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https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
Makes audio mixer optional.
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Johns
parent
2661fdf333
commit
780e2989ae
73
audio.c
73
audio.c
@ -40,6 +40,7 @@
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//#define USE_ALSA ///< enable alsa support
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//#define USE_OSS ///< enable OSS support
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#define USE_AUDIO_THREAD ///< use thread for audio playback
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#define USE_AUDIO_MIXER ///< use audio module mixer
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#include <stdio.h>
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#include <stdint.h>
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@ -187,7 +188,7 @@ static int AudioRatesInHw[AudioRatesMax];
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/// input to hardware channel matrix
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static int AudioChannelMatrix[AudioRatesMax][9];
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/// rates tables
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/// rates tables (must be sorted by frequency)
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static const unsigned AudioRatesTable[AudioRatesMax] = {
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44100, 48000,
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};
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@ -400,6 +401,8 @@ static void AudioSoftAmplifier(int16_t * samples, int count)
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}
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}
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#ifdef USE_AUDIO_MIXER
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/**
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** Upmix mono to stereo.
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**
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@ -603,6 +606,8 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
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}
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}
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#endif
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//----------------------------------------------------------------------------
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// ring buffer
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//----------------------------------------------------------------------------
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@ -633,7 +638,7 @@ static atomic_t AudioRingFilled; ///< how many of the ring is used
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static unsigned AudioStartThreshold; ///< start play, if filled
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/**
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** Add sample-rate, number of channel change to ring.
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** Add sample-rate, number of channels change to ring.
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**
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** @param sample_rate sample-rate frequency
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** @param channels number of channels
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@ -641,6 +646,8 @@ static unsigned AudioStartThreshold; ///< start play, if filled
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**
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** @retval -1 error
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** @retval 0 okay
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**
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** @note this function shouldn't fail. Checks are done during AudoInit.
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*/
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static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
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{
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@ -649,13 +656,16 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3)
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// search supported sample-rates
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for (u = 0; u < AudioRatesMax; ++u) {
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if (AudioRatesTable[u] == sample_rate) {
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goto found;
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}
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if (AudioRatesTable[u] > sample_rate) {
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break;
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}
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}
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if (u == AudioRatesMax) { // unsupported sample-rate
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Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate);
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return -1;
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}
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Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate);
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return -1; // unsupported sample-rate
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found:
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if (!AudioChannelMatrix[u][channels]) {
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Error(_("audio: %d channels unsupported\n"), channels);
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return -1; // unsupported nr. of channels
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@ -1712,7 +1722,7 @@ static int64_t OssGetDelay(void)
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/**
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** Setup OSS audio for requested format.
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**
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** @param sample_rate sample rate/frequency
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** @param sample_rate sample rate/frequency
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** @param channels number of channels
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** @param use_ac3 use ac3/pass-through device
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**
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@ -2163,7 +2173,6 @@ void AudioEnqueue(const void *samples, int count)
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{
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size_t n;
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int16_t *buffer;
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int frames;
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#ifdef noDEBUG
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static uint32_t last_tick;
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@ -2185,28 +2194,40 @@ void AudioEnqueue(const void *samples, int count)
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AudioRing[AudioRingWrite].PacketSize = count;
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Debug(3, "audio: a/v packet size %d bytes\n", count);
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}
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if (AudioRing[AudioRingWrite].UseAc3) {
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buffer = (void *)samples;
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} else {
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//
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// Convert / resample input to hardware format
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//
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// audio sample modification allowed and needed?
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buffer = (void *)samples;
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if (!AudioRing[AudioRingWrite].UseAc3 && (AudioCompression
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|| AudioNormalize
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|| AudioRing[AudioRingWrite].InChannels !=
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AudioRing[AudioRingWrite].HwChannels)) {
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int frames;
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// resample into ring-buffer is too complex in the case of a roundabout
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// just use a temporary buffer
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frames =
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count / (AudioRing[AudioRingWrite].InChannels *
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AudioBytesProSample);
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buffer =
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alloca(frames * AudioRing[AudioRingWrite].HwChannels *
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AudioBytesProSample);
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#ifdef USE_AUDIO_MIXER
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// Convert / resample input to hardware format
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AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames,
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buffer, AudioRing[AudioRingWrite].HwChannels);
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#else
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#ifdef DEBUG
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if (AudioRing[AudioRingWrite].InChannels !=
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AudioRing[AudioRingWrite].HwChannels) {
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Debug(3, "audio: internal failure channels mismatch\n");
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return;
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}
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#endif
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memcpy(buffer, samples, count);
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#endif
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count =
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frames * AudioRing[AudioRingWrite].HwChannels *
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AudioBytesProSample;
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// resample into ring-buffer is too complex in the case of a roundabout
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// just use a temporary buffer
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if (AudioCompression) { // in place operation
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AudioCompressor(buffer, count);
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}
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@ -2551,6 +2572,8 @@ void AudioSetVolume(int volume)
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** @retval 0 everything ok
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** @retval 1 didn't support frequency/channels combination
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** @retval -1 something gone wrong
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**
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** @todo add support to report best fitting format.
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*/
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int AudioSetup(int *freq, int *channels, int use_ac3)
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{
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@ -2771,7 +2794,12 @@ void AudioInit(void)
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freq = 44100;
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AudioRatesInHw[Audio44100] = 0;
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for (chan = 1; chan < 9; ++chan) {
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if (AudioUsedModule->Setup(&freq, &chan, 0)) {
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int tchan;
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int tfreq;
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tchan = chan;
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tfreq = freq;
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if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
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AudioChannelsInHw[chan] = 0;
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} else {
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AudioChannelsInHw[chan] = chan;
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@ -2781,10 +2809,15 @@ void AudioInit(void)
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freq = 48000;
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AudioRatesInHw[Audio48000] = 0;
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for (chan = 1; chan < 9; ++chan) {
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int tchan;
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int tfreq;
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if (!AudioChannelsInHw[chan]) {
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continue;
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}
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if (AudioUsedModule->Setup(&freq, &chan, 0)) {
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tchan = chan;
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tfreq = freq;
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if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
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AudioChannelsInHw[chan] = 0;
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} else {
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AudioChannelsInHw[chan] = chan;
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