codec_audio.cpp

Go to the documentation of this file.

 
#include <cstdint>
#include <unistd.h>
 
extern "C" {
#include <libavcodec/avcodec.h>
}
 
#include "audio.h"
#include "codec_audio.h"
#include "logger.h"
#include "misc.h"
 
/*****************************************************************************
 * cAudioDecoder class
 ****************************************************************************/
 
cAudioDecoder::cAudioDecoder(cSoftHdAudio *audio)
    : m_pAudio(audio),
      m_passthroughMask(m_pAudio->GetPassthrough() & (CODEC_AC3 | CODEC_EAC3 | CODEC_DTS))
{
    if (!(m_pFrame = av_frame_alloc()))
        LOGFATAL("audiocodec: %s: can't allocate audio decoder frame buffer", __FUNCTION__);
 
    LOGDEBUG2(L_CODEC, "audiocodec: %s: Set passthrough mask %d", __FUNCTION__, m_passthroughMask);
}
 
cAudioDecoder::~cAudioDecoder(void)
{
    Close();
 
    av_frame_free(&m_pFrame);
}
 
void cAudioDecoder::Open(AVCodecID codecId, AVCodecParameters *par, AVRational timebase)
{
    const AVCodec *codec;
 
    m_codecId = codecId;
 
    // FIXME: errors shouldn't be fatal, maybe just disable audio
    if (codecId == AV_CODEC_ID_AC3) {
        if (!(codec = avcodec_find_decoder_by_name("ac3_fixed"))) {
            LOGFATAL("audiocodec: %s: codec ac3_fixed ID %#06x not found", __FUNCTION__, codecId);
        }
    } else if (codecId == AV_CODEC_ID_AAC) {
        if (!(codec = avcodec_find_decoder_by_name("aac_fixed"))) {
            LOGFATAL("audiocodec: %s: codec aac_fixed ID %#06x not found", __FUNCTION__, codecId);
        }
    } else {
        if (!(codec = avcodec_find_decoder(codecId))) {
            LOGFATAL("audiocodec: %s: codec %s ID %#06x not found", __FUNCTION__,
            avcodec_get_name(codecId), codecId);
        }
    }
 
    if (!(m_pAudioCtx = avcodec_alloc_context3(codec)))
        LOGFATAL("audiocodec: %s: can't allocate audio codec context", __FUNCTION__);
 
    m_pAudioCtx->pkt_timebase = timebase;
 
    if (par && ((avcodec_parameters_to_context(m_pAudioCtx, par)) < 0))
        LOGERROR("audiocodec: %s: insert parameters to context failed!", __FUNCTION__);
 
    if (avcodec_open2(m_pAudioCtx, m_pAudioCtx->codec, NULL) < 0)
        LOGFATAL("audiocodec: %s: can't open audio codec", __FUNCTION__);
 
    LOGDEBUG2(L_CODEC, "audiocodec: %s: Codec %s found, passthrough mask %d", __FUNCTION__, m_pAudioCtx->codec->long_name, m_passthroughMask);
 
    m_currentSampleRate = 0;
    m_currentHwSampleRate = 0;
    m_currentNumChannels = 0;
    m_currentHwNumChannels = 0;
    m_currentPassthrough = 0;
}
 
void cAudioDecoder::Close(void)
{
    LOGDEBUG2(L_CODEC, "audiocodec: %s", __FUNCTION__);
    if (m_pAudioCtx)
        avcodec_free_context(&m_pAudioCtx);
 
    m_codecId = AV_CODEC_ID_NONE;
    m_lastPts = AV_NOPTS_VALUE;
}
 
int cAudioDecoder::DecodePassthrough(const AVPacket * avpkt, AVFrame *frame)
{
    m_pAudio->SetTimebase(&m_pAudioCtx->pkt_timebase);
 
    // AC3 passthrough
    if (m_passthroughMask & CODEC_AC3 && m_pAudioCtx->codec_id == AV_CODEC_ID_AC3) {
        uint16_t *spdif = m_spdifOutput;
        int spdifSize = AC3_FRAME_SIZE * 4; // frames * channels * (samplesize / 8)
 
        if (spdifSize < avpkt->size + 8) {
            LOGERROR("audiocodec: %s: too much data for spdif buffer!", __FUNCTION__);
            return -1;
        }
 
        // build SPDIF header and append AC3 audio data to it
        int bitstreamMode = avpkt->data[5] & 0x07;
        spdif[0] = htole16(IEC61937_PREAMBLE1);
        spdif[1] = htole16(IEC61937_PREAMBLE2);
        spdif[2] = htole16(IEC61937_AC3 | bitstreamMode << 8);
        spdif[3] = htole16(avpkt->size * 8);
        // TODO: take endian into accout
        swab(avpkt->data, spdif + 4, avpkt->size);
        memset(spdif + 4 + avpkt->size / 2, 0, spdifSize - 8 - avpkt->size);
 
        m_pAudio->Enqueue(spdif, spdifSize, frame);
        return 1;
    }
 
    // EAC3 passthrough
    if (m_passthroughMask & CODEC_EAC3 && m_pAudioCtx->codec_id == AV_CODEC_ID_EAC3) {
        uint16_t *spdif = m_spdifOutput;
        int spdifSize = EAC3_FRAME_SIZE * 4; // frames * channels * (samplesize / 8)
        int repeat = 1;
 
        // spdifSize is smaller, if we don't have 192000
        if (m_currentHwSampleRate == 48000) {
            spdifSize /= 4;
        }
 
        if (spdifSize < m_spdifIndex + avpkt->size + 8) {
            LOGERROR("audiocodec: %s: too much data for spdif buffer!", __FUNCTION__);
            return -1;
        }
 
        // check if we need to pack multiple packets
        int fscod = (avpkt->data[4] >> 6) & 0x3;
        if (fscod != 0x3) {
            int fscod2 = (avpkt->data[4] >> 4) & 0x3;
            static const uint8_t eac3_repeat[4] = { 6, 3, 2, 1 };
            repeat = eac3_repeat[fscod2];
        }
 
//      LOGDEBUG2(L_CODEC, "audiocodec: %s: E-AC3: set repeat to %d (fscod = %d) avpkt->size %d (spdifSize %d)",
//          __FUNCTION__, repeat, fscod2, avpkt->size, spdifSize);
 
        // pack upto repeat EAC-3 pakets into one IEC 61937 burst
        // TODO: take endian into accout
        swab(avpkt->data, spdif + 4 + m_spdifIndex, avpkt->size);
        m_spdifIndex += avpkt->size;
 
        if (++m_spdifRepeatCount < repeat)
            return 1;
 
        // build SPDIF header and append E-AC3 audio data to it
        spdif[0] = htole16(IEC61937_PREAMBLE1);
        spdif[1] = htole16(IEC61937_PREAMBLE2);
        spdif[2] = htole16(IEC61937_EAC3);
        spdif[3] = htole16(m_spdifIndex * 8);
        memset(spdif + 4 + m_spdifIndex / 2, 0, spdifSize - 8 - m_spdifIndex);
 
        m_pAudio->Enqueue(spdif, spdifSize, frame);
        m_spdifIndex = 0;
        m_spdifRepeatCount = 0;
        return 1;
    }
 
    // DTS passthrough
    if (m_passthroughMask & CODEC_DTS && m_pAudioCtx->codec_id == AV_CODEC_ID_DTS) {
        uint16_t *spdif = m_spdifOutput;
 
        uint8_t nbs;
        int bsid;
        int burstSz;
 
        nbs = (uint8_t)((avpkt->data[4] & 0x01) << 6) |
                       ((avpkt->data[5] >> 2) & 0x3f);
        switch(nbs) {
        case 0x07:
            bsid = 0x0a;    // MPEG-2 layer 3 is used when?
            burstSz = 1024;
            break;
        case 0x0f:
            bsid = IEC61937_DTS1;
            burstSz = DTS1_FRAME_SIZE * 4; // frames * channels * (samplesize / 8)
            break;
        case 0x1f:
            bsid = IEC61937_DTS2;
            burstSz = DTS2_FRAME_SIZE * 4; // frames * channels * (samplesize / 8)
            break;
        case 0x3f:
            bsid = IEC61937_DTS3;
            burstSz = DTS3_FRAME_SIZE * 4; // frames * channels * (samplesize / 8)
            break;
        default:
            bsid = IEC61937_NULL;
            if (nbs < 5)
                nbs = 127;
            burstSz = (nbs + 1) * 32 * 2 + 2;
            break;
        }
 
        // build SPDIF header and append DTS audio data to it
        if (burstSz < avpkt->size + 8) {
            LOGERROR("audiocodec: %s: too much data for spdif buffer!", __FUNCTION__);
            return -1;
        }
        spdif[0] = htole16(IEC61937_PREAMBLE1);
        spdif[1] = htole16(IEC61937_PREAMBLE2);
        spdif[2] = htole16(bsid);
        spdif[3] = htole16(avpkt->size * 8);
        spdif[4] = htole16(DTS_PREAMBLE_16BE_1);
        spdif[5] = htole16(DTS_PREAMBLE_16BE_2);
        // TODO: take endian into accout
        swab(avpkt->data, spdif + 4, avpkt->size);
        memset(spdif + 4 + avpkt->size, 0, burstSz - 8 - avpkt->size);
 
        m_pAudio->Enqueue(spdif, burstSz, frame);
        return 1;
    }
 
    return 0;
}
 
int cAudioDecoder::UpdateFormat(void)
{
    int isPassthrough = 0;
    int err;
 
    LOGDEBUG2(L_SOUND, "audiocodec: %s: format change %s %dHz *%d channels%s%s%s%s%d", __FUNCTION__,
        av_get_sample_fmt_name(m_pAudioCtx->sample_fmt), m_pAudioCtx->sample_rate, m_pAudioCtx->ch_layout.nb_channels,
        m_passthroughMask & CODEC_AC3 ? " AC3" : "",
        m_passthroughMask & CODEC_EAC3 ? " EAC3" : "",
        m_passthroughMask & CODEC_DTS ? " DTS" : "",
        m_passthroughMask ? " passthrough mask " : "",
        m_passthroughMask ? m_passthroughMask : 0);
 
    m_currentSampleRate = m_pAudioCtx->sample_rate;
    m_currentHwSampleRate = m_pAudioCtx->sample_rate;
    m_currentNumChannels = m_pAudioCtx->ch_layout.nb_channels;
    m_currentHwNumChannels = m_pAudioCtx->ch_layout.nb_channels;
    m_currentPassthrough = m_passthroughMask;
 
    if ((m_currentPassthrough & CODEC_AC3  && m_pAudioCtx->codec_id == AV_CODEC_ID_AC3) ||
        (m_currentPassthrough & CODEC_EAC3 && m_pAudioCtx->codec_id == AV_CODEC_ID_EAC3) ||
        (m_currentPassthrough & CODEC_DTS  && m_pAudioCtx->codec_id == AV_CODEC_ID_DTS)) {
 
        // E-AC3 over HDMI: some receivers need HBR
        if (m_pAudioCtx->codec_id == AV_CODEC_ID_EAC3)
            m_currentHwSampleRate *= 4;
 
        m_currentHwNumChannels = 2;
        m_spdifIndex = 0;
        m_spdifRepeatCount = 0;
        isPassthrough = 1;
    }
 
    if ((err = m_pAudio->Setup(m_pAudioCtx, m_currentHwSampleRate, m_currentHwNumChannels, isPassthrough))) {
        // E-AC3 over HDMI: try without HBR
        m_currentHwSampleRate /= 4;
 
        if (m_pAudioCtx->codec_id != AV_CODEC_ID_EAC3 ||
            (err = m_pAudio->Setup(m_pAudioCtx, m_currentHwSampleRate, m_currentHwNumChannels, isPassthrough))) {
 
            m_currentHwSampleRate = 0;
            m_currentHwNumChannels = 0;
            LOGERROR("audiocodec: %s: format change update error", __FUNCTION__);
            return err;
        }
    }
    return 0;
}
 
void cAudioDecoder::Decode(const AVPacket * avpkt)
{
    int retSend, retRec;
    AVFrame *frame;
 
    // decoded frame is also needed for passthrough to set the PTS
    frame = m_pFrame;
    av_frame_unref(frame);
 
    do {
        retSend = avcodec_send_packet(m_pAudioCtx, avpkt);
        if (retSend < 0)
            LOGERROR("audiocodec: %s: avcodec_send_packet error: %s", __FUNCTION__, av_err2str(retSend));
 
        retRec = avcodec_receive_frame(m_pAudioCtx, frame);
 
        if (retRec < 0) {
            if (retRec != AVERROR(EAGAIN))
                LOGERROR("audiocodec: %s: avcodec_receive_frame error: %s", __FUNCTION__, av_err2str(retRec));
        } else {
            if (m_lastPts == AV_NOPTS_VALUE && avpkt->pts == AV_NOPTS_VALUE) {
                // the first AVPacket has no valid PTS, if its PES packet has been truncated while searching for the sync word
                av_frame_unref(frame);
                continue;
            }
 
            // update audio clock and remeber last PTS or guess the next PTS
            if (frame->pts != AV_NOPTS_VALUE) {
                m_lastPts = frame->pts;
            } else if (m_lastPts != AV_NOPTS_VALUE) {
                frame->pts = m_lastPts +
                    (int64_t)(frame->nb_samples / av_q2d(m_pAudioCtx->pkt_timebase) / frame->sample_rate);
                m_lastPts = frame->pts;
            }
 
            if (m_currentPassthrough != m_passthroughMask ||
                m_currentNumChannels != m_pAudioCtx->ch_layout.nb_channels ||
                m_currentSampleRate != m_pAudioCtx->sample_rate) {
                UpdateFormat();
            }
 
            if (!m_currentHwNumChannels || !m_currentHwSampleRate) {
                LOGERROR("audiocodec: %s: unsupported format!", __FUNCTION__);
                av_frame_unref(frame);
                return;
            }
 
            if (DecodePassthrough(avpkt, frame)) {
                av_frame_unref(frame);
                return;
            }
 
            m_pAudio->Filter(frame, m_pAudioCtx);
        }
 
    } while (retSend == AVERROR(EAGAIN));
}
 
void cAudioDecoder::FlushBuffers(void)
{
    LOGDEBUG2(L_CODEC, "audiocodec: %s", __FUNCTION__);
    if (m_pAudioCtx)
        avcodec_flush_buffers(m_pAudioCtx);
 
    m_lastPts = AV_NOPTS_VALUE;
    m_codecId = AV_CODEC_ID_NONE;
}
 
void cAudioDecoder::SetPassthrough(int mask)
{
    LOGDEBUG2(L_CODEC, "audiocodec: %s: %d", __FUNCTION__, mask);
    m_passthroughMask = mask & (CODEC_AC3 | CODEC_EAC3 | CODEC_DTS);
}