init_ok) {
			ret = codec_init(amlvdec->pcodec);
			if (ret != CODEC_ERROR_NONE) {
				GST_ERROR("codec init failed, ret=-0x%x", -ret);
				return FALSE;
			}
			set_fb0_blank(1);
			set_fb1_blank(1);
			set_tsync_enable(1);
			set_display_axis(coordinate);
			//set_video_axis(coordinate);
			amlvdec->codec_init_ok = 1;
			if (amlvdec->trickRate > 0) {
				if (amlvdec->pcodec && amlvdec->pcodec->cntl_handle) {
//T					codec_set_video_playrate(amlvdec->pcodec, (int) (amlvdec->trickRate * (1 << 16)));
				}
			}
			GST_DEBUG_OBJECT(amlvdec, "video codec_init ok");
		}

	}
	return TRUE;
}
コード例 #18
0
ファイル: TsPlayer.cpp プロジェクト: cg8530/WorkSplace
bool CTsPlayer::StartPlay()
{
	int ret;
	memset(pcodec,0,sizeof(*pcodec));
	pcodec->stream_type=STREAM_TYPE_TS;
	pcodec->video_type = vPara.vFmt;
	pcodec->has_video=1;
	pcodec->audio_type= aPara.aFmt;
	pcodec->has_audio=1;
	pcodec->video_pid=(int)vPara.pid;
	pcodec->audio_pid=(int)aPara.pid;
	//pcodec->audio_channels = 1;
	//pcodec->audio_samplerate = 48000;
	// 解决AC3无声的问题
    if(IS_AUIDO_NEED_EXT_INFO(pcodec->audio_type)){
        pcodec->audio_info.valid = 1;
		LOGI("set audio_info.valid to 1");
    }
    if (pcodec->video_type == VFORMAT_H264) {
         pcodec->am_sysinfo.format = VIDEO_DEC_FORMAT_H264;
         pcodec->am_sysinfo.param = (void *)(0);
    }
    else if(pcodec->video_type ==VFORMAT_MPEG12){
        pcodec->am_sysinfo.param = (void *)(0);
    }
	printf("set %d,%d,%d,%d\n",vPara.vFmt,aPara.aFmt,vPara.pid,aPara.pid);
	pcodec->noblock = 0;
	/*other setting*/
	ret=codec_init(pcodec);

 #if DUMP_TSPACKET1	
	 m_fStream = fopen("/data/data/novel.supertv.dvb/new_1.ts","wb");
 #endif 
	
	return !ret;
}
コード例 #19
0
ファイル: xiph.c プロジェクト: 12307/VLC-for-VS2010
void xiph_decode (demux_t *demux, void *data, block_t *block)
{
    rtp_xiph_t *self = (rtp_xiph_t *)data;			// sunqueen modify

    if (!data || block->i_buffer < 4)
        goto drop;

    /* 32-bits RTP header (§2.2) */
    uint32_t ident = GetDWBE (block->p_buffer);
    block->i_buffer -= 4;
    block->p_buffer += 4;

    unsigned fragtype = (ident >> 6) & 3;
    unsigned datatype = (ident >> 4) & 3;
    unsigned pkts = (ident) & 15;
    ident >>= 8;

    /* RTP defragmentation */
    if (self->block && (block->i_flags & BLOCK_FLAG_DISCONTINUITY))
    {   /* Screwed! discontinuity within a fragmented packet */
        msg_Warn (demux, self->vorbis ?
                  "discontinuity in fragmented Vorbis packet" :
                  "discontinuity in fragmented Theora packet");
        block_Release (self->block);
        self->block = NULL;
    }

    if (fragtype <= 1)
    {
        if (self->block) /* Invalid first fragment */
        {
            block_Release (self->block);
            self->block = NULL;
        }
    }
    else
    {
        if (!self->block)
            goto drop; /* Invalid non-first fragment */
    }

    if (fragtype > 0)
    {   /* Fragment */
        if (pkts > 0 || block->i_buffer < 2)
            goto drop;

        size_t fraglen = GetWBE (block->p_buffer);
        if (block->i_buffer < (fraglen + 2))
            goto drop; /* Invalid payload length */
        block->i_buffer = fraglen;
        if (fragtype == 1)/* Keep first fragment */
        {
            block->i_buffer += 2;
            self->block = block;
        }
        else
        {   /* Append non-first fragment */
            size_t len = self->block->i_buffer;
            self->block = block_Realloc (self->block, 0, len + fraglen);
            if (!self->block)
            {
                block_Release (block);
                return;
            }
            memcpy (self->block->p_buffer + len, block->p_buffer + 2,
                    fraglen);
            block_Release (block);
        }
        if (fragtype < 3)
            return; /* Non-last fragment */

        /* Last fragment reached, process it */
        block = self->block;
        self->block = NULL;
        SetWBE (block->p_buffer, block->i_buffer - 2);
        pkts = 1;
    }

    /* RTP payload packets processing */
    while (pkts > 0)
    {
        if (block->i_buffer < 2)
            goto drop;

        size_t len = GetWBE (block->p_buffer);
        block->i_buffer -= 2;
        block->p_buffer += 2;
        if (block->i_buffer < len)
            goto drop;

        switch (datatype)
        {
            case 0: /* Raw payload */
            {
                if (self->ident != ident)
                {
                    msg_Warn (demux, self->vorbis ?
                        "ignoring raw Vorbis payload without configuration" :
                        "ignoring raw Theora payload without configuration");
                    break;
                }
                block_t *raw = block_Alloc (len);
                memcpy (raw->p_buffer, block->p_buffer, len);
                raw->i_pts = block->i_pts; /* FIXME: what about pkts > 1 */
                codec_decode (demux, self->id, raw);
                break;
            }

            case 1: /* Packed configuration frame (§3.1.1) */
            {
                if (self->ident == ident)
                    break; /* Ignore config retransmission */

                void *extv;
                ssize_t extc = xiph_header (&extv, block->p_buffer, len);
                if (extc < 0)
                    break;

                es_format_t fmt;
                es_format_Init (&fmt, self->vorbis ? AUDIO_ES : VIDEO_ES,
                                self->vorbis ? VLC_CODEC_VORBIS
                                             : VLC_CODEC_THEORA);
                fmt.p_extra = extv;
                fmt.i_extra = extc;
                codec_destroy (demux, self->id);
                msg_Dbg (demux, self->vorbis ?
                         "Vorbis packed configuration received (%06"PRIx32")" :
                         "Theora packed configuration received (%06"PRIx32")",
                         ident);
                self->ident = ident;
                self->id = (es_out_id_t *)codec_init (demux, &fmt);			// sunqueen modify
                break;
            }
        }

        block->i_buffer -= len;
        block->p_buffer += len;
        pkts--;
    }

drop:
    block_Release (block);
}
コード例 #20
0
ファイル: a52.c プロジェクト: claymodel/rockbox
/* this is the codec entry point */
enum codec_status codec_main(void)
{
    size_t n;
    unsigned char *filebuf;
    int sample_loc;
    int retval;

    /* Generic codec initialisation */
    ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);

next_track:
    retval = CODEC_OK;

    if (codec_init()) {
        retval = CODEC_ERROR;
        goto exit;
    }

    if (codec_wait_taginfo() != 0)
        goto request_next_track;

    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);
    
    /* Intialise the A52 decoder and check for success */
    state = a52_init(0);

    /* The main decoding loop */
    if (ci->id3->offset) {
        if (ci->seek_buffer(ci->id3->offset)) {
            samplesdone = (ci->id3->offset / ci->id3->bytesperframe) *
                A52_SAMPLESPERFRAME;
            ci->set_elapsed(samplesdone/(ci->id3->frequency / 1000));
        }
    }
    else {
        samplesdone = 0;
    }

    while (1) {
        if (ci->stop_codec || ci->new_track)
            break;

        if (ci->seek_time) {
            sample_loc = (ci->seek_time - 1)/1000 * ci->id3->frequency;

            if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
                samplesdone = sample_loc;
                ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
            }
            ci->seek_complete();
        }

        filebuf = ci->request_buffer(&n, BUFFER_SIZE);

        if (n == 0) /* End of Stream */
            break;
  
        a52_decode_data(filebuf, filebuf + n);
        ci->advance_buffer(n);
    }

request_next_track:
    if (ci->request_next_track())
        goto next_track;

exit:
    a52_free(state);
    return retval;
}
コード例 #21
0
ファイル: PreenFM_init.c プロジェクト: norbim1/preenFM2
void CS43L22_Config() {
	  // configure STM32F4Discovery I2S pins

	RCC_AHB1PeriphClockCmd(RCC_AHB1Periph_GPIOA | RCC_AHB1Periph_GPIOB | RCC_AHB1Periph_GPIOC | RCC_AHB1Periph_GPIOD, ENABLE);

	GPIO_InitTypeDef GPIO_InitStructure;
	  GPIO_StructInit(&GPIO_InitStructure);
	  GPIO_InitStructure.GPIO_Speed = GPIO_Speed_50MHz;
	  GPIO_InitStructure.GPIO_Mode = GPIO_Mode_AF;
	  GPIO_InitStructure.GPIO_OType = GPIO_OType_PP;

	  GPIO_InitStructure.GPIO_Pin = STM32F4D_I2S_WS_PIN;
	  GPIO_Init(STM32F4D_I2S_WS_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(STM32F4D_I2S_WS_PORT, STM32F4D_I2S_WS_PINSRC, GPIO_AF_SPI3);

	  GPIO_InitStructure.GPIO_Pin = STM32F4D_I2S_CK_PIN;
	  GPIO_Init(STM32F4D_I2S_CK_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(STM32F4D_I2S_CK_PORT, STM32F4D_I2S_CK_PINSRC, GPIO_AF_SPI3);

	  GPIO_InitStructure.GPIO_Pin = STM32F4D_I2S_SD_PIN;
	  GPIO_Init(STM32F4D_I2S_SD_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(STM32F4D_I2S_SD_PORT, STM32F4D_I2S_SD_PINSRC, GPIO_AF_SPI3);

	  GPIO_InitStructure.GPIO_Pin = STM32F4D_I2S_MCLK_PIN;
	  GPIO_Init(STM32F4D_I2S_MCLK_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(STM32F4D_I2S_MCLK_PORT, STM32F4D_I2S_MCLK_PINSRC, GPIO_AF_SPI3);

	  // configure I2C pins to access the CS43L22 configuration registers

	  GPIO_InitStructure.GPIO_Mode = GPIO_Mode_AF;
	  GPIO_InitStructure.GPIO_OType = GPIO_OType_OD;
	  GPIO_InitStructure.GPIO_PuPd = GPIO_PuPd_NOPULL;
	  GPIO_InitStructure.GPIO_Speed = GPIO_Speed_50MHz;

	  GPIO_InitStructure.GPIO_Pin = CODEC_I2C_SCL_PIN;
	  GPIO_Init(CODEC_I2C_SCL_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(CODEC_I2C_SCL_PORT, GPIO_PinSource6, GPIO_AF_I2C1);

	  GPIO_InitStructure.GPIO_Pin = CODEC_I2C_SDA_PIN;
	  GPIO_Init(CODEC_I2C_SDA_PORT, &GPIO_InitStructure);
	  GPIO_PinAFConfig(CODEC_I2C_SDA_PORT, GPIO_PinSource9, GPIO_AF_I2C1);

	  // CS43L22 reset pin
	  GPIO_InitStructure.GPIO_Mode = GPIO_Mode_OUT;
	  GPIO_InitStructure.GPIO_OType = GPIO_OType_PP;
	  GPIO_InitStructure.GPIO_Pin = CODEC_RESET_PIN;
	  GPIO_Init(CODEC_RESET_PORT, &GPIO_InitStructure);

	  GPIO_ResetBits(CODEC_RESET_PORT, CODEC_RESET_PIN); // activate reset

	  // I2S initialisation
	  RCC_APB1PeriphClockCmd(RCC_APB1Periph_SPI3, ENABLE);
	  RCC_PLLI2SCmd(ENABLE); // new for STM32F4: enable I2S PLL
	  SPI_I2S_DeInit(SPI3);
	  I2S_InitTypeDef I2S_InitStructure;
	  I2S_StructInit(&I2S_InitStructure);
	  I2S_InitStructure.I2S_Standard = STM32F4D_I2S_STANDARD;
	  I2S_InitStructure.I2S_DataFormat = STM32F4D_I2S_DATA_FORMAT;
	  I2S_InitStructure.I2S_MCLKOutput = STM32F4D_I2S_MCLK_ENABLE ? I2S_MCLKOutput_Enable : I2S_MCLKOutput_Disable;
	  I2S_InitStructure.I2S_AudioFreq  = (u16)(STM32F4D_I2S_AUDIO_FREQ);
	  I2S_InitStructure.I2S_CPOL = I2S_CPOL_Low; // configuration required as well?
	  I2S_InitStructure.I2S_Mode = I2S_Mode_MasterTx;
	  I2S_Init(SPI3, &I2S_InitStructure);
	  I2S_Cmd(SPI3, ENABLE);

	  // DMA Configuration for SPI Tx Event
	  RCC_AHB1PeriphClockCmd(RCC_AHB1Periph_DMA1, ENABLE);

	  DMA_InitTypeDef DMA_InitStructure;
	  DMA_StructInit(&DMA_InitStructure);

	  DMA_Cmd(DMA1_Stream5, DISABLE);
	  DMA_ClearFlag(DMA1_Stream5, DMA_FLAG_TCIF5 | DMA_FLAG_TEIF5 | DMA_FLAG_HTIF5 | DMA_FLAG_FEIF5);
	  DMA_InitStructure.DMA_Channel = DMA_Channel_0;
	  DMA_InitStructure.DMA_PeripheralBaseAddr = (u32)&SPI3->DR;
	  //  DMA_InitStructure.DMA_MemoryBaseAddr = ...; // configured in CS43L22_Start
	  DMA_InitStructure.DMA_DIR = DMA_DIR_MemoryToPeripheral;
	  //  DMA_InitStructure.DMA_BufferSize = ...; // configured in CS43L22_Start
	  DMA_InitStructure.DMA_PeripheralInc = DMA_PeripheralInc_Disable;
	  DMA_InitStructure.DMA_MemoryInc = DMA_MemoryInc_Enable;
	  DMA_InitStructure.DMA_PeripheralDataSize = DMA_PeripheralDataSize_HalfWord;
	  DMA_InitStructure.DMA_MemoryDataSize = DMA_MemoryDataSize_HalfWord;
	  DMA_InitStructure.DMA_Mode = DMA_Mode_Circular;
	  DMA_InitStructure.DMA_Priority = DMA_Priority_Medium;
	  DMA_InitStructure.DMA_FIFOMode = DMA_FIFOMode_Disable;
	  DMA_InitStructure.DMA_MemoryBurst = DMA_MemoryBurst_Single;
	  DMA_InitStructure.DMA_PeripheralBurst = DMA_PeripheralBurst_Single;

	  DMA_Init(DMA1_Stream5, &DMA_InitStructure);
	  // DMA_Cmd(DMA1_Stream5, ENABLE); // done on CS43L22_Start

	  DMA_ITConfig(DMA1_Stream5, DMA_IT_TE | DMA_IT_FE, DISABLE);
	  // trigger interrupt when transfer half complete/complete
	  DMA_ITConfig(DMA1_Stream5, DMA_IT_HT | DMA_IT_TC, ENABLE);

	  // enable SPI interrupts to DMA
	  SPI_I2S_DMACmd(SPI3, SPI_I2S_DMAReq_Tx, ENABLE);

	  // Configure and enable DMA interrupt
	  /* Configure the DMA IRQ handler priority */
	  NVIC_SetPriority(DMA1_Stream5_IRQn, 0x0);
	  NVIC_EnableIRQ(DMA1_Stream5_IRQn);

	  // configure I2C
	  RCC_APB1PeriphClockCmd(RCC_APB1Periph_I2C1, ENABLE);
	  I2C_DeInit(CODEC_I2C);
	  I2C_InitTypeDef I2C_InitStructure;
	  I2C_StructInit(&I2C_InitStructure);
	  I2C_InitStructure.I2C_ClockSpeed = 100000;
	  I2C_InitStructure.I2C_Mode = I2C_Mode_I2C;
	  I2C_InitStructure.I2C_OwnAddress1 = CORE_I2C_ADDRESS;
	  I2C_InitStructure.I2C_Ack = I2C_Ack_Enable;
	  I2C_InitStructure.I2C_AcknowledgedAddress = I2C_AcknowledgedAddress_7bit;
	  I2C_InitStructure.I2C_DutyCycle = I2C_DutyCycle_2;

	  I2C_Cmd(CODEC_I2C, ENABLE);
	  I2C_Init(CODEC_I2C, &I2C_InitStructure);

	  codec_init();
}
コード例 #22
0
ファイル: atrac3_oma.c プロジェクト: Rockbox/rockbox
/* this is called for each file to process */
enum codec_status codec_run(void)
{
    static size_t buff_size;
    int datasize, res, frame_counter, total_frames, seek_frame_offset;
    uint8_t *bit_buffer;
    int elapsed = 0;
    size_t resume_offset;
    intptr_t param;
    long action;

    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }

    action = CODEC_ACTION_NULL;
    param = ci->id3->elapsed;
    resume_offset = ci->id3->offset;

    codec_set_replaygain(ci->id3);
    ci->memset(&q,0,sizeof(ATRAC3Context));
 
    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
    ci->configure(DSP_SET_SAMPLE_DEPTH, 17); /* Remark: atrac3 uses s15.0 by default, s15.2 was hacked. */
    ci->configure(DSP_SET_STEREO_MODE, ci->id3->channels == 1 ?
        STEREO_MONO : STEREO_NONINTERLEAVED);

    ci->seek_buffer(0);

    res = atrac3_decode_init(&q, ci->id3);
    if(res < 0) {
        DEBUGF("failed to initialize OMA atrac decoder\n");
        return CODEC_ERROR;
    }

    total_frames = (ci->id3->filesize - ci->id3->first_frame_offset) / FRAMESIZE;
    frame_counter = 0;
    
    /* check for a mid-track resume and force a seek time accordingly */
    if (resume_offset) {
        resume_offset -= MIN(resume_offset, ci->id3->first_frame_offset);
        /* calculate resume_offset in frames */
        param = (resume_offset/FRAMESIZE) * ((FRAMESIZE * 8)/BITRATE);
    }

    if ((unsigned long)param) {
        action = CODEC_ACTION_SEEK_TIME;
    }
    else {
        ci->set_elapsed(0);
        ci->seek_buffer(ci->id3->first_frame_offset);
    }

    /* The main decoder loop */  
    while(frame_counter < total_frames)
    {
        if (action == CODEC_ACTION_NULL)
            action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        if (action == CODEC_ACTION_SEEK_TIME) {
            /* Do not allow seeking beyond the file's length */
            if ((unsigned long) param > ci->id3->length) {
                ci->set_elapsed(ci->id3->length);
                ci->seek_complete();
                break;
            }       

            /* Seek to the start of the track */
            if (param == 0) {
                elapsed = 0;
                ci->set_elapsed(0);
                ci->seek_buffer(ci->id3->first_frame_offset);
                ci->seek_complete();
                action = CODEC_ACTION_NULL;
                continue;           
            }                                                                

            seek_frame_offset = (param * BITRATE) / (8 * FRAMESIZE);
            frame_counter = seek_frame_offset;
            ci->seek_buffer(ci->id3->first_frame_offset + seek_frame_offset* FRAMESIZE);
            elapsed = param;
            ci->set_elapsed(elapsed);
            ci->seek_complete(); 
        }

        action = CODEC_ACTION_NULL;

        bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, FRAMESIZE);

        res = atrac3_decode_frame(FRAMESIZE, &q, &datasize, bit_buffer, FRAMESIZE);

        if(res != (int)FRAMESIZE) {
            DEBUGF("codec error\n");
            return CODEC_ERROR;
        }

        if(datasize)
            ci->pcmbuf_insert(q.outSamples, q.outSamples + 1024,
                              q.samples_per_frame / ci->id3->channels);

        elapsed += (FRAMESIZE * 8) / BITRATE;
        ci->set_elapsed(elapsed);

        ci->advance_buffer(FRAMESIZE);
        frame_counter++;
    }

    return CODEC_OK;    
}
コード例 #23
0
/* this is the codec entry point */
enum codec_status codec_main(void)
{
    uint32_t elapsedtime;
    int retval;
    asf_waveformatex_t wfx;     /* Holds the stream properties */
    size_t resume_offset;
    int res;                    /* Return values from asf_read_packet() and decode_packet() */
    uint8_t* audiobuf;          /* Pointer to the payload of one wma pro packet */
    int audiobufsize;           /* Payload size */
    int packetlength = 0;       /* Logical packet size (minus the header size) */          
    int outlen = 0;             /* Number of bytes written to the output buffer */
    int pktcnt = 0;             /* Count of the packets played */

    /* Generic codec initialisation */
    ci->configure(DSP_SET_SAMPLE_DEPTH, 31);
    

next_track:
    retval = CODEC_OK;

    /* Wait for the metadata to be read */
    if (codec_wait_taginfo() != 0)
        goto done;

    /* Remember the resume position */
    resume_offset = ci->id3->offset;
restart_track:
    retval = CODEC_OK;

    if (codec_init()) {
        LOGF("(WMA Voice) Error: Error initialising codec\n");
        retval = CODEC_ERROR;
        goto done;
    }

    /* Copy the format metadata we've stored in the id3 TOC field.  This
       saves us from parsing it again here. */
    memcpy(&wfx, ci->id3->toc, sizeof(wfx));
    memset(&avctx, 0, sizeof(AVCodecContext));
    memset(&avpkt, 0, sizeof(AVPacket));
    
    ci->configure(DSP_SWITCH_FREQUENCY, wfx.rate);
    ci->configure(DSP_SET_STEREO_MODE, wfx.channels == 1 ?
                  STEREO_MONO : STEREO_INTERLEAVED);
    codec_set_replaygain(ci->id3);
    
    /* Initialise the AVCodecContext */
    init_codec_ctx(&avctx, &wfx);

    if (wmavoice_decode_init(&avctx) < 0) {
        LOGF("(WMA Voice) Error: Unsupported or corrupt file\n");
        retval = CODEC_ERROR;
        goto done;
    }

    /* Now advance the file position to the first frame */
    ci->seek_buffer(ci->id3->first_frame_offset);
    
    elapsedtime = 0;
    resume_offset = 0;
    
    /* The main decoding loop */

    while (pktcnt < wfx.numpackets)
    {
        ci->yield();
        if (ci->stop_codec || ci->new_track) {
            goto done;
        }
        
        /* Deal with any pending seek requests */
        if (ci->seek_time){

            if (ci->seek_time == 1) {
                ci->seek_complete();
                goto restart_track; /* Pretend you never saw this... */
            }

            elapsedtime = asf_seek(ci->seek_time, &wfx);
            if (elapsedtime < 1){
                ci->seek_complete();
                goto next_track;
            }

            ci->set_elapsed(elapsedtime);
            ci->seek_complete();
        }
        
new_packet:
        res = asf_read_packet(&audiobuf, &audiobufsize, &packetlength, &wfx);

        if (res < 0) {
            LOGF("(WMA Voice) read_packet error %d\n",res);
            goto done;
        } else {
            avpkt.data = audiobuf;
            avpkt.size = audiobufsize;
            pktcnt++;
            
            while(avpkt.size > 0)
            {
                /* wmavoice_decode_packet checks for the output buffer size to 
                   avoid overflows */
                outlen = BUFSIZE*sizeof(int32_t);

                res = wmavoice_decode_packet(&avctx, decoded, &outlen, &avpkt);
                if(res < 0) {
                    LOGF("(WMA Voice) Error: decode_packet returned %d", res);
                    if(res == ERROR_WMAPRO_IN_WMAVOICE){
                    /* Just skip this packet */
                        ci->advance_buffer(packetlength);
                        goto new_packet;    
                    }
                    else
                        goto done;
                }
                avpkt.data += res;
                avpkt.size -= res;
                if(outlen) {
                    ci->yield ();
                    outlen /= sizeof(int32_t);
                    ci->pcmbuf_insert(decoded, NULL, outlen);
                    elapsedtime += outlen*10/(wfx.rate/100);
                    ci->set_elapsed(elapsedtime);
                    ci->yield ();
                }
            }

        }

        /* Advance to the next logical packet */
        ci->advance_buffer(packetlength);
    }

done:
    if (ci->request_next_track())
        goto next_track;
    
    return retval;
}
コード例 #24
0
int 
main(int argc, char *argv[])
{
        const char *codec_name, *repair_name;
        codec_id_t cid;
        repair_id_t rid;
        struct s_sndfile *sf_in = NULL, *sf_out = NULL;
        sndfile_fmt_t     sff;
        double drop = 0.0;
        int ac, did_query = FALSE;
        int csra  = TRUE; /* codec specific repair allowed */
        long seed = 100;

        codec_init();

        ac = 1;
        while(ac < argc && argv[ac][0] == '-') {
                if (strlen(argv[ac]) > 2) {
                        /* should be -codecs or -repairs */
                        switch(argv[ac][1]) {
                        case 'c':
                                list_codecs();
                                break;
                        case 'r':
                                list_repairs(); 
                                break;
                        default:
                                usage();
                        }
                        did_query = TRUE;
                } else {
                        if (argc - ac < 1) {
                                usage();
                        } 
                        switch(argv[ac][1]) {
                        case 's':
                                seed = atoi(argv[++ac]);
                                break;
                        case 'd':
                                drop = strtod(argv[++ac], NULL);
                                break;
                        case 'c':
                                cid  = codec_get_by_name(argv[++ac]);
                                codec_name = argv[ac];
                                break;
                        case 'n':
                                csra = FALSE;
                                break;
                        case 'r':
                                resolve_repair(argv[++ac], &rid, &repair_name);
                                break;
                        case 'u':
                                units_per_packet = atoi(argv[++ac]);
                                break;
                        default:
                                usage();
                        }
                }
                ac++;
        }
        
        if (did_query == TRUE) {
                /* Not sensible to be running query and executing test */
                exit(-1);
        }

        if (argc - ac != 2) {
                usage();
        }


        if (snd_read_open(&sf_in, argv[ac], NULL) == FALSE) {
                fprintf(stderr, "Could not open %s\n", argv[ac]);
                exit(-1);
        }
        ac++;

        if (snd_get_format(sf_in, &sff) == FALSE) {
                fprintf(stderr, "Failed to get format of %s\n", argv[ac]);
                exit(-1);
        }

        if (snd_write_open(&sf_out, argv[ac], "au", &sff) == FALSE) {
                fprintf(stderr, "Could not open %s\n", argv[ac]);
                exit(-1);
        }

        if (file_and_codec_compatible(&sff, cid) == FALSE) {
                fprintf(stderr, "Codec and file type are not compatible\n");
                exit(-1);
        }

        printf("# Parameters
#\tseed: %ld
#\tdrop: %.2f
#\tcodec:  %s
#\tunits per packet: %d
#\trepair: %s
#\tcodec specific repair (when available): %d
#\tsource file: %s
#\tdestination file %s\n",
seed, drop, codec_name, units_per_packet, repair_name, csra, argv[argc - 2], argv[argc - 1]);

	repair_set_codec_specific_allowed(csra);

	init_drop(seed, drop);
        test_repair(sf_out, cid, rid, sf_in);

        /* snd_read_close(&sf_in) not needed because files gets closed
         * at eof automatically. 
         */
        snd_write_close(&sf_out);

        codec_exit();
        xmemdmp();

        return 0;
}
コード例 #25
0
/* this is called for each file to process */
enum codec_status codec_run(void)
{
    WavpackContext *wpc;
    char error [80];
    /* rockbox: comment 'set but unused' variables
    int bps;
    */
    int nchans, sr_100;
    intptr_t param;

    if (codec_init())
        return CODEC_ERROR;

    ci->seek_buffer (ci->id3->offset);

    /* Create a decoder instance */
    wpc = WavpackOpenFileInput (read_callback, error);

    if (!wpc)
        return CODEC_ERROR;

    ci->configure(DSP_SWITCH_FREQUENCY, WavpackGetSampleRate (wpc));
    codec_set_replaygain(ci->id3);
    /* bps = WavpackGetBytesPerSample (wpc); */
    nchans = WavpackGetReducedChannels (wpc);
    ci->configure(DSP_SET_STEREO_MODE, nchans == 2 ? STEREO_INTERLEAVED : STEREO_MONO);
    sr_100 = ci->id3->frequency / 100;

    ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);

    /* The main decoder loop */

    while (1) {
        int32_t nsamples;
        enum codec_command_action action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        if (action == CODEC_ACTION_SEEK_TIME) {
            int curpos_ms = WavpackGetSampleIndex (wpc) / sr_100 * 10;
            int n, d, skip;

            if (param > curpos_ms) {
                n = param - curpos_ms;
                d = ci->id3->length - curpos_ms;
                skip = (int)((int64_t)(ci->filesize - ci->curpos) * n / d);
                ci->seek_buffer (ci->curpos + skip);
            }
            else if (curpos_ms != 0) {
                n = curpos_ms - param;
                d = curpos_ms;
                skip = (int)((int64_t) ci->curpos * n / d);
                ci->seek_buffer (ci->curpos - skip);
            }

            wpc = WavpackOpenFileInput (read_callback, error);
            if (!wpc)
            {
                ci->seek_complete();
                break;
            }

            ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
            ci->seek_complete();
        }

        nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / nchans);  

        if (!nsamples)
            break;

        ci->pcmbuf_insert (temp_buffer, NULL, nsamples);
        ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
    }

    return CODEC_OK;
}
コード例 #26
0
ファイル: ape.c プロジェクト: claymodel/rockbox
/* this is the codec entry point */
enum codec_status codec_main(void)
{
    struct ape_ctx_t ape_ctx;
    uint32_t samplesdone;
    uint32_t elapsedtime;
    size_t bytesleft;
    int retval;

    uint32_t currentframe;
    uint32_t newfilepos;
    uint32_t samplestoskip;
    int nblocks;
    int bytesconsumed;
    unsigned char* inbuffer;
    uint32_t blockstodecode;
    int res;
    int firstbyte;
    size_t resume_offset;

    /* Generic codec initialisation */
    ci->configure(DSP_SET_SAMPLE_DEPTH, APE_OUTPUT_DEPTH-1);

next_track:
    retval = CODEC_OK;

    if (codec_init()) {
        LOGF("APE: Error initialising codec\n");
        retval = CODEC_ERROR;
        goto exit;
    }

    if (codec_wait_taginfo() != 0)
        goto done;

    /* Remember the resume position - when the codec is opened, the
       playback engine will reset it. */
    resume_offset = ci->id3->offset;

    inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);

    /* Read the file headers to populate the ape_ctx struct */
    if (ape_parseheaderbuf(inbuffer,&ape_ctx) < 0) {
        LOGF("APE: Error reading header\n");
        retval = CODEC_ERROR;
        goto exit;
    }

    /* Initialise the seektable for this file */
    ape_ctx.seektable = seektablebuf;
    ape_ctx.numseekpoints = MIN(MAX_SEEKPOINTS,ape_ctx.numseekpoints);

    ci->advance_buffer(ape_ctx.seektablefilepos);

    /* The seektable may be bigger than the guard buffer (32KB), so we
       do a read() */
    ci->read_filebuf(ape_ctx.seektable, ape_ctx.numseekpoints * sizeof(uint32_t));

#ifdef ROCKBOX_BIG_ENDIAN
    /* Byte-swap the little-endian seekpoints */
    {
        uint32_t i;

        for(i = 0; i < ape_ctx.numseekpoints; i++)
            ape_ctx.seektable[i] = swap32(ape_ctx.seektable[i]);
    }
#endif

    /* Now advance the file position to the first frame */
    ci->advance_buffer(ape_ctx.firstframe - 
                       (ape_ctx.seektablefilepos +
                        ape_ctx.numseekpoints * sizeof(uint32_t)));

    ci->configure(DSP_SWITCH_FREQUENCY, ape_ctx.samplerate);
    ci->configure(DSP_SET_STEREO_MODE, ape_ctx.channels == 1 ?
                  STEREO_MONO : STEREO_NONINTERLEAVED);
    codec_set_replaygain(ci->id3);

    /* The main decoding loop */

    if (resume_offset) {
        /* The resume offset is a value in bytes - we need to
           turn it into a frame number and samplestoskip value */

        ape_resume(&ape_ctx, resume_offset, 
                   &currentframe, &samplesdone, &samplestoskip, &firstbyte);
    } else {
        currentframe = 0;
        samplesdone = 0;
        samplestoskip = 0;
        firstbyte = 3;  /* Take account of the little-endian 32-bit byte ordering */
    }

    /* Initialise the buffer */
    inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);

    /* The main decoding loop - we decode the frames a small chunk at a time */
    while (currentframe < ape_ctx.totalframes)
    {
frame_start:
        /* Calculate how many blocks there are in this frame */
        if (currentframe == (ape_ctx.totalframes - 1))
            nblocks = ape_ctx.finalframeblocks;
        else
            nblocks = ape_ctx.blocksperframe;

        ape_ctx.currentframeblocks = nblocks;

        /* Initialise the frame decoder */
        init_frame_decoder(&ape_ctx, inbuffer, &firstbyte, &bytesconsumed);

        ci->advance_buffer(bytesconsumed);
        inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);

        /* Decode the frame a chunk at a time */
        while (nblocks > 0)
        {
            ci->yield();
            if (ci->stop_codec || ci->new_track) {
                goto done;
            }

            /* Deal with any pending seek requests */
            if (ci->seek_time) 
            {
                if (ape_calc_seekpos(&ape_ctx,
                    ((ci->seek_time-1)/10) * (ci->id3->frequency/100),
                    &currentframe,
                    &newfilepos,
                    &samplestoskip))
                {
                    samplesdone = currentframe * ape_ctx.blocksperframe;

                    /* APE's bytestream is weird... */
                    firstbyte = 3 - (newfilepos & 3);
                    newfilepos &= ~3;

                    ci->seek_buffer(newfilepos);
                    inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);

                    ci->seek_complete();
                    goto frame_start;  /* Sorry... */
                }
                ci->seek_complete();
            }

            blockstodecode = MIN(BLOCKS_PER_LOOP, nblocks);

            if ((res = decode_chunk(&ape_ctx, inbuffer, &firstbyte,
                                    &bytesconsumed,
                                    decoded0, decoded1,
                                    blockstodecode)) < 0)
            {
                /* Frame decoding error, abort */
                LOGF("APE: Frame %lu, error %d\n",(unsigned long)currentframe,res);
                retval = CODEC_ERROR;
                goto done;
            }

            ci->yield();

            if (samplestoskip > 0) {
                if (samplestoskip < blockstodecode) {
                    ci->pcmbuf_insert(decoded0 + samplestoskip, 
                                      decoded1 + samplestoskip, 
                                      blockstodecode - samplestoskip);
                    samplestoskip = 0;
                } else {
                    samplestoskip -= blockstodecode;
                }
            } else {
                ci->pcmbuf_insert(decoded0, decoded1, blockstodecode);
            }
        
            samplesdone += blockstodecode;

            if (!samplestoskip) {
                /* Update the elapsed-time indicator */
                elapsedtime = (samplesdone*10)/(ape_ctx.samplerate/100);
                ci->set_elapsed(elapsedtime);
            }

            ci->advance_buffer(bytesconsumed);
            inbuffer = ci->request_buffer(&bytesleft, INPUT_CHUNKSIZE);

            /* Decrement the block count */
            nblocks -= blockstodecode;
        }

        currentframe++;
    }

done:
    LOGF("APE: Decoded %lu samples\n",(unsigned long)samplesdone);

    if (ci->request_next_track())
        goto next_track;

exit:
    return retval;
}
コード例 #27
0
int main(int argc, char * const argv[])
{
	int ret = CODEC_ERROR_NONE;
  std::string input_filename;

  if (argc > 1) {
    for (int i = 1; i < argc; i++) {
      if (strncasecmp(argv[i], "--input", 7) == 0) {
        // check the next arg with the proper value.
        int next = i + 1;
        if (next < argc) {
          input_filename = argv[next];
          i++;
        }
      } else if (strncasecmp(argv[i], "-h", 2) == 0 || strncasecmp(argv[i], "--help", 6) == 0) {
        printf("Usage: %s [OPTIONS]...\n", argv[0]);
        printf("Arguments:\n");
        printf("  --input <filename> \tInput video filename\n");
        exit(0);
      }
    }
  }
  if (input_filename.empty()) {
    printf("no input file specified\n");
    exit(0);
  }

  // install signal handlers
  signal(SIGINT, signal_handler);
  signal(SIGTERM, signal_handler);

  // initialize App contex.
  AVPacket avpkt;
  AppContext ctx;
  memset(&ctx, 0, sizeof(ctx));

  player_para_init(&ctx);
  am_packet_init(&ctx.am_pkt);

  ctx.am_pkt.avpkt = &avpkt;
  av_init_packet(ctx.am_pkt.avpkt);

  // create the ffmepg file reader/demuxer
  ctx.demuxer = new FFmpegFileReader(input_filename.c_str());
  if (!ctx.demuxer->Initialize()) {
    fprintf(stderr, "ERROR: Can't initialize FFmpegFileReader\n");
    goto fail;
  }
  
  ctx.codec_context = ctx.demuxer->GetCodecContext();
  ctx.format_context = ctx.demuxer->GetFormatContext();
  if (!ctx.codec_context) {
    fprintf(stderr, "ERROR: Invalid FFmpegFileReader Codec Context\n");
    goto fail;
  }

  dump_extradata(&ctx);

  printf("video width(%d), height(%d), extradata_size(%d)\n",
    (int)ctx.codec_context->width, (int)ctx.codec_context->height, ctx.codec_context->extradata_size);

  AVStream *pStream;
  pStream = ctx.format_context->streams[ctx.demuxer->GetVideoIndex()];
  //ctx.vstream_info.video_pid = (unsigned short)ctx.codec_context->id;
  if (pStream->time_base.den) {
    ctx.vstream_info.start_time     = pStream->start_time * pStream->time_base.num * PTS_FREQ / pStream->time_base.den;
    ctx.vstream_info.video_duration = ((float)pStream->time_base.num / pStream->time_base.den) * UNIT_FREQ;
    ctx.vstream_info.video_pts      = ((float)pStream->time_base.num / pStream->time_base.den) * PTS_FREQ;
  }
  ctx.vstream_info.video_width  = ctx.codec_context->width;
  ctx.vstream_info.video_height = ctx.codec_context->height;
  ctx.vstream_info.video_ratio  = (float)pStream->sample_aspect_ratio.num / pStream->sample_aspect_ratio.den;
  if (ctx.codec_context->time_base.den) {
    ctx.vstream_info.video_codec_rate = (int64_t)UNIT_FREQ * ctx.codec_context->time_base.num / ctx.codec_context->time_base.den;
  }
  if (pStream->r_frame_rate.num) {
    ctx.vstream_info.video_rate = (int64_t)UNIT_FREQ * pStream->r_frame_rate.den / pStream->r_frame_rate.num;
  }


  log_print("time_base.num(%d), pStream->time_base.den(%d)\n",
    pStream->time_base.num, pStream->time_base.den);
    ctx.vstream_info.video_duration = ((float)pStream->time_base.num / pStream->time_base.den) * UNIT_FREQ;
    ctx.vstream_info.video_pts      = ((float)pStream->time_base.num / pStream->time_base.den) * PTS_FREQ;

  printf("\n*********AMLOGIC CODEC PLAYER DEMO************\n\n");
	osd_blank("/sys/class/graphics/fb0/blank", 1);
	osd_blank("/sys/class/graphics/fb1/blank", 1);
	//osd_blank("/sys/class/tsync/enable", 1);
  osd_blank("/sys/class/tsync/enable", 0);

  switch(ctx.codec_context->codec_id)
  {
    case CODEC_ID_H264:
      printf("CODEC_ID_H264\n");

      ctx.vcodec.has_video = 1;
      ctx.vcodec.has_audio = 0;
      ctx.stream_type = STREAM_ES;
      //ctx.vcodec.video_pid = ctx.demuxer->GetVideoIndex();
      ctx.vcodec.video_type = VFORMAT_H264;
      ctx.vcodec.stream_type = STREAM_TYPE_ES_VIDEO;
      //ctx.vcodec.noblock = !!p_para->buffering_enable;
      ctx.vcodec.am_sysinfo.format = VIDEO_DEC_FORMAT_H264;
      ctx.vcodec.am_sysinfo.width  = ctx.vstream_info.video_width;
      ctx.vcodec.am_sysinfo.height = ctx.vstream_info.video_height;
      ctx.vcodec.am_sysinfo.rate   = ctx.vstream_info.video_rate;
      ctx.vcodec.am_sysinfo.ratio  = ctx.vstream_info.video_ratio;
      ctx.vcodec.am_sysinfo.param = (void*)(EXTERNAL_PTS | SYNC_OUTSIDE);

    break;
    case CODEC_ID_MPEG4:
      printf("CODEC_ID_MPEG4\n");
      ctx.vcodec.video_type = VFORMAT_MPEG4;
    break;
    case CODEC_ID_MPEG2VIDEO:
      printf("CODEC_ID_MPEG2VIDEO\n");
      ctx.vcodec.video_type = VFORMAT_MPEG12;
    break;
    default:
      fprintf(stderr, "ERROR: Invalid FFmpegFileReader Codec Format (not h264/mpeg4) = %d\n",
        ctx.codec_context->codec_id);
      goto fail;
    break;
  }

	ret = codec_init(&ctx.vcodec);
	if(ret != CODEC_ERROR_NONE)
	{
		printf("codec init failed, ret=-0x%x", -ret);
		return -1;
	}
	printf("video codec ok!\n");


	ret = codec_init_cntl(&ctx.vcodec);
	if( ret != CODEC_ERROR_NONE )
	{
		printf("codec_init_cntl error\n");
		return -1;
	}

	codec_set_cntl_avthresh(&ctx.vcodec, AV_SYNC_THRESH);
	codec_set_cntl_syncthresh(&ctx.vcodec, ctx.vcodec.has_audio);

  {
    int frame_count, total = 0;
    int64_t bgn_us, end_us;

    ctx.am_pkt.codec = &ctx.vcodec;

    pre_header_feeding(&ctx, &ctx.am_pkt);

    ctx.demuxer->Read(ctx.am_pkt.avpkt);
    printf("byte_count(%d), dts(%llu), pts(%llu)\n",
      ctx.am_pkt.avpkt->size, ctx.am_pkt.avpkt->dts, ctx.am_pkt.avpkt->pts);
    if (!ctx.am_pkt.avpkt->size) {
      fprintf(stderr, "ERROR: Zero bytes read from input\n");
      //goto fail;
    }
    ctx.am_pkt.type = CODEC_VIDEO;
    ctx.am_pkt.data = ctx.am_pkt.avpkt->data;
    ctx.am_pkt.data_size = ctx.am_pkt.avpkt->size;
    ctx.am_pkt.avpkt_newflag = 1;
    ctx.am_pkt.avpkt_isvalid = 1;

    frame_count = 0;
    bool done = false;
    struct buf_status vbuf;
    struct vdec_status vdec;
    float  vlevel;
    while (!g_signal_abort && !done && (frame_count < 5000)) {
      h264_update_frame_header(&ctx.am_pkt);
      
      bgn_us = CurrentHostCounter() * 1000 / CurrentHostFrequency();

      log_print("avpts(%lld), avdts(%lld)\n",
        ctx.am_pkt.avpkt->pts, ctx.am_pkt.avpkt->dts);

      ret = write_av_packet(&ctx, &ctx.am_pkt);

      end_us = CurrentHostCounter() * 1000 / CurrentHostFrequency();

      frame_count++;
      //fprintf(stdout, "decode time(%llu) us\n", end_us-bgn_us);
      check_vcodec_state(&ctx.vcodec, &vdec, &vbuf);
      vlevel = 100.0 * (float)vbuf.data_len / vbuf.size;
      //log_print("buffering_states,vlevel=%d,vsize=%d,level=%f\n", vbuf.data_len, vbuf.size, vlevel);
      usleep(vlevel * 5000);

      ctx.demuxer->Read(ctx.am_pkt.avpkt);
      ctx.am_pkt.type = CODEC_VIDEO;
      ctx.am_pkt.data = ctx.am_pkt.avpkt->data;
      ctx.am_pkt.data_size = ctx.am_pkt.avpkt->size;
      ctx.am_pkt.avpkt_newflag = 1;
      ctx.am_pkt.avpkt_isvalid = 1;

      if (!ctx.am_pkt.data_size) done = true;
      total += ctx.am_pkt.data_size;
    }
  }

fail:
	codec_close_cntl(&ctx.vcodec);
	codec_close(&ctx.vcodec);

  return 0;
}
コード例 #28
0
ファイル: speex.c プロジェクト: lyzkov/rockbox
/* this is called for each file to process */
enum codec_status codec_run(void)
{
    int error = CODEC_ERROR;

    SpeexBits bits;
    int eof = 0;
    spx_ogg_sync_state oy;
    spx_ogg_page og;
    spx_ogg_packet op;
    spx_ogg_stream_state os;
    spx_int64_t page_granule = 0;
    spx_int64_t cur_granule = 0;
    int enh_enabled = 1;
    int nframes = 2;
    int eos = 0;
    SpeexStereoState *stereo;
    int channels = -1;
    int samplerate = ci->id3->frequency;
    int extra_headers = 0;
    int stream_init = 0;
    /* rockbox: comment 'set but unused' variables
    int page_nb_packets;
    */
    int frame_size;
    int packet_count = 0;
    int lookahead;
    int headerssize = 0;
    unsigned long strtoffset = ci->id3->offset;
    void *st = NULL;
    int j = 0;
    intptr_t param;

    memset(&bits, 0, sizeof(bits));
    memset(&oy, 0, sizeof(oy));

    /* Ogg handling still uses mallocs, so reset the malloc buffer per track */
    if (codec_init()) {
        goto exit;
    }

    ci->seek_buffer(0);
    ci->set_elapsed(0);

    stereo = speex_stereo_state_init();
    spx_ogg_sync_init(&oy);
    spx_ogg_alloc_buffer(&oy,2*CHUNKSIZE);

    codec_set_replaygain(ci->id3);

    eof = 0;
    while (!eof) {
        enum codec_command_action action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        /*seek (seeks to the page before the position) */
        if (action == CODEC_ACTION_SEEK_TIME) {
            if(samplerate!=0&&packet_count>1){
                LOGF("Speex seek page:%lld,%lld,%ld,%lld,%d\n",
                     ((spx_int64_t)param/1000) *
                     (spx_int64_t)samplerate,
                     page_granule, param,
                     (page_granule/samplerate)*1000, samplerate);

                speex_seek_page_granule(((spx_int64_t)param/1000) *
                                        (spx_int64_t)samplerate,
                                        page_granule, &oy, headerssize);
            }

            ci->set_elapsed(param);
            ci->seek_complete();
        }

next_page:
        /*Get the ogg buffer for writing*/
        if(get_more_data(&oy)<1){/*read error*/
            goto done;
        }

        /* Loop for all complete pages we got (most likely only one) */
        while (spx_ogg_sync_pageout(&oy, &og) == 1) {
            int packet_no;
            if (stream_init == 0) {
                spx_ogg_stream_init(&os, spx_ogg_page_serialno(&og));
                stream_init = 1;
            }

            /* Add page to the bitstream */
            spx_ogg_stream_pagein(&os, &og);

            page_granule = spx_ogg_page_granulepos(&og);
            /* page_nb_packets = spx_ogg_page_packets(&og); */

            cur_granule = page_granule;

            /* Extract all available packets */
            packet_no=0;

            while (!eos && spx_ogg_stream_packetout(&os, &op)==1){
                /* If first packet, process as Speex header */
                if (packet_count==0){
                    st = process_header(&op, enh_enabled, &frame_size,
                                         &samplerate, &nframes, &channels,
                                         stereo, &extra_headers);

                    speex_decoder_ctl(st, SPEEX_GET_LOOKAHEAD, &lookahead);
                    if (!nframes)
                        nframes=1;

                    if (!st){
                        goto done;
                    }

                    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
                    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
                    if (channels == 2) {
                        ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
                    } else if (channels == 1) {
                        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
                    }

                    /* Speex header in its own page, add the whole page
                       headersize */
                    headerssize += og.header_len+og.body_len;

                } else if (packet_count<=1+extra_headers){
                    /* add packet to headersize */
                    headerssize += op.bytes;

                    /* Ignore extra headers */
                } else {
                    if (packet_count <= 2+extra_headers) {
                        if (strtoffset) {
                            ci->seek_buffer(strtoffset);
                            spx_ogg_sync_reset(&oy);
                            packet_count++;
                            goto next_page;
                        }
                    }
                    packet_no++;

                    if (op.e_o_s) /* End of stream condition */
                        eos=1;

                    /* Set Speex bitstream to point to Ogg packet */
                    speex_bits_set_bit_buffer(&bits, (char *)op.packet,
                                                     op.bytes);
                    for (j = 0; j != nframes; j++){
                        int ret;

                        /* Decode frame */
                        ret = speex_decode_int(st, &bits, output);

                        if (ret == -1)
                            break;

                        if (ret == -2)
                            break;

                        if (speex_bits_remaining(&bits) < 0)
                            break;

                        if (channels == 2)
                            speex_decode_stereo_int(output, frame_size, stereo);

                        if (frame_size > 0) {
                            spx_int16_t *frame_start = output + lookahead;

                            if (channels == 2)
                                frame_start += lookahead;
                            ci->pcmbuf_insert(frame_start, NULL,
                                              frame_size - lookahead);
                            lookahead = 0;
                            /* 2 bytes/sample */
                            cur_granule += frame_size / 2;

                            ci->set_offset((long) ci->curpos);

                            ci->set_elapsed((samplerate == 0) ? 0 :
                                             cur_granule * 1000 / samplerate);
                         }
                    }
                }
                packet_count++;
            }
        }
    }

    error = CODEC_OK;
done:
    /* Clean things up for the next track */
    speex_bits_destroy(&bits);

    if (st)
        speex_decoder_destroy(st);

    if (stream_init)
       spx_ogg_stream_destroy(&os);

    spx_ogg_sync_destroy(&oy);

exit:
    return error;
}
コード例 #29
0
ファイル: asap.c プロジェクト: BurntBrunch/rockbox-fft
/* this is the codec entry point */
enum codec_status codec_main(void)
{
    int n_bytes;
    int song;
    int duration;
    char* module;
    int bytesPerSample =2;
    
next_track:
    if (codec_init()) {
        DEBUGF("codec init failed\n");
        return CODEC_ERROR;
    }

    while (!*ci->taginfo_ready && !ci->stop_codec)
        ci->sleep(1);

    codec_set_replaygain(ci->id3);
        
    int bytes_done =0;   
    size_t filesize;
    ci->seek_buffer(0);
    module = ci->request_buffer(&filesize, ci->filesize);
    if (!module || (size_t)filesize < (size_t)ci->filesize) 
    {
        DEBUGF("loading error\n");
        return CODEC_ERROR;
    }

    /*Init ASAP */
    if (!ASAP_Load(&asap, ci->id3->path, module, filesize))
    {
        DEBUGF("%s: format not supported",ci->id3->path);
        return CODEC_ERROR;
    }  
    
      /* Make use of 44.1khz */
    ci->configure(DSP_SET_FREQUENCY, 44100);
    /* Sample depth is 16 bit little endian */
    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
    /* Stereo or Mono output ? */
    if(asap.module_info.channels ==1)
    {
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
        bytesPerSample = 2;
    }
    else
    {
        ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
        bytesPerSample = 4; 
    }    
    /* reset eleapsed */
    ci->set_elapsed(0);

    song = asap.module_info.default_song;
    duration = asap.module_info.durations[song];
    if (duration < 0)
        duration = 180 * 1000;
    
    /* set id3 length, because metadata parse might not have done it */
    ci->id3->length = duration;
    
    ASAP_PlaySong(&asap, song, duration);
    ASAP_MutePokeyChannels(&asap, 0);
    
    /* The main decoder loop */    
    while (1) {
        ci->yield();
        if (ci->stop_codec || ci->new_track)
            break;

        if (ci->seek_time) {
            /* New time is ready in ci->seek_time */
                      
            /* seek to pos */
            ASAP_Seek(&asap,ci->seek_time);
            /* update elapsed */
            ci->set_elapsed(ci->seek_time);
            /* update bytes_done */
            bytes_done = ci->seek_time*44.1*2;    
            /* seek ready */    
            ci->seek_complete();            
        }
        
        /* Generate a buffer full of Audio */
        #ifdef ROCKBOX_LITTLE_ENDIAN
        n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_LE);
        #else
        n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_BE);
        #endif
        
        ci->pcmbuf_insert(samples, NULL, n_bytes /bytesPerSample);
        
        bytes_done += n_bytes;
        ci->set_elapsed((bytes_done / 2) / 44.1);
        
        if(n_bytes != sizeof(samples))
            break;
    }

    if (ci->request_next_track())
        goto next_track;
 
    return CODEC_OK;    
}
コード例 #30
0
ファイル: alac.c プロジェクト: ifroz/rockbox
/* this is called for each file to process */
enum codec_status codec_run(void)
{
    size_t n;
    demux_res_t demux_res;
    stream_t input_stream;
    uint32_t samplesdone;
    uint32_t elapsedtime = 0;
    int samplesdecoded;
    unsigned int i;
    unsigned char* buffer;
    alac_file alac;
    intptr_t param;

    /* Clean and initialize decoder structures */
    memset(&demux_res , 0, sizeof(demux_res));
    if (codec_init()) {
        LOGF("ALAC: Error initialising codec\n");
        return CODEC_ERROR;
    }

    ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
    codec_set_replaygain(ci->id3);

    ci->seek_buffer(0);

    stream_create(&input_stream,ci);

    /* Read from ci->id3->offset before calling qtmovie_read. */
    samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /  
                  (ci->id3->bitrate*128));
  
    /* if qtmovie_read returns successfully, the stream is up to
     * the movie data, which can be used directly by the decoder */
    if (!qtmovie_read(&input_stream, &demux_res)) {
        LOGF("ALAC: Error initialising file\n");
        return CODEC_ERROR;
    }

    /* initialise the sound converter */
    alac_set_info(&alac, demux_res.codecdata);
  
    /* Set i for first frame, seek to desired sample position for resuming. */
    i=0;
    if (samplesdone > 0) {
        if (m4a_seek(&demux_res, &input_stream, samplesdone,
                      &samplesdone, (int*) &i)) {
            elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
            ci->set_elapsed(elapsedtime);
        } else {
            samplesdone = 0;
        }
    }

    ci->set_elapsed(elapsedtime);

    /* The main decoding loop */
    while (i < demux_res.num_sample_byte_sizes) {
        enum codec_command_action action = ci->get_command(&param);

        if (action == CODEC_ACTION_HALT)
            break;

        /* Request the required number of bytes from the input buffer */
        buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);

        /* Deal with any pending seek requests */
        if (action == CODEC_ACTION_SEEK_TIME) {
            if (m4a_seek(&demux_res, &input_stream,
                         (param/10) * (ci->id3->frequency/100),
                         &samplesdone, (int *)&i)) {
                elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
            }
            ci->set_elapsed(elapsedtime);
            ci->seek_complete();
        }

        /* Request the required number of bytes from the input buffer */
        buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);

        /* Decode one block - returned samples will be host-endian */
        samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
        ci->yield();

        /* Advance codec buffer by amount of consumed bytes */
        ci->advance_buffer(alac.bytes_consumed);

        /* Output the audio */
        ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);

        /* Update the elapsed-time indicator */
        samplesdone+=samplesdecoded;
        elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
        ci->set_elapsed(elapsedtime);

        i++;
    }

    LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
    return CODEC_OK;
}