Merge svn changes up to r29277

5 files changed, 49 insertions, 98 deletions

diff --git a/libao2/ao_macosx.c b/libao2/ao_coreaudio.c
index fb5883b3fb..3ef6d3367f 100644
--- a/libao2/ao_macosx.c
+++ b/libao2/ao_coreaudio.c
@@ -1,5 +1,5 @@
 /*
- * Mac OS X audio output driver
+ * CoreAudio audio output driver for Mac OS X
  *
  * original copyright (C) Timothy J. Wood - Aug 2000
  * ported to MPlayer libao2 by Dan Christiansen
@@ -52,21 +52,22 @@
 #include "audio_out_internal.h"
 #include "libaf/af_format.h"
 #include "osdep/timer.h"
+#include "libavutil/fifo.h"
 
 static const ao_info_t info =
   {
     "Darwin/Mac OS X native audio output",
-    "macosx",
+    "coreaudio",
     "Timothy J. Wood & Dan Christiansen & Chris Roccati",
     ""
   };
 
-LIBAO_EXTERN(macosx)
+LIBAO_EXTERN(coreaudio)
 
 /* Prefix for all mp_msg() calls */
-#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)
+#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c)
 
-typedef struct ao_macosx_s
+typedef struct ao_coreaudio_s
 {
   AudioDeviceID i_selected_dev;             /* Keeps DeviceID of the selected device. */
   int b_supports_digital;                   /* Does the currently selected device support digital mode? */
@@ -91,87 +92,35 @@ typedef struct ao_macosx_s
   int paused;
 
   /* Ring-buffer */
-  /* does not need explicit synchronization, but needs to allocate
-   * (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size
-   * data */
-  unsigned char *buffer;
-  unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size
+  AVFifoBuffer *buffer;
+  unsigned int buffer_len; ///< must always be num_chunks * chunk_size
   unsigned int num_chunks;
   unsigned int chunk_size;
-  
-  unsigned int buf_read_pos;
-  unsigned int buf_write_pos;
-} ao_macosx_t;
+} ao_coreaudio_t;
 
-static ao_macosx_t *ao = NULL;
-
-/**
- * \brief return number of free bytes in the buffer
- *    may only be called by mplayer's thread
- * \return minimum number of free bytes in buffer, value may change between
- *    two immediately following calls, and the real number of free bytes
- *    might actually be larger!
- */
-static int buf_free(void) {
-  int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size;
-  if (free < 0) free += ao->buffer_len;
-  return free;
-}
-
-/**
- * \brief return number of buffered bytes
- *    may only be called by playback thread
- * \return minimum number of buffered bytes, value may change between
- *    two immediately following calls, and the real number of buffered bytes
- *    might actually be larger!
- */
-static int buf_used(void) {
-  int used = ao->buf_write_pos - ao->buf_read_pos;
-  if (used < 0) used += ao->buffer_len;
-  return used;
-}
+static ao_coreaudio_t *ao = NULL;
 
 /**
  * \brief add data to ringbuffer
  */
 static int write_buffer(unsigned char* data, int len){
-  int first_len = ao->buffer_len - ao->buf_write_pos;
-  int free = buf_free();
+  int free = ao->buffer_len - av_fifo_size(ao->buffer);
   if (len > free) len = free;
-  if (first_len > len) first_len = len;
-  // till end of buffer
-  memcpy (&ao->buffer[ao->buf_write_pos], data, first_len);
-  if (len > first_len) { // we have to wrap around
-    // remaining part from beginning of buffer
-    memcpy (ao->buffer, &data[first_len], len - first_len);
-  }
-  ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len;
-  return len;
+  return av_fifo_generic_write(ao->buffer, data, len, NULL);
 }
 
 /**
  * \brief remove data from ringbuffer
  */
 static int read_buffer(unsigned char* data,int len){
-  int first_len = ao->buffer_len - ao->buf_read_pos;
-  int buffered = buf_used();
+  int buffered = av_fifo_size(ao->buffer);
   if (len > buffered) len = buffered;
-  if (first_len > len) first_len = len;
-  // till end of buffer
-  if (data) {
-    memcpy (data, &ao->buffer[ao->buf_read_pos], first_len);
-    if (len > first_len) { // we have to wrap around
-      // remaining part from beginning of buffer
-      memcpy (&data[first_len], ao->buffer, len - first_len);
-    }
-  }
-  ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len;
-  return len;
+  return av_fifo_generic_read(ao->buffer, data, len, NULL);
 }
 
 OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData)
 {
-int amt=buf_used();
+int amt=av_fifo_size(ao->buffer);
 int req=(inNumFrames)*ao->packetSize;
 
 	if(amt>req)
@@ -266,7 +215,7 @@ static void print_format(int lev, const char* str, const AudioStreamBasicDescrip
 
 static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
 static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
-static int OpenSPDIF();
+static int OpenSPDIF(void);
 static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
 static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
                                     const AudioTimeStamp * inNow,
@@ -299,7 +248,7 @@ int b_alive;
 
     ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
 
-    ao = calloc(1, sizeof(ao_macosx_t));
+    ao = calloc(1, sizeof(ao_coreaudio_t));
 
     ao->i_selected_dev = 0;
     ao->b_supports_digital = 0;
@@ -493,8 +442,8 @@ int b_alive;
 	ao_data.buffersize = ao_data.bps;
 
 	ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
-    ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
-    ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
+    ao->buffer_len = ao->num_chunks * ao->chunk_size;
+    ao->buffer = av_fifo_alloc(ao->buffer_len);
 	
 	ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
 
@@ -515,7 +464,7 @@ err_out2:
 err_out1:
     CloseComponent(ao->theOutputUnit);
 err_out:
-    free(ao->buffer);
+    av_fifo_free(ao->buffer);
     free(ao);
     ao = NULL;
     return CONTROL_FALSE; 
@@ -524,7 +473,7 @@ err_out:
 /*****************************************************************************
  * Setup a encoded digital stream (SPDIF)
  *****************************************************************************/
-static int OpenSPDIF()
+static int OpenSPDIF(void)
 {
     OSStatus                err = noErr;
     UInt32                  i_param_size, b_mix = 0;
@@ -737,8 +686,8 @@ static int OpenSPDIF()
     ao_data.buffersize = ao_data.bps;
 
     ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
-    ao->buffer_len = (ao->num_chunks + 1) * ao->chunk_size;
-    ao->buffer = calloc(ao->num_chunks + 1, ao->chunk_size);
+    ao->buffer_len = ao->num_chunks * ao->chunk_size;
+    ao->buffer = av_fifo_alloc(ao->buffer_len);
 
     ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
 
@@ -782,7 +731,7 @@ err_out:
             ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
                    (char *)&err);
     }
-    free(ao->buffer);
+    av_fifo_free(ao->buffer);
     free(ao);
     ao = NULL;
     return CONTROL_FALSE; 
@@ -981,7 +930,7 @@ static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
                                     const AudioTimeStamp * inOutputTime,
                                     void * threadGlobals )
 {
-    int amt = buf_used();
+    int amt = av_fifo_size(ao->buffer);
     int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
 
     if (amt > req)
@@ -1030,27 +979,22 @@ static int play(void* output_samples,int num_bytes,int flags)
 static void reset(void)
 {
   audio_pause();
-  /* reset ring-buffer state */
-  ao->buf_read_pos=0;
-  ao->buf_write_pos=0;
-  
-  return;
+  av_fifo_reset(ao->buffer);
 }
 
 
 /* return available space */
 static int get_space(void)
 {
-  return buf_free();
+  return ao->buffer_len - av_fifo_size(ao->buffer);
 }
 
 
 /* return delay until audio is played */
 static float get_delay(void)
 {
-  int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less
   // inaccurate, should also contain the data buffered e.g. by the OS
-  return (float)(buffered)/(float)ao_data.bps;
+  return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
 }
 
 
@@ -1061,8 +1005,8 @@ static void uninit(int immed)
   UInt32              i_param_size = 0;
 
   if (!immed) {
-    long long timeleft=(1000000LL*buf_used())/ao_data.bps;
-    ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", buf_used(), ao_data.bps, (int)timeleft);
+    long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
+    ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
     usec_sleep((int)timeleft);
   }
 
@@ -1115,7 +1059,7 @@ static void uninit(int immed)
       }
   }
 
-  free(ao->buffer);
+  av_fifo_free(ao->buffer);
   free(ao);
   ao = NULL;
 }
diff --git a/libao2/ao_jack.c b/libao2/ao_jack.c
index 494d44aa19..8ee5550602 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -82,7 +82,7 @@ static AVFifoBuffer *buffer;
  * If there is not enough room, the buffer is filled up
  */
 static int write_buffer(unsigned char* data, int len) {
-  int free = BUFFSIZE - av_fifo_size(buffer);
+  int free = av_fifo_space(buffer);
   if (len > free) len = free;
   return av_fifo_generic_write(buffer, data, len, NULL);
 }
@@ -337,7 +337,7 @@ static void audio_resume(void) {
 }
 
 static int get_space(void) {
-  return BUFFSIZE - av_fifo_size(buffer);
+  return av_fifo_space(buffer);
 }
 
 /**
diff --git a/libao2/ao_pulse.c b/libao2/ao_pulse.c
index 2d27c85540..b2fdcbb175 100644
--- a/libao2/ao_pulse.c
+++ b/libao2/ao_pulse.c
@@ -102,7 +102,10 @@ static void success_cb(pa_stream *s, int success, void *userdata) {
  */
 static int waitop(pa_operation *op) {
     pa_operation_state_t state;
-    if (!op) return 0;
+    if (!op) {
+        pa_threaded_mainloop_unlock(mainloop);
+        return 0;
+    }
     state = pa_operation_get_state(op);
     while (state == PA_OPERATION_RUNNING) {
         pa_threaded_mainloop_wait(mainloop);
@@ -150,11 +153,11 @@ static int init(int rate_hz, int channels, int format, int flags) {
     }
 
     broken_pause = 0;
-    // not sure which versions are affected, assume 0.9.1*
+    // not sure which versions are affected, assume 0.9.11* to 0.9.14*
     // known bad: 0.9.14, 0.9.13
-    // known good: 0.9.9, 0.9.10
+    // known good: 0.9.9, 0.9.10, 0.9.15
     // to test: pause, wait ca. 5 seconds framestep and see if MPlayer hangs somewhen
-    if (strncmp(version, "0.9.1", 5) == 0 && strncmp(version, "0.9.10", 6) != 0) {
+    if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1' && version[5] <= '4') {
         mp_msg(MSGT_AO, MSGL_WARN, "[pulse] working around probably broken pause functionality,\n"
                                    "        see http://www.pulseaudio.org/ticket/440\n");
         broken_pause = 1;
@@ -389,12 +392,16 @@ static int control(int cmd, void *arg) {
                 volume.values[1] = (pa_volume_t)vol->right*PA_VOLUME_NORM/100;
             }
 
-            if (!(o = pa_context_set_sink_input_volume(context, pa_stream_get_index(stream), &volume, NULL, NULL))) {
+            pa_threaded_mainloop_lock(mainloop);
+            o = pa_context_set_sink_input_volume(context, pa_stream_get_index(stream), &volume, NULL, NULL);
+            if (!o) {
+                pa_threaded_mainloop_unlock(mainloop);
                 GENERIC_ERR_MSG(context, "pa_context_set_sink_input_volume() failed");
                 return CONTROL_ERROR;
             }
             /* We don't wait for completion here */
             pa_operation_unref(o);
+            pa_threaded_mainloop_unlock(mainloop);
             return CONTROL_OK;
         }
 
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index 63ddec6f29..df643cb62a 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -67,7 +67,7 @@ static unsigned char volume=SDL_MIX_MAXVOLUME;
 #endif
 
 static int write_buffer(unsigned char* data,int len){
-  int free = BUFFSIZE - av_fifo_size(buffer);
+  int free = av_fifo_space(buffer);
   if (len > free) len = free;
   return av_fifo_generic_write(buffer, data, len, NULL);
 }
@@ -280,7 +280,7 @@ static void audio_resume(void)
 
 // return: how many bytes can be played without blocking
 static int get_space(void){
-    return BUFFSIZE - av_fifo_size(buffer);
+    return av_fifo_space(buffer);
 }
 
 // plays 'len' bytes of 'data'
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index 8c64abbb95..fe95ab192f 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -31,7 +31,7 @@ ao_data_t ao_data={0,0,0,0,OUTBURST,-1,0};
 char *ao_subdevice = NULL;
 
 extern const ao_functions_t audio_out_oss;
-extern const ao_functions_t audio_out_macosx;
+extern const ao_functions_t audio_out_coreaudio;
 extern const ao_functions_t audio_out_arts;
 extern const ao_functions_t audio_out_esd;
 extern const ao_functions_t audio_out_pulse;
@@ -67,7 +67,7 @@ const ao_functions_t* const audio_out_drivers[] =
         &audio_out_dart,
 #endif
 #ifdef CONFIG_COREAUDIO
-        &audio_out_macosx,
+        &audio_out_coreaudio,
 #endif
 #ifdef CONFIG_OSS_AUDIO
         &audio_out_oss,