Merge branch 'master' into ps3

src/audio/alsa/alsa_audio.c

@@ -95,15 +95,6 @@ alsa_open(alsa_audio_mode_t *aam, int format, int rate)
  snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED);
  snd_pcm_hw_params_set_format(h, hwp, aam->aam_format);
 
- switch(rate) {
- default:
- case AM_SR_96000: rate = 96000; break;
- case AM_SR_48000: rate = 48000; break;
- case AM_SR_44100: rate = 44100; break;
- case AM_SR_32000: rate = 32000; break;
- case AM_SR_24000: rate = 24000; break;
- }
-
  snd_pcm_hw_params_set_rate(h, hwp, rate, 0);
 
  aam->aam_sample_rate = rate;
@@ -388,10 +379,11 @@ alsa_audio_start(audio_mode_t *am, audio_fifo_t *af)
  continue;
  }
 
- if(h == NULL || ab->ab_format != cur_format || ab->ab_rate != cur_rate) {
+ if(h == NULL ||
+ ab->ab_format != cur_format ||
+ ab->ab_samplerate != cur_rate) {
 
- if(!(ab->ab_format & am->am_formats) || 
- !(ab->ab_rate & am->am_sample_rates)) {
+ if(!(ab->ab_format & am->am_formats)) {
  /* Rate / format is not supported by this mode */
  ab_free(ab);
  if(h == NULL)
@@ -404,7 +396,7 @@ alsa_audio_start(audio_mode_t *am, audio_fifo_t *af)
  snd_pcm_close(h);
 
  cur_format = ab->ab_format;
- cur_rate = ab->ab_rate;
+ cur_rate = ab->ab_samplerate;
 
  pts = AV_NOPTS_VALUE;
 

src/audio/audio.h

@@ -59,7 +59,7 @@ typedef struct audio_mode {
 #define AM_SR_44100 0x4
 #define AM_SR_32000 0x8
 #define AM_SR_24000 0x10
-
+#define AM_SR_ANY 0x20
  unsigned int am_sample_rate;
 
  char *am_title;

src/audio/audio_decoder.c

@@ -209,7 +209,7 @@ audio_deliver_passthru(media_buf_t *mb, audio_decoder_t *ad, int format,
  ab = af_alloc(mb->mb_size, mp);
  ab->ab_channels = 2;
  ab->ab_format = format;
- ab->ab_rate = AM_SR_48000;
+ ab->ab_samplerate= 48000;
  ab->ab_frames = mb->mb_size;
  ab->ab_pts = mb->mb_pts;
  ab->ab_epoch = mb->mb_epoch;
@@ -365,11 +365,11 @@ ad_decode_buf(audio_decoder_t *ad, media_pipe_t *mp, media_queue_t *mq,
  break;
  case SAMPLE_FMT_FLT:
  for(i = 0; i < frames; i++)
- ad->ad_outbuf[i] = rintf(((float *)ad->ad_outbuf)[i]) * (1 << 15);
+ ad->ad_outbuf[i] = rintf(((float *)ad->ad_outbuf)[i] * 32768);
  break;
  case SAMPLE_FMT_DBL:
  for(i = 0; i < frames; i++)
- ad->ad_outbuf[i] = rint(((float *)ad->ad_outbuf)[i]) * (1 << 15);
+ ad->ad_outbuf[i] = rint(((double *)ad->ad_outbuf)[i] * 32768);
  break;
  }
 
@@ -708,7 +708,10 @@ audio_mix1(audio_decoder_t *ad, audio_mode_t *am,
  /**
  * Resampling
  */
- if(!(rf & am->am_sample_rates)) {
+ if(rf & am->am_sample_rates || am->am_sample_rates & AM_SR_ANY) {
+ close_resampler(ad);
+ audio_mix2(ad, am, channels, rate, data0, frames, pts, epoch, mp);
+ } else {
 
  int dstrate = 48000;
  int consumed;
@@ -753,9 +756,6 @@ audio_mix1(audio_decoder_t *ad, audio_mode_t *am,
  if(consumed == 0 && written == 0)
  break;
  }
- } else {
- close_resampler(ad);
- audio_mix2(ad, am, channels, rate, data0, frames, pts, epoch, mp);
  }
 }
 
@@ -911,7 +911,6 @@ audio_deliver(audio_decoder_t *ad, audio_mode_t *am, int16_t *src,
  int c, r;
 
  int format;
- int rf = audio_rateflag_from_rate(rate);
 
  switch(channels) {
  case 2: format = AM_FORMAT_PCM_STEREO; break;
@@ -934,7 +933,7 @@ audio_deliver(audio_decoder_t *ad, audio_mode_t *am, int16_t *src,
  ab->ab_channels = channels;
  ab->ab_alloced = outsize;
  ab->ab_format = format;
- ab->ab_rate = rf;
+ ab->ab_samplerate = rate;
  ab->ab_frames = 0;
  ab->ab_pts = AV_NOPTS_VALUE;
  }

src/audio/audio_fifo.h

@@ -27,7 +27,7 @@ typedef struct audio_buf {
  TAILQ_ENTRY(audio_buf) link;
  unsigned int ab_flush;
  unsigned int ab_format;
- unsigned int ab_rate;
+ unsigned int ab_samplerate;
  unsigned int ab_channels;
  int64_t ab_pts;
  int ab_epoch;

src/audio/coreaudio/coreaudio.c

@@ -187,10 +187,11 @@ audioDeviceIOProc(AudioDeviceID inDevice,
  return 0;
  }
 
- if(ab->ab_format != cam->cam_cur_format || ab->ab_rate != cam->cam_cur_rate) {
- coreaudio_change_format(cam, ab->ab_format, ab->ab_rate);
+ if(ab->ab_format != cam->cam_cur_format ||
+ ab->ab_samplerate != cam->cam_cur_rate) {
+ coreaudio_change_format(cam, ab->ab_format, ab->ab_samplerate);
  cam->cam_cur_format = ab->ab_format;
- cam->cam_cur_rate = ab->ab_rate;
+ cam->cam_cur_rate = ab->ab_samplerate;
  }
 
  if(ab->ab_pts != AV_NOPTS_VALUE) {
@@ -235,7 +236,7 @@ coreaudio_change_format(coreaudio_audio_mode_t *cam, int format, int rate)
  AudioStreamBasicDescription asbd;
 
  asbd.mFormatID = kAudioFormatLinearPCM;
- asbd.mSampleRate = audio_rate_from_rateflag(rate);
+ asbd.mSampleRate = rate;
  asbd.mFormatFlags = 
  kAudioFormatFlagIsFloat | 
  kAudioFormatFlagIsPacked |

src/audio/pulseaudio/pulseaudio.c

@@ -106,14 +106,7 @@ stream_setup(pa_audio_mode_t *pam, audio_buf_t *ab)
  memset(&pam->ss, 0, sizeof(pa_sample_spec));
 
  pam->ss.format = PA_SAMPLE_S16NE; 
- switch(ab->ab_rate) {
- default:
- case AM_SR_96000: pam->ss.rate = 96000; break;
- case AM_SR_48000: pam->ss.rate = 48000; break;
- case AM_SR_44100: pam->ss.rate = 44100; break;
- case AM_SR_32000: pam->ss.rate = 32000; break;
- case AM_SR_24000: pam->ss.rate = 24000; break;
- }
+ pam->ss.rate = ab->ab_samplerate;
 
  switch(ab->ab_format) {
  case AM_FORMAT_PCM_STEREO:
@@ -189,7 +182,7 @@ stream_setup(pa_audio_mode_t *pam, audio_buf_t *ab)
  n = pa_stream_connect_playback(s, NULL, &pba, flags, &cv, NULL);
 
  pam->stream = s;
- pam->cur_rate = ab->ab_rate;
+ pam->cur_rate = ab->ab_samplerate;
  pam->cur_format = ab->ab_format;
 }
 
@@ -463,7 +456,7 @@ pa_audio_start(audio_mode_t *am, audio_fifo_t *af)
  }
  if(pam->stream != NULL &&
  (pam->cur_format != ab->ab_format ||
- pam->cur_rate != ab->ab_rate)) {
+ pam->cur_rate != ab->ab_samplerate)) {
  stream_destroy(pam);
  }
 
@@ -603,8 +596,7 @@ audio_pa_init(void)
  am = &pam->am;
  am->am_formats = 
  AM_FORMAT_PCM_STEREO | AM_FORMAT_PCM_5DOT1 | AM_FORMAT_PCM_7DOT1;
- am->am_sample_rates = AM_SR_96000 | AM_SR_48000 | AM_SR_44100 | 
- AM_SR_32000 | AM_SR_24000;
+ am->am_sample_rates = AM_SR_ANY;
  am->am_title = strdup("Pulseaudio");
  am->am_id = strdup("pulseaudio");
  am->am_preferred_size = 1024;