diff options
Diffstat (limited to 'audio.c')
| -rw-r--r-- | audio.c | 729 |
1 files changed, 422 insertions, 307 deletions
@@ -13,6 +13,8 @@ #include <string.h> +#define AVPKT_BUFFER_SIZE (64 * 1024) /* 1 PES packet */ + class cAudioParser { @@ -21,13 +23,40 @@ public: cAudioParser() { } ~cAudioParser() { } - inline AVPacket* Packet(void) { return &m_packet; } - inline unsigned int Size(void) { return m_packet.stream_index; } - inline unsigned char *Data(void) { return m_packet.data; } + AVPacket* Packet(void) + { + return &m_packet; + } + + cAudioCodec::eCodec GetCodec(void) + { + if (!m_parsed) + Parse(); + return m_codec; + } + + unsigned int GetChannels(void) + { + if (!m_parsed) + Parse(); + return m_channels; + } + + bool Empty(void) + { + if (!m_parsed) + Parse(); + return m_packet.size == 0; + } int Init(void) { - return av_new_packet(&m_packet, 64 * 1024 /* 1 PES packet */); + if (!av_new_packet(&m_packet, AVPKT_BUFFER_SIZE)) + { + Reset(); + return 0; + } + return -1; } int DeInit(void) @@ -38,42 +67,66 @@ public: void Reset(void) { - m_packet.stream_index = 0; + m_codec = cAudioCodec::eInvalid; + m_channels = 0; + m_packet.size = 0; + m_size = 0; + m_parsed = false; memset(m_packet.data, 0, FF_INPUT_BUFFER_PADDING_SIZE); } bool Append(const unsigned char *data, unsigned int length) { - if (m_packet.stream_index + length + FF_INPUT_BUFFER_PADDING_SIZE > m_packet.size) + if (m_size + length + FF_INPUT_BUFFER_PADDING_SIZE > AVPKT_BUFFER_SIZE) return false; - memcpy(m_packet.data + m_packet.stream_index, data, length); - m_packet.stream_index += length; - memset(m_packet.data + m_packet.stream_index, 0, FF_INPUT_BUFFER_PADDING_SIZE); + memcpy(m_packet.data + m_size, data, length); + m_size += length; + memset(m_packet.data + m_size, 0, FF_INPUT_BUFFER_PADDING_SIZE); + + m_parsed = false; return true; } void Shrink(unsigned int length) { - if (length < m_packet.stream_index) + if (length < m_size) { - memmove(m_packet.data, m_packet.data + length, m_packet.stream_index - length); - m_packet.stream_index -= length; - memset(m_packet.data + m_packet.stream_index, 0, FF_INPUT_BUFFER_PADDING_SIZE); + memmove(m_packet.data, m_packet.data + length, m_size - length); + m_size -= length; + memset(m_packet.data + m_size, 0, FF_INPUT_BUFFER_PADDING_SIZE); + + m_parsed = false; } else Reset(); } - cAudioCodec::eCodec Parse(unsigned int &offset) +private: + + // 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. + // + // To do: + // - parse sampling rate to allow non-48kHz audio + // - consider codec change for next frame check + + void Parse() { cAudioCodec::eCodec codec = cAudioCodec::eInvalid; - - while (Size() - offset >= 5) + int channels = 0; + int offset = 0; + int frameSize = 0; + + while (m_size - offset >= 3) { - const uint8_t *p = Data() + offset; - int n = Size() - offset; - int r = 0; + const uint8_t *p = m_packet.data + offset; + int n = m_size - offset; // 4 bytes 0xFFExxxxx MPEG audio // 3 bytes 0x56Exxx AAC LATM audio @@ -84,49 +137,73 @@ public: if (FastMpegCheck(p)) { - r = MpegCheck(p, n); - codec = cAudioCodec::eMPG; + if (MpegCheck(p, n, frameSize)) + { + codec = cAudioCodec::eMPG; + channels = 2; + } + break; } else if (FastAc3Check(p)) { - r = Ac3Check(p, n); - codec = cAudioCodec::eAC3; - - if (r > 0 && p[5] > (10 << 3)) - codec = cAudioCodec::eEAC3; + if (Ac3Check(p, n, frameSize, channels)) + { + codec = cAudioCodec::eAC3; + if (n > 5 && p[5] > (10 << 3)) + codec = cAudioCodec::eEAC3; + } + break; } else if (FastLatmCheck(p)) { - r = LatmCheck(p, n); - codec = cAudioCodec::eAAC; + if (LatmCheck(p, n, frameSize)) + { + codec = cAudioCodec::eAAC; + channels = 2; + } + break; } else if (FastAdtsCheck(p)) { - r = AdtsCheck(p, n); - codec = cAudioCodec::eDTS; + if (AdtsCheck(p, n, frameSize, channels)) + codec = cAudioCodec::eADTS; + break; } - if (r < 0) // need more bytes - break; + ++offset; + } - if (r > 0) - return codec; + if (codec != cAudioCodec::eInvalid) + { + if (offset) + { + dsyslog("rpihddevice: audio packet shrinked by %d bytes", offset); + Shrink(offset); + } - ++offset; + m_codec = codec; + m_channels = channels; + m_packet.size = frameSize; } - return cAudioCodec::eInvalid; - } + else + Reset(); -private: + m_parsed = true; + } - AVPacket m_packet; + AVPacket m_packet; + cAudioCodec::eCodec m_codec; + unsigned int m_channels; + unsigned int m_size; + bool m_parsed; /* ------------------------------------------------------------------------- */ /* audio codec parser helper functions, taken from vdr-softhddevice */ /* ------------------------------------------------------------------------- */ static const uint16_t BitRateTable[2][3][16]; - static const uint16_t SampleRateTable[4]; + static const uint16_t MpegSampleRateTable[4]; + static const uint16_t Ac3SampleRateTable[4]; static const uint16_t Ac3FrameSizeTable[38][3]; /// @@ -155,13 +232,6 @@ private: /// /// 0xFFEx already checked. /// - /// @param data incomplete PES packet - /// @param size number of bytes - /// - /// @retval <0 possible MPEG audio, but need more data - /// @retval 0 no valid MPEG audio - /// @retval >0 valid MPEG audio - /// /// From: http://www.mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm /// /// AAAAAAAA AAABBCCD EEEEFFGH IIJJKLMM @@ -170,7 +240,7 @@ private: /// 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 (4100, 48000, 32000, 0) + /// o f 2x SampleRate index (41000, 48000, 32000, 0) /// o g 1x Padding bit /// o .. Doesn't care /// @@ -180,24 +250,27 @@ private: /// Layer II & III: /// FrameLengthInBytes = 144 * BitRate / SampleRate + Padding /// - static int MpegCheck(const uint8_t *data, int size) + static bool MpegCheck(const uint8_t *data, int size, int &frame_size) { - int frame_size; + frame_size = 0; + if (size < 3) + return true; + int mpeg2 = !(data[1] & 0x08) && (data[1] & 0x10); int mpeg25 = !(data[1] & 0x08) && !(data[1] & 0x10); int layer = 4 - ((data[1] >> 1) & 0x03); int padding = (data[2] >> 1) & 0x01; - int sample_rate = SampleRateTable[(data[2] >> 2) & 0x03]; + int sample_rate = MpegSampleRateTable[(data[2] >> 2) & 0x03]; if (!sample_rate) - return 0; + return false; sample_rate >>= mpeg2; // MPEG 2 half rate sample_rate >>= mpeg25; // MPEG 2.5 quarter rate int bit_rate = BitRateTable[mpeg2 | mpeg25][layer - 1][(data[2] >> 4) & 0x0F]; if (!bit_rate) - return 0; + return false; switch (layer) { @@ -212,14 +285,14 @@ private: frame_size = frame_size + padding; break; } - if (frame_size + 4 > size) - return -frame_size - 4; - // check if after this frame a new MPEG frame starts - if (cAudioParser::FastMpegCheck(data + frame_size)) - return frame_size; + if (size >= frame_size + 3 && !FastMpegCheck(data + frame_size)) + return false; - return 0; + if (frame_size > size) + frame_size = 0; + + return true; } /// @@ -241,22 +314,16 @@ private: /// /// 0x0B77xxxxxx already checked. /// - /// @param data incomplete PES packet - /// @param size number of bytes - /// - /// @retval <0 possible AC-3 audio, but need more data - /// @retval 0 no valid AC-3 audio - /// @retval >0 valid AC-3 audio - /// /// o AC-3 Header - /// AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF + /// AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF GGGxxxxx /// /// o a 16x Frame sync, always 0x0B77 /// o b 16x CRC 16 - /// o c 2x Sample rate + /// 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... @@ -267,43 +334,76 @@ private: /// o d 10x 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 int Ac3Check(const uint8_t *p, int size) + static bool Ac3Check(const uint8_t *p, int size, int &frame_size, int &channels) { - int frame_size; + frame_size = 0; + if (size < 7) + return true; - if (size < 5) // need 5 bytes to see if AC-3/E-AC-3 - return -5; + int acmod; + bool lfe; + int sample_rate; // for future use, E-AC3 t.b.d. - if (p[5] > (10 << 3)) // E-AC-3 + if (p[5] > (10 << 3)) // E-AC-3 { if ((p[4] & 0xF0) == 0xF0) // invalid fscod fscod2 - return 0; + return false; + + acmod = (p[4] & 0x0E) >> 1; // number of channels, LFE excluded + lfe = p[4] & 0x01; frame_size = ((p[2] & 0x03) << 8) + p[3] + 1; frame_size *= 2; } else // AC-3 { + sample_rate = Ac3SampleRateTable[(p[4] >> 6) & 0x03]; + int fscod = p[4] >> 6; if (fscod == 0x03) // invalid sample rate - return 0; + return false; int frmsizcod = p[4] & 0x3F; if (frmsizcod > 37) // invalid frame size - return 0; + 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 frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2; } - if (frame_size + 5 > size) - return -frame_size - 5; - // check if after this frame a new AC-3 frame starts - if (FastAc3Check(p + frame_size)) - return frame_size; + 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 - return 0; + if (lfe) channels++; + + if (size >= frame_size + 2 && !FastAc3Check(p + frame_size)) + return false; + + if (frame_size > size) + frame_size = 0; + + return true; } /// @@ -325,27 +425,23 @@ private: /// /// 0x56Exxx already checked. /// - /// @param data incomplete PES packet - /// @param size number of bytes - /// - /// @retval <0 possible AAC LATM audio, but need more data - /// @retval 0 no valid AAC LATM audio - /// @retval >0 valid AAC LATM audio - /// - static int LatmCheck(const uint8_t *p, int size) + static bool LatmCheck(const uint8_t *p, int size, int &frame_size) { + frame_size = 0; + if (size < 3) + return true; + // 13 bit frame size without header - int frame_size = ((p[1] & 0x1F) << 8) + p[2]; + frame_size = ((p[1] & 0x1F) << 8) + p[2]; frame_size += 3; - if (frame_size + 2 > size) - return -frame_size - 2; + if (size >= frame_size + 3 && !FastLatmCheck(p + frame_size)) + return false; - // check if after this frame a new AAC LATM frame starts - if (FastLatmCheck(p + frame_size)) - return frame_size; + if (frame_size > size) + frame_size = 0; - return 0; + return true; } /// @@ -369,13 +465,6 @@ private: /// /// 0xFFF already checked. /// - /// @param data incomplete PES packet - /// @param size number of bytes - /// - /// @retval <0 possible ADTS audio, but need more data - /// @retval 0 no valid ADTS audio - /// @retval >0 valid AC-3 audio - /// /// AAAAAAAA AAAABCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP /// (QQQQQQQQ QQQQQQQ) /// @@ -385,25 +474,40 @@ private: /// 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 int AdtsCheck(const uint8_t *p, int size) + static bool AdtsCheck(const uint8_t *p, int size, int &frame_size, int &channels) { - if (size < 6) - return -6; + frame_size = 0; + if (size < 6) + return true; - int frame_size = (p[3] & 0x03) << 11; + frame_size = (p[3] & 0x03) << 11; frame_size |= (p[4] & 0xFF) << 3; frame_size |= (p[5] & 0xE0) >> 5; - if (frame_size + 3 > size) - return -frame_size - 3; + int ch_config = (p[2] & 0x01) << 7; + ch_config |= (p[3] & 0xC0) >> 6; + channels = + ch_config == 0x00 ? 0 : // defined in AOT specific config + ch_config == 0x01 ? 1 : // C + ch_config == 0x02 ? 2 : // L, R + ch_config == 0x03 ? 3 : // C, L, R + ch_config == 0x04 ? 4 : // C, L, R, RC + ch_config == 0x05 ? 5 : // C, L, R, RL, RR + ch_config == 0x06 ? 6 : // C, L, R, RL, RR, LFE + ch_config == 0x07 ? 8 : // C, L, R, SL, SR, RL, RR, LFE + 0; + + if (size >= frame_size + 3 && !FastAdtsCheck(p + frame_size)) + return false; - // check if after this frame a new ADTS frame starts - if (FastAdtsCheck(p + frame_size)) - return frame_size; + if (frame_size > size) + frame_size = 0; - return 0; + return true; } }; @@ -429,7 +533,12 @@ const uint16_t cAudioParser::BitRateTable[2][3][16] = /// /// MPEG sample rate table. /// -const uint16_t cAudioParser::SampleRateTable[4] = { 44100, 48000, 32000, 0 }; +const uint16_t cAudioParser::MpegSampleRateTable[4] = { 44100, 48000, 32000, 0 }; + +/// +/// AC-3 sample rate table. +/// +const uint16_t cAudioParser::Ac3SampleRateTable[4] = { 48000, 44100, 32000, 0 }; /// /// Possible AC-3 frame sizes. @@ -454,17 +563,12 @@ const uint16_t cAudioParser::Ac3FrameSizeTable[38][3] = cAudioDecoder::cAudioDecoder(cOmx *omx) : cThread(), - m_codec(cAudioCodec::eInvalid), - m_outputFormat(cAudioCodec::ePCM), - m_outputPort(cAudioPort::eLocal), - m_channels(0), - m_samplingRate(0), m_passthrough(false), - m_outputFormatChanged(true), + m_reset(false), + m_ready(false), m_pts(0), - m_frame(0), m_mutex(new cMutex()), - m_newData(new cCondWait()), + m_wait(new cCondWait()), m_parser(new cAudioParser()), m_omx(omx) { @@ -473,7 +577,7 @@ cAudioDecoder::cAudioDecoder(cOmx *omx) : cAudioDecoder::~cAudioDecoder() { delete m_parser; - delete m_newData; + delete m_wait; delete m_mutex; } @@ -490,7 +594,7 @@ int cAudioDecoder::Init(void) m_codecs[cAudioCodec::eAC3 ].codec = avcodec_find_decoder(CODEC_ID_AC3); m_codecs[cAudioCodec::eEAC3].codec = avcodec_find_decoder(CODEC_ID_EAC3); m_codecs[cAudioCodec::eAAC ].codec = avcodec_find_decoder(CODEC_ID_AAC_LATM); - m_codecs[cAudioCodec::eDTS ].codec = avcodec_find_decoder(CODEC_ID_DTS); + m_codecs[cAudioCodec::eADTS].codec = avcodec_find_decoder(CODEC_ID_AAC); for (int i = 0; i < cAudioCodec::eNumCodecs; i++) { @@ -513,14 +617,6 @@ int cAudioDecoder::Init(void) } } - m_frame = avcodec_alloc_frame(); - if (!m_frame) - { - esyslog("rpihddevice: failed to allocate audio frame!"); - ret = -1; - } - avcodec_get_frame_defaults(m_frame); - if (ret < 0) DeInit(); @@ -540,41 +636,42 @@ int cAudioDecoder::DeInit(void) av_free(m_codecs[codec].context); } - av_free(m_frame); m_parser->DeInit(); return 0; } -bool cAudioDecoder::WriteData(const unsigned char *buf, unsigned int length, uint64_t pts) +int cAudioDecoder::WriteData(const unsigned char *buf, unsigned int length, uint64_t pts) { - m_mutex->Lock(); - - bool ret = m_parser->Append(buf, length); - if (ret) + int ret = 0; + if (m_ready) { - // set current pts as reference - m_pts = pts; - - ProbeCodec(); - m_newData->Signal(); + m_mutex->Lock(); + if (m_parser->Append(buf, length)) + { + m_pts = pts; + if (!m_parser->Empty()) + { + m_ready = false; + m_wait->Signal(); + } + ret = length; + } + m_mutex->Unlock(); } - - m_mutex->Unlock(); return ret; } void cAudioDecoder::Reset(void) { m_mutex->Lock(); - m_parser->Reset(); - m_codec = cAudioCodec::eInvalid; - avcodec_get_frame_defaults(m_frame); + m_reset = true; + m_wait->Signal(); m_mutex->Unlock(); } bool cAudioDecoder::Poll(void) { - return !m_parser->Size() && m_omx->PollAudioBuffers(); + return m_ready && m_omx->PollAudioBuffers(); } void cAudioDecoder::Action(void) @@ -583,214 +680,232 @@ void cAudioDecoder::Action(void) while (Running()) { - if (m_parser->Size()) + unsigned int channels = 0; + unsigned int outputChannels = 0; + + bool bufferFull = false; + + cAudioCodec::eCodec codec = cAudioCodec::eInvalid; + OMX_BUFFERHEADERTYPE *buf = 0; + + AVFrame *frame = avcodec_alloc_frame(); + if (!frame) { - m_mutex->Lock(); - if (m_outputFormatChanged) - { - m_outputFormatChanged = false; - m_omx->SetupAudioRender( - m_outputFormat, m_channels, - m_samplingRate, m_outputPort); - } + esyslog("rpihddevice: failed to allocate audio frame!"); + return; + } - OMX_BUFFERHEADERTYPE *buf = m_omx->GetAudioBuffer(m_pts); - if (buf) + m_reset = false; + m_ready = true; + + while (!m_reset) + { + // check for data if no decoded samples are pending + if (!m_parser->Empty() && !frame->nb_samples) { - while (DecodeFrame()) - buf->nFilledLen += ReadFrame( - buf->pBuffer + buf->nFilledLen, - buf->nAllocLen - buf->nFilledLen); + if (codec != m_parser->GetCodec() || + channels != m_parser->GetChannels()) + { + // to change codec config, we need to empty buffer first + if (buf) + bufferFull = true; + else + { + codec = m_parser->GetCodec(); + channels = m_parser->GetChannels(); - if (!m_omx->EmptyAudioBuffer(buf)) - esyslog("rpihddevice: failed to empty audio buffer!"); + outputChannels = channels; + SetCodec(codec, outputChannels); + } + } } - else + + // if codec has been configured but we don't have a buffer, get one + while (codec != cAudioCodec::eInvalid && !buf && !m_reset) { - esyslog("rpihddevice: failed to get audio buffer!"); - cCondWait::SleepMs(5); + buf = m_omx->GetAudioBuffer(m_pts); + if (buf) + m_pts = 0; + else + m_wait->Wait(10); } - m_mutex->Unlock(); - } - else - m_newData->Wait(50); - } - dsyslog("rpihddevice: cAudioDecoder() thread ended"); -} -unsigned int cAudioDecoder::DecodeFrame() -{ - unsigned int ret = 0; + // we have a non-full buffer and data to encode / copy + if (buf && !bufferFull && !m_parser->Empty()) + { + int copied = 0; + if (m_passthrough) + { + // for pass-through directly copy AV packet to buffer + if (m_parser->Packet()->size <= buf->nAllocLen - buf->nFilledLen) + { + m_mutex->Lock(); - if (m_passthrough) - ret = m_parser->Size(); + memcpy(buf->pBuffer + buf->nFilledLen, + m_parser->Packet()->data, m_parser->Packet()->size); + buf->nFilledLen += m_parser->Packet()->size; + m_parser->Shrink(m_parser->Packet()->size); - else if (m_parser->Size()) - { - int frame = 0; - int len = avcodec_decode_audio4(m_codecs[m_codec].context, - m_frame, &frame, m_parser->Packet()); + m_mutex->Unlock(); + } + else + if (m_parser->Packet()->size > buf->nAllocLen) + { + esyslog("rpihddevice: encoded audio frame too big!"); + m_reset = true; + break; + } + else + bufferFull = true; + } + else + { + // decode frame if we do not pass-through + m_mutex->Lock(); - // decoding error or number of channels changed ? - if (len < 0 || m_channels != m_codecs[m_codec].context->channels) - { - m_parser->Reset(); - m_codec = cAudioCodec::eInvalid; - } - else - { - m_parser->Shrink(len); - ret = frame ? len : 0; - } - } - return ret; -} + int gotFrame = 0; + int len = avcodec_decode_audio4(m_codecs[codec].context, + frame, &gotFrame, m_parser->Packet()); -unsigned int cAudioDecoder::ReadFrame(unsigned char *buf, unsigned int bufsize) -{ - unsigned int ret = 0; + if (len > 0) + m_parser->Shrink(len); - if (m_passthrough) - { - // for pass-through directly read from AV packet - if (m_parser->Size() > bufsize) - ret = bufsize; - else - ret = m_parser->Size(); + m_mutex->Unlock(); - memcpy(buf, m_parser->Data(), ret); - m_parser->Shrink(ret); - } - else - { - if (m_frame->nb_samples > 0) - { - ret = av_samples_get_buffer_size(NULL, - m_channels == 6 ? 8 : m_channels, m_frame->nb_samples, - m_codecs[m_codec].context->sample_fmt, 1); - - if (ret > bufsize) - { - esyslog("rpihddevice: decoded audio frame too big!"); - ret = 0; + if (len < 0) + { + esyslog("rpihddevice: failed to decode audio frame!"); + m_reset = true; + break; + } + } } - else + + // we have decoded samples we need to copy to buffer + if (buf && !bufferFull && frame->nb_samples > 0) { - if (m_channels == 6) - { - // interleaved copy to fit 5.1 data into 8 channels - int32_t* src = (int32_t*)m_frame->data[0]; - int32_t* dst = (int32_t*)buf; + int length = av_samples_get_buffer_size(NULL, + outputChannels == 6 ? 8 : outputChannels, frame->nb_samples, + m_codecs[codec].context->sample_fmt, 1); - for (int i = 0; i < m_frame->nb_samples; i++) + if (length <= buf->nAllocLen - buf->nFilledLen) + { + if (outputChannels == 6) { - *dst++ = *src++; // LF & RF - *dst++ = *src++; // CF & LFE - *dst++ = *src++; // LR & RR - *dst++ = 0; // empty channels + // interleaved copy to fit 5.1 data into 8 channels + int32_t* src = (int32_t*)frame->data[0]; + int32_t* dst = (int32_t*)buf->pBuffer + buf->nFilledLen; + + for (int i = 0; i < frame->nb_samples; i++) + { + *dst++ = *src++; // LF & RF + *dst++ = *src++; // CF & LFE + *dst++ = *src++; // LR & RR + *dst++ = 0; // empty channels + } } + else + memcpy(buf->pBuffer + buf->nFilledLen, frame->data[0], length); + + buf->nFilledLen += length; + frame->nb_samples = 0; } else - memcpy(buf, m_frame->data[0], ret); + { + if (length > buf->nAllocLen) + { + esyslog("rpihddevice: decoded audio frame too big!"); + m_reset = true; + break; + } + else + bufferFull = true; + } } - } - } - return ret; -} -bool cAudioDecoder::ProbeCodec(void) -{ - bool ret = false; - - unsigned int offset = 0; - cAudioCodec::eCodec codec = m_parser->Parse(offset); - - if (codec != cAudioCodec::eInvalid) - { - if (offset) - m_parser->Shrink(offset); - - // if new codec has been found, decode one packet to determine number of - // channels, since they are needed to properly set audio output format - if (codec != m_codec || cRpiSetup::HasAudioSetupChanged()) - { - m_codecs[codec].context->flags |= CODEC_FLAG_TRUNCATED; - m_codecs[codec].context->request_channel_layout = AV_CH_LAYOUT_NATIVE; - m_codecs[codec].context->request_channels = 0; - - int frame = 0; - avcodec_get_frame_defaults(m_frame); - int len = avcodec_decode_audio4(m_codecs[codec].context, m_frame, - &frame, m_parser->Packet()); + // check if no decoded samples are pending and parser is empty + if (!frame->nb_samples && m_parser->Empty()) + { + // if no more data but buffer with data -> end of PES packet + if (buf && buf->nFilledLen > 0) + bufferFull = true; + else + if (m_ready) + m_wait->Wait(50); + } - if (len > 0 && frame) + // we have a buffer to empty + if (buf && bufferFull) { - SetCodec(codec); - ret = true; + if (m_omx->EmptyAudioBuffer(buf)) + { + bufferFull = false; + buf = 0; + + // if parser is empty, get new data + if (m_parser->Empty()) + m_ready = true; + } + else + { + esyslog("rpihddevice: failed to empty audio buffer!"); + m_reset = true; + break; + } } } + + dsyslog("reset"); + if (buf && m_omx->EmptyAudioBuffer(buf)) + buf = 0; + + av_free(frame); + m_parser->Reset(); } - return ret; + dsyslog("rpihddevice: cAudioDecoder() thread ended"); } -void cAudioDecoder::SetCodec(cAudioCodec::eCodec codec) +void cAudioDecoder::SetCodec(cAudioCodec::eCodec codec, unsigned int &channels) { - if (codec != cAudioCodec::eInvalid) + if (codec != cAudioCodec::eInvalid && channels > 0) { - if (m_codec == cAudioCodec::eInvalid) - m_outputFormatChanged = true; + dsyslog("rpihddevice: set audio codec to %dch %s", + channels, cAudioCodec::Str(codec)); - m_codec = codec; - m_codecs[m_codec].context->request_channel_layout = AV_CH_LAYOUT_NATIVE; - m_codecs[m_codec].context->request_channels = 0; + m_codecs[codec].context->request_channel_layout = AV_CH_LAYOUT_NATIVE; + m_codecs[codec].context->request_channels = 0; m_passthrough = false; cAudioCodec::eCodec outputFormat = cAudioCodec::ePCM; cAudioPort::ePort outputPort = cAudioPort::eLocal; - int channels = m_codecs[m_codec].context->channels; - int samplingRate = m_codecs[m_codec].context->sample_rate; - - dsyslog("rpihddevice: set audio codec to %s with %d channels, %dHz", - cAudioCodec::Str(m_codec), channels, samplingRate); - if (cRpiSetup::GetAudioPort() == cAudioPort::eHDMI && - cRpiSetup::IsAudioFormatSupported(cAudioCodec::ePCM, - m_codecs[m_codec].context->channels, - m_codecs[m_codec].context->sample_rate)) + cRpiSetup::IsAudioFormatSupported(cAudioCodec::ePCM, channels, 48000)) { outputPort = cAudioPort::eHDMI; if (cRpiSetup::IsAudioPassthrough() && - cRpiSetup::IsAudioFormatSupported(m_codec, - m_codecs[m_codec].context->channels, - m_codecs[m_codec].context->sample_rate)) + cRpiSetup::IsAudioFormatSupported(codec, channels, 48000)) { m_passthrough = true; - outputFormat = m_codec; + outputFormat = codec; } } else { - m_codecs[m_codec].context->request_channel_layout = AV_CH_LAYOUT_STEREO_DOWNMIX; - m_codecs[m_codec].context->request_channels = 2; + m_codecs[codec].context->request_channel_layout = AV_CH_LAYOUT_STEREO_DOWNMIX; + m_codecs[codec].context->request_channels = 2; channels = 2; - } - if ((outputFormat != m_outputFormat) || - (outputPort != m_outputPort ) || - (channels != m_channels ) || - (samplingRate != m_samplingRate)) - { - m_outputFormat = outputFormat; - m_outputPort = outputPort; - m_channels = channels; - m_samplingRate = samplingRate; - m_outputFormatChanged = true; - - dsyslog("rpihddevice: set %s audio output format to %s%s", - cAudioPort::Str(m_outputPort), cAudioCodec::Str(m_outputFormat), - m_passthrough ? " (pass-through)" : ""); + // if 2ch PCM audio on HDMI is supported + if (cRpiSetup::GetAudioPort() == cAudioPort::eHDMI && + cRpiSetup::IsAudioFormatSupported(cAudioCodec::ePCM, 2, 48000)) + outputPort = cAudioPort::eHDMI; } + + m_omx->SetupAudioRender(outputFormat, channels, outputPort, 48000); + dsyslog("rpihddevice: set %s audio output format to %dch %s%s", + cAudioPort::Str(outputPort), channels, cAudioCodec::Str(outputFormat), + m_passthrough ? " (pass-through)" : ""); } } |