Commit 58a31eb2 authored by Rafaël Carré's avatar Rafaël Carré

opensles: take in account all buffered audio in latency

We only considered the audio already buffered through opensles engine,
so the calculation was incorrect.

Use samples rather than bytes for calculation, and a function rather than
hardcoding 2*2 all over the place when a size in bytes is needed.
parent 8e0f4af9
...@@ -103,16 +103,18 @@ struct aout_sys_t ...@@ -103,16 +103,18 @@ struct aout_sys_t
/* audio buffered through opensles */ /* audio buffered through opensles */
uint8_t *buf; uint8_t *buf;
size_t buf_unit_size; size_t samples_per_buf;
int next_buf; int next_buf;
int rate;
/* if we can measure latency already */ /* if we can measure latency already */
bool started; bool started;
/* audio not yet buffered through opensles */ /* audio not yet buffered through opensles */
block_t *p_buffer_chain; block_t *p_buffer_chain;
block_t **pp_buffer_last; block_t **pp_buffer_last;
size_t samples;
}; };
/***************************************************************************** /*****************************************************************************
...@@ -140,6 +142,11 @@ vlc_module_end () ...@@ -140,6 +142,11 @@ vlc_module_end ()
* *
*****************************************************************************/ *****************************************************************************/
static inline int bytesPerSample(void)
{
return 2 /* S16 */ * 2 /* stereo */;
}
static int TimeGet(audio_output_t* aout, mtime_t* restrict drift) static int TimeGet(audio_output_t* aout, mtime_t* restrict drift)
{ {
aout_sys_t *sys = aout->sys; aout_sys_t *sys = aout->sys;
...@@ -158,8 +165,12 @@ static int TimeGet(audio_output_t* aout, mtime_t* restrict drift) ...@@ -158,8 +165,12 @@ static int TimeGet(audio_output_t* aout, mtime_t* restrict drift)
if (!started) if (!started)
return -1; return -1;
*drift = CLOCK_FREQ * OPENSLES_BUFLEN * st.count / 1000; *drift = (CLOCK_FREQ * OPENSLES_BUFLEN * st.count / 1000)
//msg_Dbg(aout, "latency %"PRId64"", *drift); + sys->samples * CLOCK_FREQ / sys->rate;
/* msg_Dbg(aout, "latency %"PRId64" ms, %d/%d buffers", *drift / 1000,
(int)st.count, OPENSLES_BUFFERS); */
return 0; return 0;
} }
...@@ -221,7 +232,7 @@ static void Pause(audio_output_t *aout, bool pause, mtime_t date) ...@@ -221,7 +232,7 @@ static void Pause(audio_output_t *aout, bool pause, mtime_t date)
static int WriteBuffer(audio_output_t *aout) static int WriteBuffer(audio_output_t *aout)
{ {
aout_sys_t *sys = aout->sys; aout_sys_t *sys = aout->sys;
const size_t unit_size = sys->buf_unit_size; const size_t unit_size = sys->samples_per_buf * bytesPerSample();
block_t *b = sys->p_buffer_chain; block_t *b = sys->p_buffer_chain;
if (!b) if (!b)
...@@ -282,6 +293,8 @@ static int WriteBuffer(audio_output_t *aout) ...@@ -282,6 +293,8 @@ static int WriteBuffer(audio_output_t *aout)
SLresult r = Enqueue(sys->playerBufferQueue, SLresult r = Enqueue(sys->playerBufferQueue,
&sys->buf[unit_size * sys->next_buf], unit_size); &sys->buf[unit_size * sys->next_buf], unit_size);
sys->samples -= sys->samples_per_buf;
if (r == SL_RESULT_SUCCESS) { if (r == SL_RESULT_SUCCESS) {
if (++sys->next_buf == OPENSLES_BUFFERS) if (++sys->next_buf == OPENSLES_BUFFERS)
sys->next_buf = 0; sys->next_buf = 0;
...@@ -305,6 +318,8 @@ static void Play(audio_output_t *aout, block_t *p_buffer) ...@@ -305,6 +318,8 @@ static void Play(audio_output_t *aout, block_t *p_buffer)
p_buffer->p_next = NULL; /* Make sur our linked list doesn't use old references */ p_buffer->p_next = NULL; /* Make sur our linked list doesn't use old references */
vlc_mutex_lock(&sys->lock); vlc_mutex_lock(&sys->lock);
sys->samples += p_buffer->i_buffer / bytesPerSample();
/* Hold this block until we can write it into the OpenSL buffer */ /* Hold this block until we can write it into the OpenSL buffer */
block_ChainLastAppend(&sys->pp_buffer_last, p_buffer); block_ChainLastAppend(&sys->pp_buffer_last, p_buffer);
...@@ -400,9 +415,9 @@ static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt) ...@@ -400,9 +415,9 @@ static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt)
CHECK_OPENSL_ERROR("Failed to switch to playing state"); CHECK_OPENSL_ERROR("Failed to switch to playing state");
/* XXX: rounding shouldn't affect us at normal sampling rate */ /* XXX: rounding shouldn't affect us at normal sampling rate */
sys->buf_unit_size = OPENSLES_BUFLEN * fmt->i_rate * 2 /* channels */ * 2 /* bps */ / 1000; sys->rate = fmt->i_rate;
sys->buf_unit_size = (sys->buf_unit_size + 3) & ~3; // align on sample boundary sys->samples_per_buf = OPENSLES_BUFLEN * fmt->i_rate / 1000;
sys->buf = malloc(OPENSLES_BUFFERS * sys->buf_unit_size); sys->buf = malloc(OPENSLES_BUFFERS * sys->samples_per_buf * bytesPerSample());
if (!sys->buf) if (!sys->buf)
goto error; goto error;
......
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