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audio.c
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/*
* rpihddevice - VDR HD output device for Raspberry Pi
* Copyright (C) 2014, 2015, 2016 Thomas Reufer
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "audio.h"
#include "setup.h"
#include <vdr/tools.h>
#include <vdr/remux.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavutil/log.h>
#include <libavutil/opt.h>
#ifdef ENABLE_AAC_LATM
#warning "experimental AAC-LATM frame parser enabled, only 2ch/48kHz supported!"
#endif
// ffmpeg's resampling
#ifdef HAVE_LIBSWRESAMPLE
# include <libswresample/swresample.h>
# define DO_RESAMPLE
#endif
// libav's resampling
#ifdef HAVE_LIBAVRESAMPLE
# include <libavresample/avresample.h>
# include <libavutil/samplefmt.h>
# define DO_RESAMPLE
# define SwrContext AVAudioResampleContext
# define swr_alloc avresample_alloc_context
# define swr_init avresample_open
# define swr_free avresample_free
# define swr_convert(ctx, dst, out_cnt, src, in_cnt) \
avresample_convert(ctx, dst, 0, out_cnt, (uint8_t**)src, 0, in_cnt)
#endif
#if LIBAVCODEC_VERSION_MAJOR > 58
# define avcodec_register_all()
#endif
#if LIBAVCODEC_VERSION_MAJOR == 58 && LIBAVCODEC_VERSION_MINOR > 9
# define avcodec_register_all()
#endif
#if LIBAVCODEC_VERSION_MAJOR < 56
# define av_packet_unref av_free_packet
#endif
// legacy libavcodec
#if LIBAVCODEC_VERSION_MAJOR < 55
# define av_frame_alloc avcodec_alloc_frame
# define av_frame_free avcodec_free_frame
# define av_frame_unref avcodec_get_frame_defaults
# define AV_CODEC_ID_MP3 CODEC_ID_MP3
# define AV_CODEC_ID_AC3 CODEC_ID_AC3
# define AV_CODEC_ID_EAC3 CODEC_ID_EAC3
# define AV_CODEC_ID_AAC CODEC_ID_AAC
# define AV_CODEC_ID_AAC_LATM CODEC_ID_AAC_LATM
# define AV_CODEC_ID_DTS CODEC_ID_DTS
#endif
#if LIBAVCODEC_VERSION_MAJOR < 54
# define avcodec_free_frame av_free
#endif
// prevent depreciated warnings for >ffmpeg-1.2.x and >libav-9.x
#if LIBAVCODEC_VERSION_MAJOR > 54
# undef FF_API_REQUEST_CHANNELS
#endif
}
#include <string.h>
#define AVPKT_BUFFER_SIZE (KILOBYTE(256))
class cRpiAudioDecoder::cParser
{
public:
cParser() :
m_codec(cAudioCodec::eInvalid),
m_channels(0),
m_samplingRate(0),
m_size(0),
m_parsed(true)
{
}
AVPacket* Packet(void)
{
return &m_packet;
}
cAudioCodec::eCodec GetCodec(void)
{
Parse();
return m_codec;
}
unsigned int GetChannels(void)
{
Parse();
return m_channels;
}
unsigned int GetSamplingRate(void)
{
Parse();
return m_samplingRate;
}
unsigned int GetFrameSize(void)
{
Parse();
return m_packet.size;
}
int64_t GetPts(void)
{
int64_t pts = OMX_INVALID_PTS;
m_mutex.Lock();
if (!m_ptsQueue.empty())
pts = m_ptsQueue.front().pts;
m_mutex.Unlock();
return pts;
}
unsigned int GetFreeSpace(void)
{
return AVPKT_BUFFER_SIZE - m_size - AV_INPUT_BUFFER_PADDING_SIZE;
}
bool Empty(void)
{
Parse();
return m_packet.size == 0;
}
int Init(void)
{
if (!av_new_packet(&m_packet, AVPKT_BUFFER_SIZE))
{
Reset();
return 0;
}
return -1;
}
int DeInit(void)
{
av_packet_unref(&m_packet);
return 0;
}
void Reset(void)
{
m_mutex.Lock();
m_codec = cAudioCodec::eInvalid;
m_channels = 0;
m_samplingRate = 0;
m_packet.size = 0;
m_size = 0;
m_parsed = true; // parser is empty, no need for parsing
memset(m_packet.data, 0, AV_INPUT_BUFFER_PADDING_SIZE);
m_ptsQueue.clear();
m_mutex.Unlock();
}
bool Append(const unsigned char *data, int64_t pts, unsigned int length)
{
bool ret = true;
m_mutex.Lock();
if (m_size + length + AV_INPUT_BUFFER_PADDING_SIZE > AVPKT_BUFFER_SIZE)
ret = false;
else
{
memcpy(m_packet.data + m_size, data, length);
m_size += length;
memset(m_packet.data + m_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
m_ptsQueue.emplace(Pts(pts, length));
m_parsed = false;
}
m_mutex.Unlock();
return ret;
}
void Shrink(unsigned int length, bool retainPts = false)
{
m_mutex.Lock();
if (length < m_size)
{
memmove(m_packet.data, m_packet.data + length, m_size - length);
m_size -= length;
memset(m_packet.data + m_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
while (!m_ptsQueue.empty())
{
Pts& front = m_ptsQueue.front();
if (front.length <= length)
{
length -= front.length;
m_ptsQueue.pop();
if (!length)
break;
}
else
{
// clear current PTS since it's not valid anymore after
// shrinking the packet
if (!retainPts)
front.pts = OMX_INVALID_PTS;
front.length -= length;
break;
}
}
m_parsed = false;
}
else
Reset();
m_mutex.Unlock();
}
private:
cParser(const cParser&);
cParser& operator= (const cParser&);
// Check format of first audio packet in buffer. If format has been
// guessed, but packet is not yet complete, codec is set with a length
// of 0. Once the buffer contains either the exact amount of expected
// data or another valid packet start after the first frame, packet
// size is set to the first frame length.
// Valid packets are always moved to the buffer start, if no valid
// audio frame has been found, packet gets cleared.
void Parse()
{
cAudioCodec::eCodec codec = cAudioCodec::eInvalid;
unsigned int channels = 0;
unsigned int offset = 0;
unsigned int frameSize = 0;
unsigned int samplingRate = 0;
m_mutex.Lock();
if (m_parsed)
goto done;
while (m_size - offset >= 4)
{
// 0xFFE... MPEG audio
// 0x0B77... (E)AC-3 audio
// 0xFFF... AAC audio
// 0x7FFE8001... DTS audio
// PCM audio can't be found
const uint8_t *p = m_packet.data + offset;
unsigned int n = m_size - offset;
switch (FastCheck(p))
{
case cAudioCodec::eMPG:
if (MpegCheck(p, n, frameSize, channels, samplingRate))
codec = cAudioCodec::eMPG;
break;
case cAudioCodec::eAC3:
if (Ac3Check(p, n, frameSize, channels, samplingRate))
{
codec = cAudioCodec::eAC3;
if (n > 5 && p[5] > (10 << 3))
codec = cAudioCodec::eEAC3;
}
break;
case cAudioCodec::eAAC:
if (AdtsCheck(p, n, frameSize, channels, samplingRate))
codec = cAudioCodec::eAAC;
break;
#ifdef ENABLE_AAC_LATM
case cAudioCodec::eAAC_LATM:
if (LatmCheck(p, n, frameSize, channels, samplingRate))
codec = cAudioCodec::eAAC_LATM;
break;
#endif
case cAudioCodec::eDTS:
if (DtsCheck(p, n, frameSize, channels, samplingRate))
codec = cAudioCodec::eDTS;
break;
default:
break;
}
if (codec != cAudioCodec::eInvalid)
{
// if there is enough data in buffer, check if predicted next
// frame start is valid
if (n < frameSize + 4 ||
FastCheck(p + frameSize) != cAudioCodec::eInvalid)
{
// if codec has been detected but buffer does not yet
// contains a complete frame, set size to zero to prevent
// frame from being decoded
if (frameSize > n)
frameSize = 0;
break;
}
}
++offset;
}
if (offset)
{
DBG("audio parser skipped %u of %u bytes", offset, m_size);
Shrink(offset, true);
}
if (codec != cAudioCodec::eInvalid)
{
m_codec = codec;
m_channels = channels;
m_samplingRate = samplingRate;
m_packet.size = frameSize;
}
else
m_packet.size = 0;
m_parsed = true;
done:
m_mutex.Unlock();
}
struct Pts
{
Pts(int64_t _pts, unsigned int _length)
: pts(_pts), length(_length) { };
int64_t pts;
unsigned int length;
};
struct PtsQueue : public std::queue<Pts>
{
void clear() { c.clear(); }
};
cMutex m_mutex;
AVPacket m_packet;
cAudioCodec::eCodec m_codec;
unsigned int m_channels;
unsigned int m_samplingRate;
unsigned int m_size;
PtsQueue m_ptsQueue;
bool m_parsed;
/* ---------------------------------------------------------------------- */
/* audio codec parser helper functions, based on vdr-softhddevice */
/* ---------------------------------------------------------------------- */
static const uint16_t BitRateTable[2][3][16];
static const uint16_t MpegSampleRateTable[4];
static const uint32_t Mpeg4SampleRateTable[16];
static const uint16_t Ac3SampleRateTable[4];
static const uint16_t Ac3FrameSizeTable[38][3];
static const uint32_t DtsSampleRateTable[16];
static cAudioCodec::eCodec FastCheck(const uint8_t *p)
{
return FastMpegCheck(p) ? cAudioCodec::eMPG :
FastAc3Check (p) ? cAudioCodec::eAC3 :
FastAdtsCheck(p) ? cAudioCodec::eAAC :
#ifdef ENABLE_AAC_LATM
FastLatmCheck(p) ? cAudioCodec::eAAC_LATM :
#endif
FastDtsCheck (p) ? cAudioCodec::eDTS :
cAudioCodec::eInvalid;
}
///
/// Fast check for MPEG audio.
///
/// 0xFFE... MPEG audio
///
static bool FastMpegCheck(const uint8_t *p)
{
if (p[0] != 0xFF) // 11bit frame sync
return false;
if ((p[1] & 0xE0) != 0xE0)
return false;
if ((p[1] & 0x18) == 0x08) // version ID - 01 reserved
return false;
if (!(p[1] & 0x06)) // layer description - 00 reserved
return false;
if ((p[2] & 0xF0) == 0xF0) // bit rate index - 1111 reserved
return false;
if ((p[2] & 0x0C) == 0x0C) // sampling rate index - 11 reserved
return false;
return true;
}
/// Check for MPEG audio.
///
/// 0xFFEx already checked.
///
/// From: http://www.mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm
///
/// AAAAAAAA AAABBCCD EEEEFFGH IIJJKLMM
///
/// o a 11x Frame sync
/// o b 2x MPEG audio version (2.5, reserved, 2, 1)
/// o c 2x Layer (reserved, III, II, I)
/// o e 2x BitRate index
/// o f 2x SampleRate index (41000, 48000, 32000, 0)
/// o g 1x Padding bit
/// o h 1x Private bit
/// o i 2x Channel mode
/// o .. Doesn't care
///
/// frame length:
/// Layer I:
/// FrameLengthInBytes = (12 * BitRate / SampleRate + Padding) * 4
/// Layer II & III:
/// FrameLengthInBytes = 144 * BitRate / SampleRate + Padding
///
static bool MpegCheck(const uint8_t *p, unsigned int size,
unsigned int &frameSize, unsigned int &channels,
unsigned int &samplingRate)
{
frameSize = size;
if (size < 4)
return true;
int cmode = (p[3] >> 6) & 0x03;
int mpeg2 = !(p[1] & 0x08) && (p[1] & 0x10);
int mpeg25 = !(p[1] & 0x08) && !(p[1] & 0x10);
int layer = 4 - ((p[1] >> 1) & 0x03);
int padding = (p[2] >> 1) & 0x01;
// channel mode = [ stereo, joint stereo, dual channel, mono]
channels = cmode == 0x03 ? 1 : 2;
samplingRate = MpegSampleRateTable[(p[2] >> 2) & 0x03];
if (!samplingRate)
return false;
samplingRate >>= mpeg2; // MPEG 2 half rate
samplingRate >>= mpeg25; // MPEG 2.5 quarter rate
int bit_rate =
BitRateTable[mpeg2 | mpeg25][layer - 1][(p[2] >> 4) & 0x0F];
if (!bit_rate)
return false;
switch (layer)
{
case 1:
frameSize = (12000 * bit_rate) / samplingRate;
frameSize = (frameSize + padding) * 4;
break;
case 2:
case 3:
default:
frameSize = (144000 * bit_rate) / samplingRate;
frameSize = frameSize + padding;
break;
}
return true;
}
///
/// Fast check for (E-)AC-3 audio.
///
/// 0x0B77... AC-3 audio
///
static bool FastAc3Check(const uint8_t *p)
{
if (p[0] != 0x0B) // 16bit sync
return false;
if (p[1] != 0x77)
return false;
return true;
}
///
/// Check for (E-)AC-3 audio.
///
/// 0x0B77xxxxxx already checked.
///
/// o AC-3 Header
/// AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF GGGxxxxx
///
/// o a 16x Frame sync, always 0x0B77
/// o b 16x CRC 16
/// o c 2x Sample rate ( 48000, 44100, 32000, reserved )
/// o d 6x Frame size code
/// o e 5x Bit stream ID
/// o f 3x Bit stream mode
/// o g 3x Audio coding mode
///
/// o E-AC-3 Header
/// AAAAAAAA AAAAAAAA BBCCCDDD DDDDDDDD EEFFGGGH IIIII...
///
/// o a 16x Frame sync, always 0x0B77
/// o b 2x Frame type
/// o c 3x Sub stream ID
/// o d 11x Frame size - 1 in words
/// o e 2x Frame size code
/// o f 2x Frame size code 2
/// o g 3x Channel mode
/// 0 h 1x LFE on
///
static bool Ac3Check(const uint8_t *p, unsigned int size,
unsigned int &frameSize, unsigned int &channels,
unsigned int &samplingRate)
{
frameSize = size;
if (size < 7)
return true;
int acmod;
bool lfe;
int fscod = (p[4] & 0xC0) >> 6;
samplingRate = Ac3SampleRateTable[fscod];
if (p[5] > (10 << 3)) // E-AC-3
{
if (fscod == 0x03)
{
int fscod2 = (p[4] & 0x30) >> 4;
if (fscod2 == 0x03)
return false; // invalid fscod & fscod2
samplingRate = Ac3SampleRateTable[fscod2] / 2;
}
acmod = (p[4] & 0x0E) >> 1; // number of channels, LFE excluded
lfe = p[4] & 0x01;
frameSize = ((p[2] & 0x07) << 8) + p[3] + 1;
frameSize *= 2;
}
else // AC-3
{
if (fscod == 0x03) // invalid sample rate
return false;
int frmsizcod = p[4] & 0x3F;
if (frmsizcod > 37) // invalid frame size
return false;
acmod = p[6] >> 5; // number of channels, LFE excluded
int lfe_bptr = 51; // position of LFE bit in header for 2.0
if ((acmod & 0x01) && (acmod != 0x01))
lfe_bptr += 2; // skip center mix level
if (acmod & 0x04)
lfe_bptr += 2; // skip surround mix level
if (acmod == 0x02)
lfe_bptr += 2; // skip surround mode
lfe = (p[lfe_bptr / 8] & (1 << (7 - (lfe_bptr % 8))));
// invalid is checked above
frameSize = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
}
channels =
acmod == 0x00 ? 2 : // Ch1, Ch2
acmod == 0x01 ? 1 : // C
acmod == 0x02 ? 2 : // L, R
acmod == 0x03 ? 3 : // L, C, R
acmod == 0x04 ? 3 : // L, R, S
acmod == 0x05 ? 4 : // L, C, R, S
acmod == 0x06 ? 4 : // L, R, RL, RR
acmod == 0x07 ? 5 : 0; // L, C, R, RL, RR
if (lfe) channels++;
return true;
}
#ifdef ENABLE_AAC_LATM
///
/// Fast check for AAC LATM audio.
///
/// 0x56E... AAC LATM audio
///
static bool FastLatmCheck(const uint8_t *p)
{
if (p[0] != 0x56) // 11bit sync
return false;
if ((p[1] & 0xE0) != 0xE0)
return false;
return true;
}
///
/// Check for AAC LATM audio.
///
/// 0x56Exxx already checked.
///
static bool LatmCheck(const uint8_t *p, unsigned int size,
unsigned int &frameSize, unsigned int &channels,
unsigned int &samplingRate)
{
frameSize = size;
if (size < 3)
return true;
// to do: determine channels
channels = 2;
// to do: determine sampling rate
samplingRate = 48000;
// 13 bit frame size without header
frameSize = ((p[1] & 0x1F) << 8) + p[2];
frameSize += 3;
return true;
}
#endif
///
/// Fast check for ADTS Audio Data Transport Stream.
///
/// 0xFFF... ADTS audio
///
static bool FastAdtsCheck(const uint8_t *p)
{
if (p[0] != 0xFF) // 12bit sync
return false;
if ((p[1] & 0xF6) != 0xF0) // sync + layer must be 0
return false;
if ((p[2] & 0x3C) == 0x3C) // sampling frequency index != 15
return false;
return true;
}
///
/// Check for ADTS Audio Data Transport Stream.
///
/// 0xFFF already checked.
///
/// AAAAAAAA AAAABCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP
/// (QQQQQQQQ QQQQQQQ)
///
/// o A*12 sync word 0xFFF
/// o B*1 MPEG Version: 0 for MPEG-4, 1 for MPEG-2
/// o C*2 layer: always 0
/// o ..
/// o F*4 sampling frequency index (15 is invalid)
/// o ..
/// o H*3 MPEG-4 channel configuration
/// o ...
/// o M*13 frame length
///
static bool AdtsCheck(const uint8_t *p, unsigned int size,
unsigned int &frameSize, unsigned int &channels,
unsigned int &samplingRate)
{
frameSize = size;
if (size < 6)
return true;
samplingRate = Mpeg4SampleRateTable[(p[2] >> 2) & 0x0F];
frameSize = (p[3] & 0x03) << 11;
frameSize |= (p[4] & 0xFF) << 3;
frameSize |= (p[5] & 0xE0) >> 5;
int cConf = (p[2] & 0x01) << 7;
cConf |= (p[3] & 0xC0) >> 6;
channels =
cConf == 0x00 ? 0 : // defined in AOT specific config
cConf == 0x01 ? 1 : // C
cConf == 0x02 ? 2 : // L, R
cConf == 0x03 ? 3 : // C, L, R
cConf == 0x04 ? 4 : // C, L, R, RC
cConf == 0x05 ? 5 : // C, L, R, RL, RR
cConf == 0x06 ? 6 : // C, L, R, RL, RR, LFE
cConf == 0x07 ? 8 : // C, L, R, SL, SR, RL, RR, LFE
0;
if (!samplingRate || !channels)
return false;
return true;
}
///
/// Fast check for DTS Audio Data Transport Stream.
///
/// 0x7FFE8001.... DTS audio
///
static bool FastDtsCheck(const uint8_t *p)
{
if (p[0] != 0x7F) // 32bit sync
return false;
if (p[1] != 0xFE)
return false;
if (p[2] != 0x80)
return false;
if (p[3] != 0x01)
return false;
return true;
}
///
/// Check for DTS Audio Data Transport Stream.
///
/// 0x7FFE8001 already checked.
///
/// AAAAAAAA AAAAAAAA AAAAAAAA AAAAAAAA BCCCCCDE EEEEEEFF FFFFFFFF FFFFGGGG
/// GGHHHHII IIIJKLMN OOOPQRRS TTTTTTTT TTTTTTTT UVVVVWWX XXYZaaaa
///
/// o A*32 sync word 0x7FFE8001
/// o B*1 frame type
/// o C*5 deficit sample count
/// o D*1 CRC present flag
/// o E*7 number of PCM sample blocks
/// o F*14 primary frame size
/// o G*6 audio channel arrangement
/// o H*4 core audio sampling frequency
/// o I*5 transmission bit rate
/// o J*1 embedded downmix enabled
/// o K*1 embedded dynamic range flag
/// o L*1 embedded time stamp flag
/// o M*1 auxiliary data flag
/// o N*1 HDCD
/// o O*3 extension audio descriptor flag
/// o P*1 extended coding flag
/// o Q*1 audio sync word insertion flag
/// o R*2 low frequency effects flag
/// o S*1 predictor history flag
/// o T*16 header CRC check (if CRC present flag set)
/// o U*1 multi rate interpolator switch
/// o V*4 encoder software revision
/// o W*2 copy history
/// o X*3 source PCM resolution
/// o Y*1 front sum/difference flag
/// o Z*1 surrounds sum/difference flag
/// o a*4 dialog normalization parameter
///
static bool DtsCheck(const uint8_t *p, unsigned int size,
unsigned int &frameSize, unsigned int &channels,
unsigned int &samplingRate)
{
frameSize = size;
if (size < 11)
return true;
frameSize = ((p[5] & 0x03) << 12) + (p[6] << 4) + ((p[7] & 0xF0) >> 4);
frameSize++;
samplingRate = DtsSampleRateTable[(p[8] & 0x3C) >> 2];
int amode = ((p[7] & 0x0F) << 2) + ((p[8] & 0xC0) >> 6);
channels =
amode == 0x00 ? 1 : // mono
amode == 0x02 ? 2 : // L, R
amode == 0x03 ? 2 : // (L + R), (L - R)
amode == 0x04 ? 2 : // LT, RT
amode == 0x05 ? 3 : // L, R, C
amode == 0x06 ? 3 : // L, R, S
amode == 0x08 ? 4 : // L, R, RL, RR
amode == 0x09 ? 5 : 0; // L, C, R, RL, RR
if (!samplingRate || !channels)
return false;
if (p[10] & 0x06) channels++;
return true;
}
};
///
/// MPEG bit rate table.
///
/// BitRateTable[Version][Layer][Index]
///
const uint16_t cRpiAudioDecoder::cParser::BitRateTable[2][3][16] =
{
{ // MPEG Version 1
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, 0},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 0},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 0}
},
{ // MPEG Version 2 & 2.5
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, 0},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0}
}
};
///
/// MPEG sample rate table.
///
const uint16_t cRpiAudioDecoder::cParser::MpegSampleRateTable[4] =
{ 44100, 48000, 32000, 0 };
///
/// MPEG-4 sample rate table.
///
const uint32_t cRpiAudioDecoder::cParser::Mpeg4SampleRateTable[16] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
///
/// AC-3 sample rate table.
///
const uint16_t cRpiAudioDecoder::cParser::Ac3SampleRateTable[4] =
{ 48000, 44100, 32000, 0 };
///
/// Possible AC-3 frame sizes.
///
/// from ATSC A/52 table 5.18 frame size code table.
///
const uint16_t cRpiAudioDecoder::cParser::Ac3FrameSizeTable[38][3] =
{
{ 64, 69, 96}, { 64, 70, 96}, { 80, 87, 120}, { 80, 88, 120},
{ 96, 104, 144}, { 96, 105, 144}, { 112, 121, 168}, {112, 122, 168},
{ 128, 139, 192}, { 128, 140, 192}, { 160, 174, 240}, {160, 175, 240},
{ 192, 208, 288}, { 192, 209, 288}, { 224, 243, 336}, {224, 244, 336},
{ 256, 278, 384}, { 256, 279, 384}, { 320, 348, 480}, {320, 349, 480},
{ 384, 417, 576}, { 384, 418, 576}, { 448, 487, 672}, {448, 488, 672},
{ 512, 557, 768}, { 512, 558, 768}, { 640, 696, 960}, {640, 697, 960},
{ 768, 835, 1152}, { 768, 836, 1152}, { 896, 975, 1344}, {896, 976, 1344},
{1024, 1114, 1536}, {1024, 1115, 1536}, {1152, 1253, 1728},
{1152, 1254, 1728}, {1280, 1393, 1920}, {1280, 1394, 1920},
};
///
/// DTS sample rate table.
///
const uint32_t cRpiAudioDecoder::cParser::DtsSampleRateTable[16] =
{ 0, 8000, 16000, 32000, 64000,
0, 11025, 22050, 44100, 88200,
0, 12000, 24000, 48000, 96000, 0 };
/* ------------------------------------------------------------------------- */
#define AV_CH_LAYOUT(ch) ( \
ch == 1 ? AV_CH_LAYOUT_MONO : \
ch == 2 ? AV_CH_LAYOUT_STEREO : \
ch == 3 ? AV_CH_LAYOUT_2POINT1 : \
ch == 6 ? AV_CH_LAYOUT_5POINT1 : 0)
#define AV_SAMPLE_STR(fmt) ( \
fmt == AV_SAMPLE_FMT_U8 ? "U8" : \
fmt == AV_SAMPLE_FMT_S16 ? "S16" : \
fmt == AV_SAMPLE_FMT_S32 ? "S32" : \
fmt == AV_SAMPLE_FMT_FLT ? "float" : \
fmt == AV_SAMPLE_FMT_DBL ? "double" : \
fmt == AV_SAMPLE_FMT_U8P ? "U8, planar" : \
fmt == AV_SAMPLE_FMT_S16P ? "S16, planar" : \
fmt == AV_SAMPLE_FMT_S32P ? "S32, planar" : \
fmt == AV_SAMPLE_FMT_FLTP ? "float, planar" : \
fmt == AV_SAMPLE_FMT_DBLP ? "double, planar" : "unknown")
/* ------------------------------------------------------------------------- */
class cRpiAudioRender
{
public:
cRpiAudioRender(cOmx *omx) :
m_omx(omx),
m_port(cRpiAudioPort::eLocal),
m_codec(cAudioCodec::eInvalid),
m_inChannels(0),
m_outChannels(0),
m_samplingRate(0),
m_frameSize(0),
m_configured(false),
m_running(false),
#ifdef DO_RESAMPLE
m_resample(0),
m_resamplerConfigured(false),
#endif
m_pcmSampleFormat(AV_SAMPLE_FMT_NONE),
m_pts(0)
{
}
~cRpiAudioRender()
{
Flush();
#ifdef DO_RESAMPLE
swr_free(&m_resample);
#endif
}
int WriteSamples(uint8_t** data, int samples, int64_t pts,
AVSampleFormat sampleFormat = AV_SAMPLE_FMT_NONE)
{
if (!Ready())
return 0;
int copied = 0;
if (sampleFormat == AV_SAMPLE_FMT_NONE)
{
// pass through
while (samples > copied)
{
OMX_BUFFERHEADERTYPE *buf = m_omx->GetAudioBuffer(pts);
if (!buf)
break;
unsigned int len = samples - copied;
if (len > buf->nAllocLen)
len = buf->nAllocLen;
memcpy(buf->pBuffer, *data + copied, len);
buf->nFilledLen = len;
if (!m_omx->EmptyAudioBuffer(buf))
break;
copied += len;
pts = 0;
}
}
else
{
#ifdef DO_RESAMPLE
// local decode, do resampling
if (!m_resamplerConfigured || m_pcmSampleFormat != sampleFormat)
{
m_pcmSampleFormat = sampleFormat;
ApplyResamplerSettings();
}
if (m_resample)
{
m_pts = pts ? pts : m_pts;
OMX_BUFFERHEADERTYPE *buf = m_omx->GetAudioBuffer(m_pts);
if (buf)
{
if (buf->nAllocLen >= (samples * m_outChannels *
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)))
{
uint8_t *dst[] = { buf->pBuffer };
int copiedSamples = swr_convert(m_resample,
dst, samples, (const uint8_t **)data, samples);
buf->nFilledLen = av_samples_get_buffer_size(NULL,
m_outChannels, copiedSamples, AV_SAMPLE_FMT_S16, 1);
m_pts += copiedSamples * 90000 / m_samplingRate;
}
copied = m_omx->EmptyAudioBuffer(buf) ? samples : 0;
}
}
#else
// local decode, no resampling
if (pts)
m_pts = pts;
OMX_BUFFERHEADERTYPE *buf = m_omx->GetAudioBuffer(m_pts);
if (buf)
{
unsigned int size = samples * m_outChannels *
av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
if (buf->nAllocLen >= size)
{
memcpy(buf->pBuffer, *data, size);
buf->nFilledLen = size;
m_pts += samples * 90000 / m_samplingRate;
}
copied = m_omx->EmptyAudioBuffer(buf) ? samples : 0;
}