bool AudioProcessor::Reset(int sample_rate, int num_channels)
{
if (num_channels <= 0) {
DEBUG("chromaprint::AudioProcessor::Reset() -- No audio channels.");
return false;
}
if (sample_rate <= kMinSampleRate) {
DEBUG("chromaprint::AudioProcessor::Reset() -- Sample rate less than "
<< kMinSampleRate << " (" << sample_rate << ").");
return false;
}
m_buffer_offset = 0;
if (m_resample_ctx) {
av_resample_close(m_resample_ctx);
m_resample_ctx = 0;
}
if (sample_rate != m_target_sample_rate) {
m_resample_ctx = av_resample_init(
m_target_sample_rate, sample_rate,
kResampleFilterLength,
kResamplePhaseShift,
kResampleLinear,
kResampleCutoff);
}
m_num_channels = num_channels;
return true;
}
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_resample_t* s = (af_resample_t*)af->setup;
af_data_t *data= (af_data_t*)arg;
int out_rate, test_output_res; // helpers for checking input format
switch(cmd){
case AF_CONTROL_REINIT:
if((af->data->rate == data->rate) || (af->data->rate == 0))
return AF_DETACH;
af->data->nch = data->nch;
if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
af->mul = (double)af->data->rate / data->rate;
af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
if (s->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length ||
s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) {
if(s->avrctx) av_resample_close(s->avrctx);
s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
s->ctx_out_rate = af->data->rate;
s->ctx_in_rate = data->rate;
s->ctx_filter_size = s->filter_length;
s->ctx_phase_shift = s->phase_shift;
s->ctx_linear = s->linear;
s->ctx_cutoff = s->cutoff;
}
// hack to make af_test_output ignore the samplerate change
out_rate = af->data->rate;
af->data->rate = data->rate;
test_output_res = af_test_output(af, (af_data_t*)arg);
af->data->rate = out_rate;
return test_output_res;
case AF_CONTROL_COMMAND_LINE:{
s->cutoff= 0.0;
sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
return AF_OK;
}
case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
af->data->rate = *(int*)arg;
return AF_OK;
}
return AF_UNKNOWN;
}
int CFfAudioResample::Init( int nInRate, int nOutRate, int nFilterLength, int nLog2PhaseCount, int nLinear, double dCutoff )
{_STT();
// Out with the old
Destroy();
// Validate sample rates
if ( 0 >= nInRate || 0 >= nOutRate )
return 0;
// Build resampling context
m_ctx = av_resample_init( nOutRate, nInRate, nFilterLength, nLog2PhaseCount, nLinear, dCutoff );
if ( !m_ctx )
return 0;
// Save rates
m_nInRate = nInRate;
m_nOutRate = nOutRate;
return 1;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AResampleContext *aresample = ctx->priv;
if (aresample->out_rate == -1)
aresample->out_rate = outlink->sample_rate;
else
outlink->sample_rate = aresample->out_rate;
outlink->time_base = (AVRational) {
1, aresample->out_rate
};
//TODO: make the resampling parameters configurable
aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
16, 10, 0, 0.8);
aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
inlink->sample_rate, outlink->sample_rate);
return 0;
}
void BE_ST_InitAudio(void)
{
g_sdlAudioSubsystemUp = false;
g_sdlEmulatedOPLChipReady = false;
int inSampleRate = BE_Cross_GetSelectedGameVerSampleRate();
bool doDigitized = (inSampleRate != 0);
if (!doDigitized)
inSampleRate = OPL_SAMPLE_RATE;
if (g_refKeenCfg.sndSubSystem)
{
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
BE_Cross_LogMessage(BE_LOG_MSG_WARNING, "SDL audio system initialization failed,\n%s\n", SDL_GetError());
}
else
{
g_sdlAudioSpec.freq = g_refKeenCfg.sndSampleRate;
#ifdef MIXER_SAMPLE_FORMAT_FLOAT
g_sdlAudioSpec.format = AUDIO_F32SYS;
#elif (defined MIXER_SAMPLE_FORMAT_SINT16)
g_sdlAudioSpec.format = AUDIO_S16SYS;
#endif
g_sdlAudioSpec.channels = 1;
// Should be some power-of-two roughly proportional to the sample rate; Using 1024 for 48000Hz.
for (g_sdlAudioSpec.samples = 1; g_sdlAudioSpec.samples < g_refKeenCfg.sndSampleRate/64; g_sdlAudioSpec.samples *= 2)
{
}
if (doDigitized)
g_sdlAudioSpec.callback = (g_refKeenCfg.sndSampleRate == inSampleRate) ? BEL_ST_Simple_DigiCallBack : BEL_ST_Resampling_DigiCallBack;
else
g_sdlAudioSpec.callback = ((g_refKeenCfg.sndSampleRate == inSampleRate) || !g_refKeenCfg.oplEmulation) ? BEL_ST_Simple_EmuCallBack : BEL_ST_Resampling_EmuCallBack;
g_sdlAudioSpec.userdata = NULL;
if (SDL_OpenAudio(&g_sdlAudioSpec, NULL))
{
BE_Cross_LogMessage(BE_LOG_MSG_WARNING, "Cannot open SDL audio device,\n%s\n", SDL_GetError());
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
else
{
#ifdef REFKEEN_CONFIG_THREADS
g_sdlCallbackMutex = SDL_CreateMutex();
if (!g_sdlCallbackMutex)
{
BE_Cross_LogMessage(BE_LOG_MSG_ERROR, "Cannot create recursive mutex for SDL audio callback,\n%s\nClosing SDL audio subsystem\n", SDL_GetError());
SDL_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
else
#endif
{
BE_Cross_LogMessage(BE_LOG_MSG_NORMAL, "Audio subsystem initialized, requested spec: freq %d, format %u, channels %d, samples %u\n", (int)g_sdlAudioSpec.freq, (unsigned int)g_sdlAudioSpec.format, (int)g_sdlAudioSpec.channels, (unsigned int)g_sdlAudioSpec.samples);
g_sdlAudioSubsystemUp = true;
}
}
}
}
// If the audio subsystem is off, let us simulate a byte rate
// of 1000Hz (same as SDL_GetTicks() time units)
if (!g_sdlAudioSubsystemUp)
{
g_sdlAudioSpec.freq = doDigitized ? inSampleRate : (NUM_OF_BYTES_FOR_SOUND_CALLBACK_WITH_DISABLED_SUBSYSTEM / sizeof(BE_ST_SndSample_T));
g_sdlAudioSpec.callback = doDigitized ? BEL_ST_Resampling_DigiCallBack : BEL_ST_Resampling_EmuCallBack;
return;
}
if (g_refKeenCfg.oplEmulation)
{
YM3812Init(1, 3579545, OPL_SAMPLE_RATE);
g_sdlEmulatedOPLChipReady = true;
}
if ((doDigitized || g_sdlEmulatedOPLChipReady) && (g_sdlAudioSpec.freq != inSampleRate))
{
// Should allocate this first, for g_sdlSrcData.data_in
g_sdlMiscOutNumOfSamples = 2*g_sdlAudioSpec.samples;
g_sdlMiscOutSamples = (BE_ST_SndSample_T *)malloc(sizeof(BE_ST_SndSample_T) * g_sdlMiscOutNumOfSamples);
if (g_sdlMiscOutSamples == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: Out of memory! (Failed to allocate g_sdlMiscOutSamples.)");
#ifndef REFKEEN_RESAMPLER_NONE
if (g_refKeenCfg.useResampler)
{
#if (!defined REFKEEN_RESAMPLER_LIBRESAMPLE) && (!defined REFKEEN_RESAMPLER_LIBAVCODEC)
char errMsg[160];
#endif
#if (defined REFKEEN_RESAMPLER_LIBSWRESAMPLE)
g_sdlSwrContext = swr_alloc_set_opts(
NULL, // allocating a new context
AV_CH_LAYOUT_MONO, // out channels layout
AV_SAMPLE_FMT_S16, // out format
g_sdlAudioSpec.freq, // out rate
AV_CH_LAYOUT_MONO, // in channels layout
AV_SAMPLE_FMT_S16, // in format
inSampleRate, // in rate
0,
NULL
);
if (g_sdlSwrContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: swr_alloc_set_opts failed!");
int error = swr_init(g_sdlSwrContext);
if (error != 0)
{
// av_err2str requires libavutil/libavutil-ffmpeg, so don't convert code to string
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: swr_init failed! Error code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBAVRESAMPLE)
g_sdlAvAudioResampleContext = avresample_alloc_context();
if (g_sdlAvAudioResampleContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: avresample_alloc_context failed!");
av_opt_set_int(g_sdlAvAudioResampleContext, "in_channel_layout", AV_CH_LAYOUT_MONO, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_channel_layout", AV_CH_LAYOUT_MONO, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "in_sample_rate", inSampleRate, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_sample_rate", g_sdlAudioSpec.freq, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
int error = avresample_open(g_sdlAvAudioResampleContext);
if (error != 0)
{
// av_err2str requires libavutil/libavutil-ffmpeg, so don't convert code to string
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: swr_init failed! Error code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBAVCODEC)
avcodec_register_all();
g_sdlAvResampleContext = av_resample_init(
g_sdlAudioSpec.freq, // out rate
inSampleRate, // in rate
16, // filter length
10, // phase count
0, // linear FIR filter
1.0 // cutoff frequency
);
if (g_sdlAvResampleContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: av_resample_init failed!");
#elif (defined REFKEEN_RESAMPLER_LIBRESAMPLE)
g_sdlResampleFactor = (double)g_sdlAudioSpec.freq/inSampleRate;
g_sdlResampleHandle = resample_open(0, g_sdlResampleFactor, g_sdlResampleFactor);
if (g_sdlResampleHandle == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: resample_open failed!");
#elif (defined REFKEEN_RESAMPLER_LIBSOXR)
soxr_io_spec_t io_spec = soxr_io_spec(SOXR_INT16, SOXR_INT16);
soxr_quality_spec_t q_spec = soxr_quality_spec(SOXR_LQ, 0); // Default quality spec adds an audible latency for resampling to 8000Hz
soxr_error_t error;
g_sdlSoxr = soxr_create(
inSampleRate, // in rate
g_sdlAudioSpec.freq, // out rate
1, // channels
&error,
&io_spec,
&q_spec,
NULL // runtime spec
);
if (g_sdlSoxr == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: soxr_create failed!\nError: %s", soxr_strerror(error));
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBSPEEXDSP)
int error;
g_sdlSpeexResamplerState = speex_resampler_init(
1, // channels
inSampleRate, // in rate
g_sdlAudioSpec.freq, // out rate
0, // quality in the range 0-10 (10 is higher)
&error
);
if (g_sdlSpeexResamplerState == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: speex_resampler_init failed! Error code: %d\nError: %s", error, speex_resampler_strerror(error));
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBSAMPLERATE)
int error;
g_sdlSrcResampler = src_new(SRC_SINC_FASTEST, 1, &error);
if (g_sdlSrcResampler == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: src_new failed!\nError code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
g_sdlSrcData.data_in = doDigitized ? g_sdlMiscOutSamples : g_sdlALOutSamples;
g_sdlSrcData.src_ratio = (double)g_sdlAudioSpec.freq / inSampleRate;
#endif
}
else
#endif // REFKEEN_RESAMPLER_NONE
{
// The sum of all entries should be g_sdlAudioSpec.freq,
// "uniformly" distributed over g_sdlALSampleRateConvTable
g_sdlSampleRateConvTable = (int *)malloc(sizeof(int) * inSampleRate);
if (g_sdlSampleRateConvTable == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: Failed to allocate memory for sample rate conversion!");
g_sdlSampleRateConvTableSize = inSampleRate;
for (int i = 0; i < inSampleRate; ++i)
{
// Using uint64_t cause an overflow is possible
g_sdlSampleRateConvTable[i] = ((uint64_t)(i+1)*(uint64_t)g_sdlAudioSpec.freq/inSampleRate)-(uint64_t)i*(uint64_t)g_sdlAudioSpec.freq/inSampleRate;
}
g_sdlSampleRateConvCurrIndex = 0;
g_sdlSampleRateConvCounter = 0;
}
}
}
