audio.cpp

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// SPDX-License-Identifier: AGPL-3.0-or-later
 
#include <chrono>
#include <cmath>
#include <cstdint>
#include <mutex>
#include <string>
#include <sstream>
#include <vector>
 
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavfilter/avfilter.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
}
 
#include <vdr/thread.h>
 
#include "alsadevice.h"
#include "audio.h"
#include "audioprocessor.h"
#include "codec_audio.h"
#include "config.h"
#include "filllevel.h"
#include "logger.h"
#include "misc.h"
#include "pidcontroller.h"
#include "ringbuffer.h"
#include "softhddevice.h"
#include "statemachine.h"
 
cSoftHdAudio::cSoftHdAudio(cSoftHdDevice *device)
    : cThread("softhd audio"),
      m_pDevice(device),
      m_pConfig(m_pDevice->Config()),
      m_alsa(m_pConfig),
      m_softVolume(m_pConfig->ConfigAudioSoftvol),
      m_audioProcessor(BYTES_PER_SAMPLE),
      m_pMixerChannel(m_pConfig->ConfigAudioMixerChannel)
{
    SetNormalize(m_pConfig->ConfigAudioNormalize, m_pConfig->ConfigAudioMaxNormalize);
    SetCompression(m_pConfig->ConfigAudioCompression, m_pConfig->ConfigAudioMaxCompression);
    SetStereoDescent(m_pConfig->ConfigAudioStereoDescent);
    SetEqualizer(m_pConfig->ConfigAudioEq, m_pConfig->ConfigAudioEqBand);
}
 
/******************************************************************************
 * Audio Filter
 *****************************************************************************/
 
static std::vector<std::string> GetFFmpegChannelLayoutAsArray(const AVChannelLayout &layout)
{
    std::vector<std::string> names;
    char buf[16];
 
    for (int i = 0; i < layout.nb_channels; i++) {
        enum AVChannel ch = av_channel_layout_channel_from_index(&layout, i);
        int ret = av_channel_name(buf, sizeof(buf), ch);
        if (ret < 0)
            continue;
        names.push_back(std::string(buf));
    }
    return names;
}
 
static bool LayoutsMatch(const std::vector<std::string> &ff, const std::vector<std::string> &alsa)
{
    if (ff.size() != alsa.size())
        return false;
 
    for (size_t i = 0; i < ff.size(); i++) {
        if (ff[i] != alsa[i])
            return false;
    }
 
    return true;
}
 
static bool LayoutIsValid(const std::vector<std::string> &channelLayout)
{
    return std::find(channelLayout.begin(), channelLayout.end(), "NA") == channelLayout.end();
}
 
std::string cSoftHdAudio::BuildChannelMapFilter(const AVChannelLayout &layout)
{
    auto ff = GetFFmpegChannelLayoutAsArray(layout);
    auto alsa = m_alsa.GetChannelLayoutAsArray();
 
    if (ff.size() != alsa.size()) {
        LOGWARNING("audio: %s: Skip channelmap filter, FFmpeg and Alsa channel count differs: FFmpeg %zu ALSA %zu", __FUNCTION__, ff.size(), alsa.size());
        return "";
    }
 
    std::string ffString;
    for (size_t i = 0; i < ff.size(); i++) {
        ffString += ff[i];
        if (i < ff.size() - 1)
            ffString += " ";
    }
 
    std::string alsaString;
    for (size_t i = 0; i < alsa.size(); i++) {
        alsaString += alsa[i];
        if (i < alsa.size() - 1)
            alsaString += " ";
    }
 
    if (!LayoutIsValid(alsa)) {
        LOGDEBUG2(L_SOUND, "audio: %s: Skip channelmap filter, alsa channel layout isn't valid: %s", __FUNCTION__, alsaString.c_str());
        return "";
    }
 
    if (LayoutsMatch(ff, alsa)) {
        LOGDEBUG2(L_SOUND, "audio: %s: Skip channelmap filter, FFmpeg and Alsa channel layouts match: %s", __FUNCTION__, ffString.c_str());
        return "";
    }
 
    std::stringstream ss;
    for (size_t i = 0; i < ff.size(); i++) {
        if (i != 0)
            ss << "|";
        ss << ff[i] << "-" << alsa[i];
    }
 
    LOGDEBUG2(L_SOUND, "audio: %s: FFmpeg Channel Layout: %s", __FUNCTION__, ffString.c_str());
    LOGDEBUG2(L_SOUND, "audio: %s: Alsa Channel Layout  : %s", __FUNCTION__, alsaString.c_str());
    LOGDEBUG2(L_SOUND, "audio: %s: Layouts don't match, map FFmpeg to Alsa: %s", __FUNCTION__, ss.str().c_str());
 
    return ss.str();
}
 
int cSoftHdAudio::InitFilter(AVCodecContext *audioCtx)
{
    const AVFilter  *abuffer;
    AVFilterContext *pFilterCtx[4];
    const AVFilter *channelmap;
    const AVFilter *eq;
    const AVFilter *aformat;
    const AVFilter *abuffersink;
    char channelLayout[64];
    char optionsStr[1024];
    int err, i, numFilter = 0;
 
    // Before filter init setup HW parameter
    err = Setup(audioCtx->pkt_timebase, audioCtx->sample_rate, audioCtx->ch_layout.nb_channels, false);
    if (err < 0) {
        LOGERROR("audio: %s: failed!", __FUNCTION__);
        return err;
    }
 
#if LIBAVFILTER_VERSION_INT < AV_VERSION_INT(7,16,100)
    avfilter_register_all();
#endif
 
    if (!(m_pFilterGraph = avfilter_graph_alloc())) {
        LOGERROR("audio: %s: Unable to create filter graph.", __FUNCTION__);
        return -1;
    }
 
    // input buffer
    if (!(abuffer = avfilter_get_by_name("abuffer"))) {
        LOGWARNING("audio: %s: Could not find the abuffer filter.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    if (!(m_pBuffersrcCtx = avfilter_graph_alloc_filter(m_pFilterGraph, abuffer, "src"))) {
        LOGWARNING("audio: %s: Could not allocate the m_pBuffersrcCtx instance.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
 
    av_channel_layout_describe(&audioCtx->ch_layout, channelLayout, sizeof(channelLayout));
 
    LOGDEBUG2(L_SOUND, "audio: %s: IN channelLayout %s sample_fmt %s sample_rate %d channels %d", __FUNCTION__,
              channelLayout, av_get_sample_fmt_name(audioCtx->sample_fmt), audioCtx->sample_rate, audioCtx->ch_layout.nb_channels);
 
    av_opt_set    (m_pBuffersrcCtx, "channel_layout", channelLayout,                                AV_OPT_SEARCH_CHILDREN);
    av_opt_set    (m_pBuffersrcCtx, "sample_fmt",     av_get_sample_fmt_name(audioCtx->sample_fmt), AV_OPT_SEARCH_CHILDREN);
    av_opt_set_q  (m_pBuffersrcCtx, "time_base",      (AVRational){ 1, audioCtx->sample_rate },     AV_OPT_SEARCH_CHILDREN);
    av_opt_set_int(m_pBuffersrcCtx, "sample_rate",    audioCtx->sample_rate,                        AV_OPT_SEARCH_CHILDREN);
//  av_opt_set_int(m_pBuffersrcCtx, "channel_counts", audioCtx->channels,                           AV_OPT_SEARCH_CHILDREN);
 
    // initialize the filter with NULL options, set all options above.
    if (avfilter_init_str(m_pBuffersrcCtx, NULL) < 0) {
        LOGWARNING("audio: %s: Could not initialize the abuffer filter.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
 
    // channelmap
    //
    // Map FFmpeg channel layout to Alsa channel layout.
    // Depending on the hardware, e.g. FC and LFE have to be swapped.
    // This is the case for HDMI on RPI4, so we need to do the following:
    //    FL-FL|FR-FR|FC-LFE|LFE-FC|BL-BL|BR-BR
    //
    // The channel mapping is skipped, if
    //   - a stereo downmix is forced (downmix will be done later in aformat filter)
    //   - channel count differs, aformat will handle downmix later
    if (!(m_alsa.GetDownmix() && m_alsa.GetHwNumChannels() == 2)) {
        std::string channelMapString;
        channelMapString = BuildChannelMapFilter(audioCtx->ch_layout);
 
        if (!channelMapString.empty()) {
            if (!(channelmap = avfilter_get_by_name("channelmap"))) {
                LOGWARNING("audio: %s: Could not find the channelmap filter.", __FUNCTION__);
                return -1;
            }
            if (!(pFilterCtx[numFilter] = avfilter_graph_alloc_filter(m_pFilterGraph, channelmap, "channelmap"))) {
                LOGWARNING("audio: %s: Could not allocate the channelmap instance.", __FUNCTION__);
                return -1;
            }
            snprintf(optionsStr, sizeof(optionsStr),"map=%s", channelMapString.c_str());
            if (avfilter_init_str(pFilterCtx[numFilter], optionsStr) < 0) {
                LOGWARNING("audio: %s: Could not initialize the channelmap filter \"%s\"", __FUNCTION__, optionsStr);
                avfilter_graph_free(&m_pFilterGraph);
                return -1;
            }
            numFilter++;
        }
    }
 
    // superequalizer
    if (m_useEqualizer) {
        if (!(eq = avfilter_get_by_name("superequalizer"))) {
            LOGWARNING("audio: %s: Could not find the superequalizer filter.", __FUNCTION__);
            avfilter_graph_free(&m_pFilterGraph);
            return -1;
        }
        if (!(pFilterCtx[numFilter] = avfilter_graph_alloc_filter(m_pFilterGraph, eq, "superequalizer"))) {
            LOGWARNING("audio: %s: Could not allocate the superequalizer instance.", __FUNCTION__);
            avfilter_graph_free(&m_pFilterGraph);
            return -1;
        }
 
        std::string equalizerOptions = m_audioProcessor.GetEqualizerOptions();
        snprintf(optionsStr, sizeof(optionsStr), "%s", equalizerOptions.c_str());
 
        if (avfilter_init_str(pFilterCtx[numFilter], optionsStr) < 0) {
            LOGWARNING("audio: %s: Could not initialize the superequalizer filter.", __FUNCTION__);
            avfilter_graph_free(&m_pFilterGraph);
            return -1;
        }
        numFilter++;
    }
 
    // aformat
    AVChannelLayout channel_layout;
    if (m_alsa.GetDownmix() && m_alsa.GetHwNumChannels() == 2) {
        // explicit stereo downmix
        av_channel_layout_default(&channel_layout, 2);
    } else {
        if (av_channel_layout_copy(&channel_layout, &audioCtx->ch_layout) < 0) {
            LOGWARNING("audio: %s: Could not copy channel layout", __FUNCTION__);
            return -1;
        }
 
        // clamp channels if the hardware doesn't support them
        if (channel_layout.nb_channels > (int)m_alsa.GetHwNumChannels()) {
            LOGDEBUG2(L_SOUND, "audio: %s: clamp channels from %d -> %d", __FUNCTION__, channel_layout.nb_channels, m_alsa.GetHwNumChannels());
            av_channel_layout_uninit(&channel_layout);
            av_channel_layout_default(&channel_layout, m_alsa.GetHwNumChannels());
        }
    }
    av_channel_layout_describe(&channel_layout, channelLayout, sizeof(channelLayout));
    av_channel_layout_uninit(&channel_layout);
 
    LOGDEBUG2(L_SOUND, "audio: %s: OUT downmix %d hwNumChannels %d hwSampleRate %d channelLayout %s bytes_per_sample %d",
              __FUNCTION__, m_alsa.GetDownmix(), m_alsa.GetHwNumChannels(), m_alsa.GetHwSampleRate(), channelLayout, av_get_bytes_per_sample(AV_SAMPLE_FMT_S16));
 
    if (!(aformat = avfilter_get_by_name("aformat"))) {
        LOGWARNING("audio: %s: Could not find the aformat filter.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    if (!(pFilterCtx[numFilter] = avfilter_graph_alloc_filter(m_pFilterGraph, aformat, "aformat"))) {
        LOGWARNING("audio: %s: Could not allocate the aformat instance.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    snprintf(optionsStr, sizeof(optionsStr),
        "sample_fmts=%s:sample_rates=%d:channel_layouts=%s",
        av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), m_alsa.GetHwSampleRate(), channelLayout);
    if (avfilter_init_str(pFilterCtx[numFilter], optionsStr) < 0) {
        LOGWARNING("audio: %s: Could not initialize the aformat filter.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    numFilter++;
 
    // abuffersink
    if (!(abuffersink = avfilter_get_by_name("abuffersink"))) {
        LOGWARNING("audio: %s: Could not find the abuffersink filter.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    if (!(pFilterCtx[numFilter] = avfilter_graph_alloc_filter(m_pFilterGraph, abuffersink, "sink"))) {
        LOGWARNING("audio: %s: Could not allocate the abuffersink instance.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    if (avfilter_init_str(pFilterCtx[numFilter], NULL) < 0) {
        LOGWARNING("audio: %s: Could not initialize the abuffersink instance.", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
    numFilter++;
 
    // Connect the filters
    for (i = 0; i < numFilter; i++) {
        if (i == 0) {
            err = avfilter_link(m_pBuffersrcCtx, 0, pFilterCtx[i], 0);
        } else {
            err = avfilter_link(pFilterCtx[i - 1], 0, pFilterCtx[i], 0);
        }
    }
    if (err < 0) {
        LOGWARNING("audio: %s: Error connecting audio filters", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
 
    // Configure the graph.
    if (avfilter_graph_config(m_pFilterGraph, NULL) < 0) {
        LOGWARNING("audio: %s: Error configuring the audio filter graph", __FUNCTION__);
        avfilter_graph_free(&m_pFilterGraph);
        return -1;
    }
 
    m_pBuffersinkCtx = pFilterCtx[numFilter - 1];
    m_filterChanged = 0;
    m_filterReady = 1;
 
    return 0;
}
 
/******************************************************************************
 * Audio stream handling
 *****************************************************************************/
 
void cSoftHdAudio::DropSamplesOlderThanPtsMs(int64_t ptsMs)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    if (!HasInputPts())
        return;
 
    int64_t dropMs = std::max((int64_t)0, ptsMs - GetOutputPtsMsInternal());
    int dropFrames = m_alsa.MsToFrames(dropMs);
    int dropBytes = m_alsa.FramesToBytes(dropFrames);
 
    dropBytes = std::min(dropBytes, (int)m_pRingbuffer.UsedBytes());
 
    if (dropBytes > 0) {
        LOGDEBUG2(L_AV_SYNC, "audio: %s: dropping %dms audio samples to start in sync with the video (output PTS %s -> %s)",
            __FUNCTION__,
            dropMs,
            Timestamp2String(GetOutputPtsMsInternal(), 1),
            Timestamp2String(ptsMs, 1));
 
        m_pidController.Reset();
        m_fillLevel.Reset();
        m_fillLevel.WroteFrames(dropFrames);
        m_pRingbuffer.ReadAdvance(dropBytes);
    }
}
 
void cSoftHdAudio::EnqueueFrame(AVFrame *frame)
{
    if (!frame)
        return;
 
    uint16_t *buffer;
 
    int byteCount = frame->nb_samples * frame->ch_layout.nb_channels * BYTES_PER_SAMPLE;
    buffer = (uint16_t *)frame->data[0];
 
    if (m_useCompressor)       // in place operation
        m_audioProcessor.Compress(buffer, byteCount);
 
    if (m_useNormalizer)       // in place operation
        m_audioProcessor.Normalize(buffer, byteCount);
 
    Enqueue((uint16_t *)buffer, byteCount, frame->pts);
 
    av_frame_free(&frame);
}
 
void cSoftHdAudio::RebuildPauseBurst(int size)
{
    if (size == m_spdifBurstSize)
        return;
 
    LOGDEBUG2(L_SOUND, "audio: %s: spdif burst size changed %d -> %d, rebuild pause burst", __FUNCTION__, m_spdifBurstSize, size);
 
    m_spdifBurstSize = size;
    m_pauseBurst.resize(size / 2);
    uint16_t *spdif = m_pauseBurst.data();
 
    constexpr int IEC61937_PREAMBLE1 = 0xF872;
    constexpr int IEC61937_PREAMBLE2 = 0x4E1F;
    constexpr int IEC61937_NULL = 0x00;
 
    spdif[0] = htole16(IEC61937_PREAMBLE1);
    spdif[1] = htole16(IEC61937_PREAMBLE2);
    spdif[2] = htole16(IEC61937_NULL);
    spdif[3] = 0;
 
    memset(m_pauseBurst.data() + 4, 0, m_spdifBurstSize - 8);
}
 
void cSoftHdAudio::EnqueueSpdif(const uint16_t *buffer, int count, int64_t pts)
{
    std::lock_guard<std::mutex> lock(m_pauseMutex);
 
    RebuildPauseBurst(count);
 
    Enqueue(buffer, count, pts);
}
 
void cSoftHdAudio::Enqueue(const uint16_t *buffer, int count, int64_t pts)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    // pitch adjustment
    if (!m_alsa.IsPassthroughActive() && m_pitchAdjustFrameCounter == 0 && std::abs(m_pitchPpm) > 1) { // only adjust if pitch has a significant value to prevent overly large values/division by zero
        int oneFrameBytes = m_alsa.FramesToBytes(1);
 
        if (m_pitchPpm < 0 && m_pRingbuffer.Write((const uint16_t *)buffer, oneFrameBytes)) // insert additional frame
            m_fillLevel.ReceivedFrames(1);
        else if (m_pitchPpm > 0) // drop frame
            count = std::max(0, count - oneFrameBytes);
 
        m_pitchAdjustFrameCounter = std::round(1'000'000.0 / std::abs(m_pitchPpm));
    }
 
    m_pitchAdjustFrameCounter = std::max(0, m_pitchAdjustFrameCounter - m_alsa.BytesToFrames(count));
 
    // write to ringbuffer
    int bytesWritten = m_pRingbuffer.Write((const uint16_t *)buffer, count);
    if (bytesWritten != count)
        LOGERROR("audio: %s: can't place %d samples in ring buffer", __FUNCTION__, count);
 
    m_fillLevel.ReceivedFrames(m_alsa.BytesToFrames(bytesWritten));
 
    if (pts != AV_NOPTS_VALUE) {
        // Discontinuity check:
        // - force a resync if the new pts is more than AV_SYNC_BORDER_MS greater than the last
        // - a PTS wrap is not recognized, because of the ">"
        // - not sure, if a forward SkipSeconds() could trigger this, but then the resync is skipped in the video thread
        if (m_inputPts != AV_NOPTS_VALUE && (m_alsa.PtsToMs(pts, av_q2d(m_timebase)) - m_alsa.PtsToMs(m_inputPts, av_q2d(m_timebase))) > AV_SYNC_BORDER_MS) {
            LOGDEBUG2(L_AV_SYNC, "audio: %s: discontinuity detected in audio PTS %s -> %s", __FUNCTION__,
                Timestamp2String(m_alsa.PtsToMs(m_inputPts, av_q2d(m_timebase)), 1), Timestamp2String(m_alsa.PtsToMs(pts, av_q2d(m_timebase)), 1));
            std::lock_guard<std::mutex> lock(m_queueMutex);
            m_eventQueue.push_back(ScheduleResyncAtPtsMsEvent{m_alsa.PtsToMs(pts, av_q2d(m_timebase))});
        }
 
        m_inputPts = pts;
    } else if (m_inputPts != AV_NOPTS_VALUE) {
        m_inputPts += m_alsa.FramesToPts(m_alsa.BytesToFrames(count), av_q2d(m_timebase));
    }
}
 
int cSoftHdAudio::Setup(AVRational timebase, int samplerate, int channels, bool passthrough)
{
    int err = 0;
 
    m_timebase = timebase;
 
    // skip setup, nothing changed
    if (samplerate == (int)m_alsa.GetHwSampleRate() &&
        passthrough == m_alsa.IsPassthroughActive() &&
       (channels == (int)m_alsa.GetHwNumChannels() || (m_alsa.GetDownmix() && m_alsa.GetHwNumChannels() == 2)))
        return 1;
 
    if (Active()) {
        Stop();
        DropAlsaBuffers();
    }
 
    err = m_alsa.Setup(channels, samplerate, passthrough, m_pConfig->ConfigAudioDownmix);
    if (err)
        LOGERROR("audio: %s: failed!", __FUNCTION__);
    else
        Start();
 
    return err;
}
 
AVFrame *cSoftHdAudio::FilterGetFrame(void)
{
    AVFrame *outframe = nullptr;
    outframe = av_frame_alloc();
    if (!outframe) {
        LOGERROR("audio: %s: Error allocating frame", __FUNCTION__);
        return NULL;
    }
 
    int err = av_buffersink_get_frame(m_pBuffersinkCtx, outframe);
 
    if (err == AVERROR(EAGAIN)) {
//      LOGERROR("audio: %s: Error filtering AVERROR(EAGAIN)", __FUNCTION__);
        av_frame_free(&outframe);
    } else if (err == AVERROR_EOF) {
        LOGERROR("audio: %s: Error filtering AVERROR_EOF", __FUNCTION__);
        av_frame_free(&outframe);
    } else if (err < 0) {
        LOGERROR("audio: %s: Error filtering the data", __FUNCTION__);
        av_frame_free(&outframe);
    }
 
    return outframe;
}
 
int cSoftHdAudio::CheckForFilterReady(AVCodecContext *ctx)
{
    if (m_filterReady && m_filterChanged) {
//      LOGDEBUG2(L_SOUND, "audio: %s: m_filterReady %d sink_links_count %d channels %d nb_filters %d nb_outputs %d channels %d m_filterChanged %d",
//          __FUNCTION__, m_filterReady,
//          m_pFilterGraph->sink_links_count, m_pFilterGraph->sink_links[0]->channels,
//          m_pFilterGraph->filters[m_pFilterGraph->nb_filters - 1]->nb_outputs,
//          m_pFilterGraph->nb_filters, m_pFilterGraph->filters[m_pFilterGraph->nb_filters - 1]->outputs[m_pFilterGraph->filters[m_pFilterGraph->nb_filters - 1]->nb_outputs - 1]->channels,
//          m_filterChanged);
        avfilter_graph_free(&m_pFilterGraph);
        m_filterReady = 0;
        LOGDEBUG2(L_SOUND, "audio: %s: Free the filter graph.", __FUNCTION__);
    }
 
    if (!m_filterReady) {
        if (InitFilter(ctx)) {
            LOGDEBUG2(L_SOUND, "audio: %s: AudioFilterReady failed!", __FUNCTION__);
            return 1;
        }
    }
 
    return 0;
}
 
void cSoftHdAudio::Filter(AVFrame *inframe, AVCodecContext *ctx)
{
    AVFrame *outframe = NULL;
    int err = -1;
    int err_count = 0;
 
    if (inframe) {
        while (err < 0) {
            if (CheckForFilterReady(ctx)) {
                av_frame_unref(inframe);
                return;
            }
 
            err = av_buffersrc_add_frame(m_pBuffersrcCtx, inframe);
            if (err < 0) {
                if (err_count) {
                    char errbuf[128];
                    av_strerror(err, errbuf, sizeof(errbuf));
                    LOGERROR("audio: %s: Error submitting the frame to the filter fmt %s channels %d %s", __FUNCTION__,
                        av_get_sample_fmt_name(ctx->sample_fmt), ctx->ch_layout.nb_channels, errbuf);
                    av_frame_unref(inframe);
                    return;
                } else {
                    m_filterChanged = 1;
                    err_count++;
                    LOGDEBUG2(L_SOUND, "audio: %s: m_filterChanged %d  err_count %d", __FUNCTION__, m_filterChanged, err_count);
                }
            }
        }
    }
 
//  if (!inframe)
//      LOGDEBUG2(L_SOUND, "audio: %s: NO inframe!", __FUNCTION__);
 
    outframe = FilterGetFrame();
    EnqueueFrame(outframe);
}
 
void cSoftHdAudio::FlushBuffers(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    LOGDEBUG2(L_SOUND, "audio: %s", __FUNCTION__);
 
    if (!m_initialized)
        return;
 
    if (m_inputPts != AV_NOPTS_VALUE)
        FlushAlsaBuffers();
 
    m_fillLevel.Reset();
    m_fillLevel.ResetFramesCounters();
    m_pidController.Reset();
    m_pRingbuffer.Reset();
    m_inputPts = AV_NOPTS_VALUE;
    m_filterChanged = 1;
}
 
int cSoftHdAudio::GetUsedRingbufferBytes(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    return m_pRingbuffer.UsedBytes();
}
 
int cSoftHdAudio::GetUsedRingbufferMs(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    return m_alsa.FramesToMs(m_alsa.BytesToFrames(m_pRingbuffer.UsedBytes()));
}
 
int64_t cSoftHdAudio::GetOutputPtsMs(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    return GetOutputPtsMsInternal();
}
 
int64_t cSoftHdAudio::GetOutputPtsMsInternal(void)
{
    if (m_inputPts == AV_NOPTS_VALUE)
        return AV_NOPTS_VALUE;
 
    int64_t ptsMs = m_alsa.PtsToMs(m_inputPts, av_q2d(m_timebase)) - m_alsa.FramesToMs(m_alsa.BytesToFrames(m_pRingbuffer.UsedBytes()));
 
    // handle a PTS wrap
    if (ptsMs < 0)
        ptsMs += m_alsa.PtsToMs(PTS_WRAP, av_q2d(m_timebase));
 
    return ptsMs;
}
 
int64_t cSoftHdAudio::GetHardwareOutputPtsMs(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    if (!m_alsa.IsRunning() || m_inputPts == AV_NOPTS_VALUE)
        return AV_NOPTS_VALUE;
 
    int delayFrames = m_alsa.GetHwDelayFrames();
 
    // subtract baseline to ignore pause bursts already in the buffer
    delayFrames -= m_hwBaseline;
 
    int64_t ptsMs = GetOutputPtsMsInternal() - m_alsa.FramesToMs(delayFrames);
 
    // handle a PTS wrap
    if (ptsMs < 0)
        ptsMs += m_alsa.PtsToMs(PTS_WRAP, av_q2d(m_timebase));
 
    return ptsMs;
}
 
int64_t cSoftHdAudio::GetHardwareOutputDelayMs(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    if (!m_alsa.IsRunning() || m_inputPts == AV_NOPTS_VALUE)
        return AV_NOPTS_VALUE;
 
    int delayFrames = m_alsa.GetHwDelayFrames();
 
    return m_alsa.FramesToMs(delayFrames);
}
 
int64_t cSoftHdAudio::GetHardwareOutputPtsTimebaseUnits(void)
{
    int64_t ptsMs = GetHardwareOutputPtsMs();
    if (ptsMs == AV_NOPTS_VALUE)
        return AV_NOPTS_VALUE;
 
    return m_alsa.MsToPts(ptsMs, av_q2d(m_timebase));
}
 
void cSoftHdAudio::SetVolume(int volume)
{
    m_volume = volume;
    // reduce loudness for stereo output
    if (m_stereoDescent && m_alsa.GetHwNumChannels() == 2 && !m_alsa.IsPassthroughActive()) {
        volume -= m_stereoDescent;
        if (volume < 0)
            volume = 0;
        else if (volume > 1000)
            volume = 1000;
    }
 
    m_audioProcessor.SetAmplifier(volume);
    if (!m_softVolume)
        m_alsa.SetVolume(volume);
}
 
void cSoftHdAudio::SetPaused(bool pause)
{
    std::lock_guard<std::mutex> lock(m_pauseMutex);
    LOGDEBUG2(L_SOUND, "audio: %s: %d", __FUNCTION__, pause);
 
    m_paused = pause;
}
 
void cSoftHdAudio::SetNormalize(bool enable, int maxfac)
{
    m_useNormalizer = enable;
    m_audioProcessor.SetNormalizer(maxfac);
}
 
void cSoftHdAudio::SetCompression(bool enable, int maxfac)
{
    m_useCompressor = enable;
    m_audioProcessor.SetCompressor(maxfac);
}
 
void cSoftHdAudio::SetEqualizer(bool enable, int band[18])
{
    m_filterChanged = 1;
    m_useEqualizer = enable;
    m_audioProcessor.SetEqualizer(band);
}
 
void cSoftHdAudio::SetStereoDescent(int delta)
{
    m_stereoDescent = delta;
    SetVolume(m_volume);    // update channel delta
}
 
int cSoftHdAudio::LazyInit()
{
    if (m_initialized)
        return 0;
 
    if (!m_alsa.Init()) {
        LOGERROR("audio: could not initialize alsa");
        return -1;
    }
 
    m_initialized = true;
 
    return 0;
}
 
void cSoftHdAudio::Exit(void)
{
    LOGDEBUG2(L_SOUND, "audio: %s", __FUNCTION__);
 
    Stop();
 
    if (!m_initialized)
        return;
 
    avfilter_graph_free(&m_pFilterGraph);
    m_alsa.Exit();
    m_initialized = false;
}
 
void cSoftHdAudio::FlushAlsaBuffers(void)
{
    m_alsa.FlushBuffers(false);
 
    m_audioProcessor.ResetCompressor();
    m_audioProcessor.ResetNormalizer();
}
 
void cSoftHdAudio::DropAlsaBuffers(void)
{
    m_alsa.FlushBuffers(true);
 
    m_audioProcessor.ResetCompressor();
    m_audioProcessor.ResetNormalizer();
}
 
/******************************************************************************
 * Thread playback
 *****************************************************************************/
 
void cSoftHdAudio::Action(void)
{
    LOGDEBUG("audio: thread started");
    while (Running()) {
        bool scheduleImmediately = CyclicCall();
        ProcessEvents();
 
        if (scheduleImmediately)
            usleep(1000);
        else
            usleep(10000);
    }
    LOGDEBUG("audio: thread stopped");
}
 
void cSoftHdAudio::Stop(void)
{
    if (!Active())
        return;
 
    LOGDEBUG("audio: stopping thread");
    Cancel(2);
}
 
bool cSoftHdAudio::CyclicCall(void)
{
    std::lock_guard<std::mutex> lock1(m_pauseMutex);
 
    // do nothing in paused PCM mode
    if (m_paused && !m_alsa.IsPassthroughActive())
        return false;
 
    int err = m_alsa.WaitUntilReady();
    if (err < 0) {
        if (m_alsa.HandleError(err)) {
            std::lock_guard<std::mutex> lock(m_queueMutex);
            if (!m_pDevice->IsDraining())
                m_eventQueue.push_back(BufferUnderrunEvent{AUDIO});
        }
        return false;
    } else if (err == 0) {
        return true;
    }
 
    std::lock_guard<std::mutex> lock2(m_mutex);
 
    int freeAlsaBufferFrames = m_alsa.GetAvailableBufferFrames(false);
    if (freeAlsaBufferFrames == -EAGAIN)
        return true; // ?? is this correct?
    else if (freeAlsaBufferFrames < 0) {
        if (m_alsa.HandleError(freeAlsaBufferFrames)) {
            std::lock_guard<std::mutex> lock(m_queueMutex);
            m_eventQueue.push_back(BufferUnderrunEvent{AUDIO});
        }
        return false;
    }
 
    if (m_alsa.IsPassthroughActive() && m_paused) {
        // only write, if there is space for a full pause burst
        size_t freeAlsaBufferBytes = m_alsa.FramesToBytes(freeAlsaBufferFrames);
        if ((int)freeAlsaBufferBytes < m_spdifBurstSize)
            return false;
 
        // send a pause burst to keep the audio stream locked
        return SendPause();
    }
 
    return SendAudio(freeAlsaBufferFrames);
}
 
 
bool cSoftHdAudio::SendAudio(int freeAlsaBufferFrames)
{
    int bytesToWrite;
    int freeAlsaBufferBytes = m_alsa.FramesToBytes(freeAlsaBufferFrames);
 
    // query ringbuffer fill level
    const void *data;
    ssize_t ringBufferFillLevelBytes = m_pRingbuffer.GetReadPointer(&data);
 
    bytesToWrite = std::min(freeAlsaBufferBytes, (int)ringBufferFillLevelBytes);
 
    if (bytesToWrite == 0)
        return false;
 
    // muting pass-through AC-3, can produce disturbance
    if (m_volume == 0 || (m_softVolume && !m_alsa.IsPassthroughActive())) {
        // FIXME: quick&dirty cast
        m_audioProcessor.Amplify((int16_t *) data, bytesToWrite, m_volume);
        // FIXME: if not all are written, we double amplify them
    }
 
    int framesToWrite = m_alsa.BytesToFrames(bytesToWrite);
    int framesWritten = m_alsa.Write(data, framesToWrite);
    m_fillLevel.WroteFrames(framesWritten);
    m_pRingbuffer.ReadAdvance(m_alsa.FramesToBytes(framesWritten));
 
    return m_alsa.CheckWrittenFrames(framesWritten, framesToWrite);
}
 
bool cSoftHdAudio::SendPause(void)
{
    int framesToWrite = m_alsa.BytesToFrames(m_spdifBurstSize);
    int framesWritten = m_alsa.Write(m_pauseBurst.data(), framesToWrite);
 
    return m_alsa.CheckWrittenFrames(framesWritten, framesToWrite);
}
 
 
void cSoftHdAudio::SetHwDelayBaseline(void)
{
    if (!m_firstRealAudioReceived) {
        m_hwBaseline = 0;
 
        if (!m_alsa.IsPassthroughActive())
            return;
 
        m_hwBaseline = m_alsa.GetHwDelayFrames();
 
        LOGDEBUG2(L_SOUND, "audio: %s: first real audio was sent, hwBaseline %ld frames (%dms)", __FUNCTION__, m_hwBaseline, m_alsa.FramesToMs(m_hwBaseline));
        m_firstRealAudioReceived = true;
    }
}
 
void cSoftHdAudio::ResetHwDelayBaseline(void)
{
    std::lock_guard<std::mutex> lock(m_mutex);
 
    LOGDEBUG2(L_SOUND, "audio: %s: reset hw delay baseline to 0", __FUNCTION__);
    m_hwBaseline = 0;
    m_firstRealAudioReceived = false;
}
 
void cSoftHdAudio::ProcessEvents(void)
{
    std::lock_guard<std::mutex> lock(m_queueMutex);
    for (Event event : m_eventQueue)
        m_pDevice->TriggerEvent(event);
 
    m_eventQueue.clear();
}
 
void cSoftHdAudio::ClockDriftCompensation(void)
{
    if (m_alsa.IsPassthroughActive())
        return;
 
    double bufferFillLevelMs = m_alsa.FramesToMsDouble(m_fillLevel.GetBufferFillLevelFramesAvg());
    if (m_fillLevel.IsSettled()) {
        auto now = std::chrono::steady_clock::now();
        std::chrono::duration<double> elapsedSec = now - m_lastPidInvocation;
        m_lastPidInvocation = now;
 
        m_pitchPpm = m_pidController.Update(bufferFillLevelMs, elapsedSec.count()) * -1;
    } else
        m_pidController.SetTargetValue(bufferFillLevelMs);
 
    if (m_packetCounter++ % 1000 == 0) {
        LOGDEBUG2(L_SOUND, "audio: %s: buffer fill level: %.1fms (target: %.1fms), clock drift compensating pitch: %.1fppm, PID controller: P=%.2fppm I=%.2fppm D=%.2fppm",
            __FUNCTION__,
            bufferFillLevelMs,
            m_pidController.GetTargetValue(),
            m_pitchPpm.load(),
            m_pidController.GetPTerm(),
            m_pidController.GetITerm(),
            m_pidController.GetDTerm());
    }
 
    // buffer fill level low pass filter
    int availableFrames = m_alsa.GetAvailableBufferFrames(true);
 
    if (availableFrames >= 0)
        m_fillLevel.UpdateAvgBufferFillLevel(m_alsa.GetBufferSizeFrames() - availableFrames);
}