core: add a spsc ringbuffer implementation

Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.

I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.

The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).

Also adapted ao_coreaudio to use this ringbuffer.

1 files changed, 22 insertions, 21 deletions

diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index 930bb2d741..d3491f17f8 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -49,8 +49,7 @@
 #include "audio/format.h"
 #include "osdep/timer.h"
 #include "core/subopt-helper.h"
-
-#include "ao_coreaudio/ca_ringbuffer_internal.h"
+#include "core/mp_ring.h"
 
 #define ca_msg(a, b, c ...) mp_msg(a, b, "AO: [coreaudio] " c)
 
@@ -58,10 +57,10 @@ static void audio_pause(struct ao *ao);
 static void audio_resume(struct ao *ao);
 static void reset(struct ao *ao);
 
-static void print_buffer(struct ca_ringbuffer *buffer)
+static void print_buffer(struct mp_ring *buffer)
 {
     void *tctx = talloc_new(NULL);
-    ca_msg(MSGT_AO, MSGL_V, "%s\n", ca_ringbuffer_repr(buffer, tctx));
+    ca_msg(MSGT_AO, MSGL_V, "%s\n", mp_ring_repr(buffer, tctx));
     talloc_free(tctx);
 }
 
@@ -92,7 +91,7 @@ struct priv
     int packetSize;
     int paused;
 
-    struct ca_ringbuffer *buffer;
+    struct mp_ring *buffer;
 };
 
 static OSStatus theRenderProc(void *inRefCon,
@@ -104,14 +103,14 @@ static OSStatus theRenderProc(void *inRefCon,
     struct ao *ao  = inRefCon;
     struct priv *p = ao->priv;
 
-    int buffered  = ca_ringbuffer_buffered(p->buffer);
+    int buffered  = mp_ring_buffered(p->buffer);
     int requested = inNumFrames * p->packetSize;
 
     if (buffered > requested)
         buffered = requested;
 
     if (buffered) {
-        ca_ringbuffer_read(p->buffer,
+        mp_ring_read(p->buffer,
                            (unsigned char *)ioData->mBuffers[0].mData,
                            buffered);
     } else {
@@ -622,8 +621,9 @@ static int init(struct ao *ao, char *params)
 
     ao->bps        = ao->samplerate * inDesc.mBytesPerFrame;
     ao->buffersize = ao->bps;
-    p->buffer      = ca_ringbuffer_new2(p, ao->bps, maxFrames);
-    ao->outburst   = ca_ringbuffer_chunk_size(p->buffer);
+    int bufbytes   = af_fmt_seconds_to_bytes(ao->format, 0.5, ao->channels.num);
+    p->buffer      = mp_ring_new(p, bufbytes);
+    ao->outburst   = maxFrames;
 
     print_buffer(p->buffer);
 
@@ -859,8 +859,9 @@ static int OpenSPDIF(struct ao *ao)
     /* For ac3/dts, just use packet size 6144 bytes as chunk size. */
     int chunk_size = p->stream_format.mBytesPerPacket;
     ao->buffersize = ao->bps;
-    p->buffer      = ca_ringbuffer_new2(p, ao->bps, chunk_size);
-    ao->outburst   = ca_ringbuffer_chunk_size(p->buffer);
+    int bufbytes   = af_fmt_seconds_to_bytes(ao->format, 0.5, ao->channels.num);
+    p->buffer      = mp_ring_new(p, bufbytes);
+    ao->outburst   = chunk_size;
 
     print_buffer(p->buffer);
 
@@ -1071,7 +1072,7 @@ static OSStatus RenderCallbackSPDIF(AudioDeviceID inDevice,
 {
     struct ao *ao  = threadGlobals;
     struct priv *p = ao->priv;
-    int amt = ca_ringbuffer_buffered(p->buffer);
+    int amt = mp_ring_buffered(p->buffer);
     AudioBuffer ca_buffer = outOutputData->mBuffers[p->i_stream_index];
     int req = ca_buffer.mDataByteSize;
 
@@ -1079,9 +1080,9 @@ static OSStatus RenderCallbackSPDIF(AudioDeviceID inDevice,
         amt = req;
     if (amt) {
         if (p->b_muted) {
-            ca_ringbuffer_read(p->buffer, NULL, amt);
+            mp_ring_read(p->buffer, NULL, amt);
         } else {
-            ca_ringbuffer_read(p->buffer, (unsigned char *)ca_buffer.mData, amt);
+            mp_ring_read(p->buffer, (unsigned char *)ca_buffer.mData, amt);
         }
     }
 
@@ -1116,7 +1117,7 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
                    "Detected current stream does not support digital.\n");
     }
 
-    wrote = ca_ringbuffer_write(p->buffer, output_samples, num_bytes);
+    wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
     audio_resume(ao);
 
     return wrote;
@@ -1127,7 +1128,7 @@ static void reset(struct ao *ao)
 {
     struct priv *p = ao->priv;
     audio_pause(ao);
-    ca_ringbuffer_reset(p->buffer);
+    mp_ring_reset(p->buffer);
 }
 
 
@@ -1135,7 +1136,7 @@ static void reset(struct ao *ao)
 static int get_space(struct ao *ao)
 {
     struct priv *p = ao->priv;
-    return ca_ringbuffer_available(p->buffer);
+    return mp_ring_available(p->buffer);
 }
 
 
@@ -1144,7 +1145,7 @@ static float get_delay(struct ao *ao)
 {
     // inaccurate, should also contain the data buffered e.g. by the OS
     struct priv *p = ao->priv;
-    return ca_ringbuffer_buffered(p->buffer) / (float)ao->bps;
+    return mp_ring_buffered(p->buffer) / (float)ao->bps;
 }
 
 static void uninit(struct ao *ao, bool immed)
@@ -1154,9 +1155,9 @@ static void uninit(struct ao *ao, bool immed)
 
     if (!immed) {
         long long timeleft =
-            (1000000LL * ca_ringbuffer_buffered(p->buffer)) / ao->bps;
-        ca_msg(MSGT_AO, MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", ca_ringbuffer_buffered(
-                   p->buffer), ao->bps, (int)timeleft);
+            (1000000LL * mp_ring_buffered(p->buffer)) / ao->bps;
+        ca_msg(MSGT_AO, MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n",
+                mp_ring_buffered(p->buffer), ao->bps, (int)timeleft);
         mp_sleep_us((int)timeleft);
     }