[vlc-commits] audiotrack: implement time synchronization

[Rafaël Carré]

vlc | branch: master | Rafaël Carré <funman at videolan.org> | Tue Dec 18 23:39:24 2012 +0100| [141c6716f8f2905ec7de50979bc7126183e0733a] | committer: Rafaël Carré

audiotrack: implement time synchronization

There is still a random offset between A/V which stays
constant until there's a flush but hopefully testing should
give a clue how to fix it.

Note: the symbols required are present only since 2.2

> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=141c6716f8f2905ec7de50979bc7126183e0733a
---

 modules/audio_output/audiotrack.c |   78 +++++++++++++++++++++++++++++++++++--
 1 file changed, 74 insertions(+), 4 deletions(-)

diff --git a/modules/audio_output/audiotrack.c b/modules/audio_output/audiotrack.c
index bea55ff..5e76fa8 100644
--- a/modules/audio_output/audiotrack.c
+++ b/modules/audio_output/audiotrack.c
@@ -71,6 +71,8 @@ typedef int (*AudioSystem_getOutputSamplingRate)(int *, int);
 // _ZN7android10AudioTrack16getMinFrameCountEPiij
 typedef int (*AudioTrack_getMinFrameCount)(int *, int, unsigned int);
 
+// _ZN7android11AudioSystem17getRenderPositionEPjS1_i
+typedef int (*AudioTrack_getRenderPosition)(uint32_t *, uint32_t *, int);
 // _ZN7android10AudioTrackC1EijiiijPFviPvS1_ES1_ii
 typedef void (*AudioTrack_ctor)(void *, int, unsigned int, int, int, int, unsigned int, void (*)(int, void *, void *), void *, int, int);
 // _ZN7android10AudioTrackC1EijiiijPFviPvS1_ES1_i
@@ -94,6 +96,11 @@ struct aout_sys_t {
     float soft_gain;
     bool soft_mute;
 
+    int rate;
+    uint32_t samples_written;
+    uint32_t initial;
+    int bytes_per_frame;
+
     void *libmedia;
     void *AudioTrack;
 
@@ -111,6 +118,7 @@ struct aout_sys_t {
     AudioTrack_write at_write;
     AudioTrack_flush at_flush;
     AudioTrack_pause at_pause;
+    AudioTrack_getRenderPosition at_getRenderPosition;
 };
 
 /* Soft volume helper */
@@ -122,6 +130,7 @@ static int  Open(vlc_object_t *);
 static void Close(vlc_object_t *);
 static void Play(audio_output_t*, block_t*);
 static void Pause (audio_output_t *, bool, mtime_t);
+static void Flush (audio_output_t *, bool);
 
 vlc_module_begin ()
     set_shortname("AudioTrack")
@@ -164,6 +173,11 @@ static void *InitLibrary(struct aout_sys_t *p_sys)
     p_sys->at_flush = (AudioTrack_flush)(dlsym(p_library, "_ZN7android10AudioTrack5flushEv"));
     p_sys->at_pause = (AudioTrack_pause)(dlsym(p_library, "_ZN7android10AudioTrack5pauseEv"));
 
+    /* this symbol can have different names depending on the mangling */
+    p_sys->at_getRenderPosition = (AudioTrack_getRenderPosition)(dlsym(p_library, "_ZN7android11AudioSystem17getRenderPositionEPjS1_i"));
+    if (!p_sys->at_getRenderPosition)
+        p_sys->at_getRenderPosition = (AudioTrack_getRenderPosition)(dlsym(p_library, "_ZN7android11AudioSystem17getRenderPositionEPjS1_19audio_stream_type_t"));
+
     /* We need the first 3 or the last 1 */
     if (!((p_sys->as_getOutputFrameCount && p_sys->as_getOutputLatency && p_sys->as_getOutputSamplingRate)
         || p_sys->at_getMinFrameCount)) {
@@ -180,6 +194,34 @@ static void *InitLibrary(struct aout_sys_t *p_sys)
     return p_library;
 }
 
+static int TimeGet(audio_output_t *p_aout, mtime_t *restrict delay)
+{
+    aout_sys_t *p_sys = p_aout->sys;
+    uint32_t hal, dsp;
+
+    if (!p_sys->at_getRenderPosition)
+        return -1;
+
+    if (p_sys->at_getRenderPosition(&hal, &dsp, MUSIC))
+        return -1;
+
+    hal = (uint32_t)((uint64_t)hal * p_sys->rate / 44100);
+
+    if (p_sys->samples_written == 0) {
+        p_sys->initial = hal;
+        return -1;
+    }
+
+    hal -= p_sys->initial;
+    if (hal == 0)
+        return -1;
+
+    if (delay)
+        *delay = ((mtime_t)p_sys->samples_written - hal) * CLOCK_FREQ / p_sys->rate;
+
+    return 0;
+}
+
 static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt)
 {
     struct aout_sys_t *p_sys = aout->sys;
@@ -277,8 +319,15 @@ static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt)
     aout->time_get = NULL;
     aout->play = Play;
     aout->pause = Pause;
+    aout->flush = Flush;
+    aout->time_get = TimeGet;
+
+    p_sys->rate = rate;
+    p_sys->samples_written = 0;
+    p_sys->bytes_per_frame = aout_FormatNbChannels(fmt) * (format == PCM_16_BIT) ? 2 : 1;
 
     p_sys->at_start(p_sys->AudioTrack);
+    TimeGet(aout, NULL); /* Gets the initial value of DAC samples counter */
 
     fmt->i_rate = rate;
 
@@ -295,14 +344,20 @@ static void Stop(audio_output_t* p_aout)
     free(p_sys->AudioTrack);
 }
 
-/* FIXME: lipsync */
 static void Play(audio_output_t* p_aout, block_t* p_buffer)
 {
     aout_sys_t *p_sys = p_aout->sys;
 
-    size_t length = 0;
-    while (length < p_buffer->i_buffer) {
-        length += p_sys->at_write(p_sys->AudioTrack, (char*)(p_buffer->p_buffer) + length, p_buffer->i_buffer - length);
+    while (p_buffer->i_buffer) {
+        int ret = p_sys->at_write(p_sys->AudioTrack, p_buffer->p_buffer, p_buffer->i_buffer);
+        if (ret < 0) {
+            msg_Err(p_aout, "Write failed (error %d)", ret);
+            break;
+        }
+
+        p_sys->samples_written += ret / p_sys->bytes_per_frame;
+        p_buffer->p_buffer += ret;
+        p_buffer->i_buffer -= ret;
     }
 
     block_Release( p_buffer );
@@ -321,6 +376,21 @@ static void Pause(audio_output_t *p_aout, bool pause, mtime_t date)
     }
 }
 
+static void Flush (audio_output_t *p_aout, bool wait)
+{
+    aout_sys_t *p_sys = p_aout->sys;
+    if (wait) {
+        mtime_t delay;
+        if (!TimeGet(p_aout, &delay))
+            msleep(delay);
+    } else {
+        p_sys->at_stop(p_sys->AudioTrack);
+        p_sys->at_flush(p_sys->AudioTrack);
+        p_sys->samples_written = 0;
+        p_sys->at_start(p_sys->AudioTrack);
+    }
+}
+
 static int Open(vlc_object_t *obj)
 {
     audio_output_t *aout = (audio_output_t *)obj;