audiooutpututil.cpp

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#include "audiooutpututil.h"
 
#include <cstdint>
#include <limits> // workaround QTBUG-90395
 
#include <QtGlobal>
#include <QtEndian>
 
#include "libmythbase/mythlogging.h"
 
#include "audioconvert.h"
#include "mythaverror.h"
#include "mythavframe.h"
 
extern "C" {
#include "libavcodec/avcodec.h"
}
#include "pink.h"
 
#define LOC QString("AOUtil: ")
 
#ifdef Q_PROCESSOR_X86
// Check cpuid for SSE2 support on x86 / x86_64
static inline bool sse2_check()
{
#ifdef Q_PROCESSOR_X86_64
    return true;
#else
    static int has_sse2 = -1;
    if (has_sse2 != -1)
        return (bool)has_sse2;
    __asm__(
        // -fPIC - we may not clobber ebx/rbx
        "push       %%ebx               \n\t"
        "mov        $1, %%eax           \n\t"
        "cpuid                          \n\t"
        "and        $0x4000000, %%edx   \n\t"
        "shr        $26, %%edx          \n\t"
        "pop        %%ebx               \n\t"
        :"=d"(has_sse2)
        ::"%eax","%ecx"
    );
    return (bool)has_sse2;
#endif
}
#endif //Q_PROCESSOR_X86
 
bool AudioOutputUtil::has_optimized_SIMD()
{
#ifdef Q_PROCESSOR_X86
    return sse2_check();
#else
    return false;
#endif
}
 
int AudioOutputUtil::toFloat(AudioFormat format, void *out, const void *in,
                             int bytes)
{
    return AudioConvert::toFloat(format, out, in, bytes);
}
 
int AudioOutputUtil::fromFloat(AudioFormat format, void *out, const void *in,
                               int bytes)
{
    return AudioConvert::fromFloat(format, out, in, bytes);
}
 
void AudioOutputUtil::MonoToStereo(void *dst, const void *src, int samples)
{
    AudioConvert::MonoToStereo(dst, src, samples);
}
 
void AudioOutputUtil::AdjustVolume(void *buf, int len, int volume,
                                   bool music, bool upmix)
{
    float g     = volume / 100.0F;
    auto *fptr  = (float *)buf;
    int samples = len >> 2;
    int i       = 0;
 
    // Should be exponential - this'll do
    g *= g;
 
    // Try to ~ match stereo volume when upmixing
    if (upmix)
        g *= 1.5F;
 
    // Music is relatively loud
    if (music)
        g *= 0.4F;
 
    if (g == 1.0F)
        return;
 
#ifdef Q_PROCESSOR_X86
    if (sse2_check() && samples >= 16)
    {
        int loops = samples >> 4;
        i = loops << 4;
 
        __asm__ volatile (
            "movss      %2, %%xmm0          \n\t"
            "punpckldq  %%xmm0, %%xmm0      \n\t"
            "punpckldq  %%xmm0, %%xmm0      \n\t"
            "1:                             \n\t"
            "movups     (%0), %%xmm1        \n\t"
            "movups     16(%0), %%xmm2      \n\t"
            "mulps      %%xmm0, %%xmm1      \n\t"
            "movups     32(%0), %%xmm3      \n\t"
            "mulps      %%xmm0, %%xmm2      \n\t"
            "movups     48(%0), %%xmm4      \n\t"
            "mulps      %%xmm0, %%xmm3      \n\t"
            "movups     %%xmm1, (%0)        \n\t"
            "mulps      %%xmm0, %%xmm4      \n\t"
            "movups     %%xmm2, 16(%0)      \n\t"
            "movups     %%xmm3, 32(%0)      \n\t"
            "movups     %%xmm4, 48(%0)      \n\t"
            "add        $64,    %0          \n\t"
            "sub        $1, %%ecx           \n\t"
            "jnz        1b                  \n\t"
            :"+r"(fptr)
            :"c"(loops),"m"(g)
            :"xmm0","xmm1","xmm2","xmm3","xmm4"
        );
    }
#endif //Q_PROCESSOR_X86
    for (; i < samples; i++)
        *fptr++ *= g;
}
 
template <class AudioDataType>
void tMuteChannel(AudioDataType *buffer, int channels, int ch, int frames)
{
    AudioDataType *s1 = buffer + ch;
    AudioDataType *s2 = buffer - ch + 1;
 
    for (int i = 0; i < frames; i++)
    {
        *s1 = *s2;
        s1 += channels;
        s2 += channels;
    }
}
 
void AudioOutputUtil::MuteChannel(int obits, int channels, int ch,
                                  void *buffer, int bytes)
{
    int frames = bytes / ((obits >> 3) * channels);
 
    if (obits == 8)
        tMuteChannel((uint8_t *)buffer, channels, ch, frames);
    else if (obits == 16)
        tMuteChannel((short *)buffer, channels, ch, frames);
    else
        tMuteChannel((int *)buffer, channels, ch, frames);
}
 
char *AudioOutputUtil::GeneratePinkFrames(char *frames, int channels,
                                          int channel, int count, int bits)
{
    pink_noise_t pink{};
 
    initialize_pink_noise(&pink);
 
    auto *samp16 = (int16_t*) frames;
    auto *samp32 = (int32_t*) frames;
 
    while (count-- > 0)
    {
        for(int chn = 0 ; chn < channels; chn++)
        {
            if (chn==channel)
            {
                /* Don't use MAX volume */
                double res = generate_pink_noise_sample(&pink) *
                    static_cast<float>(0x03fffffff);
                int32_t ires = res;
                if (bits == 16)
                    *samp16++ = qToLittleEndian<qint16>(ires >> 16);
                else
                    *samp32++ = qToLittleEndian<qint32>(ires);
            }
            else
            {
                if (bits == 16)
                    *samp16++ = 0;
                else
                    *samp32++ = 0;
            }
        }
    }
    return frames;
}
 
int AudioOutputUtil::DecodeAudio(AVCodecContext *ctx,
                                 uint8_t *buffer, int &data_size,
                                 const AVPacket *pkt)
{
    MythAVFrame frame;
    bool got_frame = false;
 
    data_size = 0;
    if (!frame)
    {
        return AVERROR(ENOMEM);
    }
 
//  SUGGESTION
//  Now that avcodec_decode_audio4 is deprecated and replaced
//  by 2 calls (receive frame and send packet), this could be optimized
//  into separate routines or separate threads.
//  Also now that it always consumes a whole buffer some code
//  in the caller may be able to be optimized.
    int ret = avcodec_receive_frame(ctx,frame);
    if (ret == 0)
        got_frame = true;
    if (ret == AVERROR(EAGAIN))
        ret = 0;
    if (ret == 0)
        ret = avcodec_send_packet(ctx, pkt);
    if (ret == AVERROR(EAGAIN))
        ret = 0;
    else if (ret < 0)
    {
        std::string error;
        LOG(VB_AUDIO, LOG_ERR, LOC +
            QString("audio decode error: %1 (%2)")
            .arg(av_make_error_stdstring(error, ret))
            .arg(got_frame));
        return ret;
    }
    else
    {
        ret = pkt->size;
    }
 
    if (!got_frame)
    {
        LOG(VB_AUDIO, LOG_DEBUG, LOC +
            QString("audio decode, no frame decoded (%1)").arg(ret));
        return ret;
    }
 
    auto format = (AVSampleFormat)frame->format;
 
    data_size = frame->nb_samples * frame->ch_layout.nb_channels * av_get_bytes_per_sample(format);
 
    if (av_sample_fmt_is_planar(format))
    {
        InterleaveSamples(AudioOutputSettings::AVSampleFormatToFormat(format, ctx->bits_per_raw_sample),
                          frame->ch_layout.nb_channels, buffer, (const uint8_t **)frame->extended_data,
                          data_size);
    }
    else
    {
        // data is already compacted... simply copy it
        memcpy(buffer, frame->extended_data[0], data_size);
    }
 
    return ret;
}
 
void AudioOutputUtil::DeinterleaveSamples(AudioFormat format, int channels,
                                          uint8_t *output, const uint8_t *input,
                                          int data_size)
{
    AudioConvert::DeinterleaveSamples(format, channels, output, input, data_size);
}
 
void AudioOutputUtil::InterleaveSamples(AudioFormat format, int channels,
                                        uint8_t *output, const uint8_t * const *input,
                                        int data_size)
{
    AudioConvert::InterleaveSamples(format, channels, output, input, data_size);
}
 
void AudioOutputUtil::InterleaveSamples(AudioFormat format, int channels,
                                        uint8_t *output, const uint8_t *input,
                                        int data_size)
{
    AudioConvert::InterleaveSamples(format, channels, output, input, data_size);
}