From 780e2989aec9b5e1d5f506c5badde3dbcb45821c Mon Sep 17 00:00:00 2001 From: Johns Date: Wed, 6 Feb 2013 16:02:22 +0100 Subject: [PATCH] Makes audio mixer optional. --- audio.c | 73 +++++++++++++++++++++++++++++++++++++++++---------------- 1 file changed, 53 insertions(+), 20 deletions(-) diff --git a/audio.c b/audio.c index cc3a99d..ed6198e 100644 --- a/audio.c +++ b/audio.c @@ -40,6 +40,7 @@ //#define USE_ALSA ///< enable alsa support //#define USE_OSS ///< enable OSS support #define USE_AUDIO_THREAD ///< use thread for audio playback +#define USE_AUDIO_MIXER ///< use audio module mixer #include #include @@ -187,7 +188,7 @@ static int AudioRatesInHw[AudioRatesMax]; /// input to hardware channel matrix static int AudioChannelMatrix[AudioRatesMax][9]; - /// rates tables + /// rates tables (must be sorted by frequency) static const unsigned AudioRatesTable[AudioRatesMax] = { 44100, 48000, }; @@ -400,6 +401,8 @@ static void AudioSoftAmplifier(int16_t * samples, int count) } } +#ifdef USE_AUDIO_MIXER + /** ** Upmix mono to stereo. ** @@ -603,6 +606,8 @@ static void AudioResample(const int16_t * in, int in_chan, int frames, } } +#endif + //---------------------------------------------------------------------------- // ring buffer //---------------------------------------------------------------------------- @@ -633,7 +638,7 @@ static atomic_t AudioRingFilled; ///< how many of the ring is used static unsigned AudioStartThreshold; ///< start play, if filled /** -** Add sample-rate, number of channel change to ring. +** Add sample-rate, number of channels change to ring. ** ** @param sample_rate sample-rate frequency ** @param channels number of channels @@ -641,6 +646,8 @@ static unsigned AudioStartThreshold; ///< start play, if filled ** ** @retval -1 error ** @retval 0 okay +** +** @note this function shouldn't fail. Checks are done during AudoInit. */ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3) { @@ -649,13 +656,16 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int use_ac3) // search supported sample-rates for (u = 0; u < AudioRatesMax; ++u) { if (AudioRatesTable[u] == sample_rate) { + goto found; + } + if (AudioRatesTable[u] > sample_rate) { break; } } - if (u == AudioRatesMax) { // unsupported sample-rate - Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate); - return -1; - } + Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate); + return -1; // unsupported sample-rate + + found: if (!AudioChannelMatrix[u][channels]) { Error(_("audio: %d channels unsupported\n"), channels); return -1; // unsupported nr. of channels @@ -1712,7 +1722,7 @@ static int64_t OssGetDelay(void) /** ** Setup OSS audio for requested format. ** -** @param sample_rate sample rate/frequency +** @param sample_rate sample rate/frequency ** @param channels number of channels ** @param use_ac3 use ac3/pass-through device ** @@ -2163,7 +2173,6 @@ void AudioEnqueue(const void *samples, int count) { size_t n; int16_t *buffer; - int frames; #ifdef noDEBUG static uint32_t last_tick; @@ -2185,28 +2194,40 @@ void AudioEnqueue(const void *samples, int count) AudioRing[AudioRingWrite].PacketSize = count; Debug(3, "audio: a/v packet size %d bytes\n", count); } - if (AudioRing[AudioRingWrite].UseAc3) { - buffer = (void *)samples; - } else { - // - // Convert / resample input to hardware format - // + // audio sample modification allowed and needed? + buffer = (void *)samples; + if (!AudioRing[AudioRingWrite].UseAc3 && (AudioCompression + || AudioNormalize + || AudioRing[AudioRingWrite].InChannels != + AudioRing[AudioRingWrite].HwChannels)) { + int frames; + + // resample into ring-buffer is too complex in the case of a roundabout + // just use a temporary buffer frames = count / (AudioRing[AudioRingWrite].InChannels * AudioBytesProSample); buffer = alloca(frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); +#ifdef USE_AUDIO_MIXER + // Convert / resample input to hardware format AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames, buffer, AudioRing[AudioRingWrite].HwChannels); - +#else +#ifdef DEBUG + if (AudioRing[AudioRingWrite].InChannels != + AudioRing[AudioRingWrite].HwChannels) { + Debug(3, "audio: internal failure channels mismatch\n"); + return; + } +#endif + memcpy(buffer, samples, count); +#endif count = frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample; - // resample into ring-buffer is too complex in the case of a roundabout - // just use a temporary buffer - if (AudioCompression) { // in place operation AudioCompressor(buffer, count); } @@ -2551,6 +2572,8 @@ void AudioSetVolume(int volume) ** @retval 0 everything ok ** @retval 1 didn't support frequency/channels combination ** @retval -1 something gone wrong +** +** @todo add support to report best fitting format. */ int AudioSetup(int *freq, int *channels, int use_ac3) { @@ -2771,7 +2794,12 @@ void AudioInit(void) freq = 44100; AudioRatesInHw[Audio44100] = 0; for (chan = 1; chan < 9; ++chan) { - if (AudioUsedModule->Setup(&freq, &chan, 0)) { + int tchan; + int tfreq; + + tchan = chan; + tfreq = freq; + if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) { AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan; @@ -2781,10 +2809,15 @@ void AudioInit(void) freq = 48000; AudioRatesInHw[Audio48000] = 0; for (chan = 1; chan < 9; ++chan) { + int tchan; + int tfreq; + if (!AudioChannelsInHw[chan]) { continue; } - if (AudioUsedModule->Setup(&freq, &chan, 0)) { + tchan = chan; + tfreq = freq; + if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) { AudioChannelsInHw[chan] = 0; } else { AudioChannelsInHw[chan] = chan;