#include <vlc_input.h>
#include "aout_internal.h"
+#include "libvlc.h"
-#undef aout_DecNew
/**
* Creates an audio output
*/
-aout_input_t *aout_DecNew( audio_output_t *p_aout,
- audio_sample_format_t *p_format,
- const audio_replay_gain_t *p_replay_gain,
- const aout_request_vout_t *p_request_vout )
+int aout_DecNew( audio_output_t *p_aout,
+ const audio_sample_format_t *p_format,
+ const audio_replay_gain_t *p_replay_gain,
+ const aout_request_vout_t *p_request_vout )
{
/* Sanitize audio format */
if( p_format->i_channels > 32 )
{
msg_Err( p_aout, "too many audio channels (%u)",
p_format->i_channels );
- return NULL;
+ return -1;
}
if( p_format->i_channels <= 0 )
{
msg_Err( p_aout, "no audio channels" );
- return NULL;
+ return -1;
}
if( p_format->i_channels != aout_FormatNbChannels( p_format ) )
{
msg_Err( p_aout, "incompatible audio channels count with layout mask" );
- return NULL;
+ return -1;
}
if( p_format->i_rate > 192000 )
{
msg_Err( p_aout, "excessive audio sample frequency (%u)",
p_format->i_rate );
- return NULL;
+ return -1;
}
if( p_format->i_rate < 4000 )
{
msg_Err( p_aout, "too low audio sample frequency (%u)",
p_format->i_rate );
- return NULL;
+ return -1;
}
aout_input_t *p_input = calloc( 1, sizeof(aout_input_t));
if( !p_input )
- return NULL;
+ return -1;
p_input->b_error = true;
- p_input->b_paused = false;
- p_input->i_pause_date = 0;
-
- aout_FormatPrepare( p_format );
memcpy( &p_input->input, p_format,
sizeof(audio_sample_format_t) );
/* We can only be called by the decoder, so no need to lock
* p_input->lock. */
+ aout_owner_t *owner = aout_owner(p_aout);
aout_lock( p_aout );
- assert( p_aout->p_input == NULL );
- p_aout->p_input = p_input;
+ assert (owner->input == NULL);
var_Destroy( p_aout, "audio-device" );
var_Destroy( p_aout, "audio-channels" );
/* Recreate the output using the new format. */
if( aout_OutputNew( p_aout, p_format ) < 0 )
-#warning Input without output and mixer = bad idea.
- goto out;
+ goto error;
- assert( p_aout->p_mixer == NULL );
- if( aout_MixerNew( p_aout ) == -1 )
- {
- aout_OutputDelete( p_aout );
-#warning Memory leak.
- p_input = NULL;
- goto out;
- }
+ assert (owner->volume.mixer == NULL);
+ owner->volume.mixer = aout_MixerNew (p_aout, owner->mixer_format.i_format);
+
+ date_Init (&owner->sync.date, owner->mixer_format.i_rate, 1);
+ date_Set (&owner->sync.date, VLC_TS_INVALID);
+ owner->input = p_input;
aout_InputNew( p_aout, p_input, p_request_vout );
-out:
aout_unlock( p_aout );
- return p_input;
+ return 0;
+error:
+ aout_unlock( p_aout );
+ free( p_input );
+ return -1;
}
/*****************************************************************************
* aout_DecDelete : delete a decoder
*****************************************************************************/
-void aout_DecDelete( audio_output_t * p_aout, aout_input_t * p_input )
+void aout_DecDelete( audio_output_t * p_aout )
{
+ aout_owner_t *owner = aout_owner (p_aout);
+ aout_input_t *input;
+ struct audio_mixer *mixer;
+
aout_lock( p_aout );
/* Remove the input. */
- assert( p_input == p_aout->p_input ); /* buggy decoder? */
- p_aout->p_input = NULL;
- aout_InputDelete( p_aout, p_input );
+ input = owner->input;
+ aout_InputDelete (p_aout, input);
+ owner->input = NULL;
aout_OutputDelete( p_aout );
- aout_MixerDelete( p_aout );
+ mixer = owner->volume.mixer;
+ owner->volume.mixer = NULL;
var_Destroy( p_aout, "audio-device" );
var_Destroy( p_aout, "audio-channels" );
aout_unlock( p_aout );
- free( p_input );
+
+ aout_MixerDelete (mixer);
+ free (input);
+}
+
+static void aout_CheckRestart (audio_output_t *aout)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ aout_input_t *input = owner->input;
+
+ aout_assert_locked (aout);
+
+ if (likely(!owner->need_restart))
+ return;
+ owner->need_restart = false;
+
+ /* Reinitializes the output */
+ aout_InputDelete (aout, owner->input);
+ aout_MixerDelete (owner->volume.mixer);
+ owner->volume.mixer = NULL;
+ aout_OutputDelete (aout);
+
+ if (aout_OutputNew (aout, &input->input))
+ {
+ input->b_error = true;
+ return; /* we are officially screwed */
+ }
+
+ owner->volume.mixer = aout_MixerNew (aout, owner->mixer_format.i_format);
+
+ if (aout_InputNew (aout, input, &input->request_vout))
+ assert (input->b_error);
+ else
+ assert (!input->b_error);
}
/*****************************************************************************
* aout_DecNewBuffer : ask for a new empty buffer
*****************************************************************************/
-aout_buffer_t * aout_DecNewBuffer( aout_input_t * p_input,
- size_t i_nb_samples )
+block_t *aout_DecNewBuffer (audio_output_t *aout, size_t samples)
{
- size_t length = i_nb_samples * p_input->input.i_bytes_per_frame
- / p_input->input.i_frame_length;
+ /* NOTE: the caller is responsible for serializing input change */
+ aout_owner_t *owner = aout_owner (aout);
+ aout_input_t *input = owner->input;
+
+ size_t length = samples * input->input.i_bytes_per_frame
+ / input->input.i_frame_length;
block_t *block = block_Alloc( length );
if( likely(block != NULL) )
{
- block->i_nb_samples = i_nb_samples;
+ block->i_nb_samples = samples;
block->i_pts = block->i_length = 0;
}
return block;
/*****************************************************************************
* aout_DecDeleteBuffer : destroy an undecoded buffer
*****************************************************************************/
-void aout_DecDeleteBuffer( audio_output_t * p_aout, aout_input_t * p_input,
- aout_buffer_t * p_buffer )
+void aout_DecDeleteBuffer (audio_output_t *aout, block_t *block)
{
- (void)p_aout; (void)p_input;
- aout_BufferFree( p_buffer );
+ (void) aout;
+ aout_BufferFree (block);
}
/*****************************************************************************
* aout_DecPlay : filter & mix the decoded buffer
*****************************************************************************/
-int aout_DecPlay( audio_output_t * p_aout, aout_input_t * p_input,
- aout_buffer_t * p_buffer, int i_input_rate )
+int aout_DecPlay (audio_output_t *p_aout, block_t *p_buffer, int i_input_rate)
{
+ aout_owner_t *owner = aout_owner (p_aout);
+ aout_input_t *p_input = owner->input;
+
assert( i_input_rate >= INPUT_RATE_DEFAULT / AOUT_MAX_INPUT_RATE &&
i_input_rate <= INPUT_RATE_DEFAULT * AOUT_MAX_INPUT_RATE );
-
assert( p_buffer->i_pts > 0 );
p_buffer->i_length = (mtime_t)p_buffer->i_nb_samples * 1000000
return -1;
}
+ aout_CheckRestart( p_aout );
aout_InputCheckAndRestart( p_aout, p_input );
- aout_InputPlay( p_aout, p_input, p_buffer, i_input_rate );
- /* Run the mixer if it is able to run. */
- aout_MixerRun( p_aout, p_aout->mixer_multiplier * p_input->multiplier );
+
+ /* Input */
+ p_buffer = aout_InputPlay (p_aout, p_input, p_buffer, i_input_rate,
+ &owner->sync.date);
+ if( p_buffer != NULL )
+ {
+ date_Increment (&owner->sync.date, p_buffer->i_nb_samples);
+
+ /* Mixer */
+ float amp = owner->volume.multiplier * p_input->multiplier;
+ aout_MixerRun (owner->volume.mixer, p_buffer, amp);
+
+ /* Output */
+ aout_OutputPlay( p_aout, p_buffer );
+ }
+
aout_unlock( p_aout );
return 0;
}
-int aout_DecGetResetLost( audio_output_t *p_aout, aout_input_t *p_input )
+int aout_DecGetResetLost (audio_output_t *aout)
{
+ aout_owner_t *owner = aout_owner (aout);
+ aout_input_t *input = owner->input;
int val;
- aout_lock( p_aout );
- val = p_input->i_buffer_lost;
- p_input->i_buffer_lost = 0;
- aout_unlock( p_aout );
+ aout_lock (aout);
+ val = input->i_buffer_lost;
+ input->i_buffer_lost = 0;
+ aout_unlock (aout);
return val;
}
-void aout_DecChangePause( audio_output_t *p_aout, aout_input_t *p_input, bool b_paused, mtime_t i_date )
+void aout_DecChangePause (audio_output_t *aout, bool paused, mtime_t date)
{
- mtime_t i_duration = 0;
-
- aout_lock( p_aout );
- assert( p_aout->p_input == p_input );
- assert( !p_input->b_paused || !b_paused );
- if( p_input->b_paused )
- {
- i_duration = i_date - p_input->i_pause_date;
- }
- p_input->b_paused = b_paused;
- p_input->i_pause_date = i_date;
+ aout_owner_t *owner = aout_owner (aout);
- if( i_duration != 0 )
- {
- for( aout_buffer_t *p = p_input->mixer.fifo.p_first; p != NULL; p = p->p_next )
- {
- p->i_pts += i_duration;
- }
- }
- aout_OutputPause( p_aout, b_paused, i_date );
- aout_unlock( p_aout );
+ aout_lock (aout);
+ /* XXX: Should the date be offset by the pause duration instead? */
+ date_Set (&owner->sync.date, VLC_TS_INVALID);
+ aout_OutputPause (aout, paused, date);
+ aout_unlock (aout);
}
-void aout_DecFlush( audio_output_t *p_aout, aout_input_t *p_input )
+void aout_DecFlush (audio_output_t *aout)
{
- aout_lock( p_aout );
- aout_FifoSet( &p_input->mixer.fifo, 0 );
- aout_unlock( p_aout );
+ aout_owner_t *owner = aout_owner (aout);
+
+ aout_lock (aout);
+ date_Set (&owner->sync.date, VLC_TS_INVALID);
+ aout_OutputFlush (aout, false);
+ aout_unlock (aout);
}
-bool aout_DecIsEmpty( audio_output_t * p_aout, aout_input_t * p_input )
+bool aout_DecIsEmpty (audio_output_t *aout)
{
+ aout_owner_t *owner = aout_owner (aout);
mtime_t end_date;
- aout_lock( p_aout );
- end_date = aout_FifoNextStart( &p_input->mixer.fifo );
- aout_unlock( p_aout );
- return end_date <= mdate();
+ aout_lock (aout);
+ /* FIXME: tell output to drain */
+ end_date = date_Get (&owner->sync.date);
+ aout_unlock (aout);
+ return end_date == VLC_TS_INVALID || end_date <= mdate();
+}
+
+/**
+ * Notifies the audio input of the drift from the requested audio
+ * playback timestamp (@ref block_t.i_pts) to the anticipated playback time
+ * as reported by the audio output hardware.
+ * Depending on the drift amplitude, the input core may ignore the drift
+ * trigger upsampling or downsampling, or even discard samples.
+ * Future VLC versions may instead adjust the input decoding speed.
+ *
+ * The audio output plugin is responsible for estimating the ideal current
+ * playback time defined as follows:
+ * ideal time = buffer timestamp - (output latency + pending buffer duration)
+ *
+ * Practically, this is the PTS (block_t.i_pts) of the current buffer minus
+ * the latency reported by the output programming interface.
+ * Computing the estimated drift directly would probably be more intuitive.
+ * However the use of an absolute time value does not introduce extra
+ * measurement errors due to the CPU scheduling jitter and clock resolution.
+ * Furthermore, the ideal while it is an abstract value, is easy for most
+ * audio output plugins to compute.
+ * The following definition is equivalent but depends on the clock time:
+ * ideal time = real time + drift
+
+ * @note If aout_LatencyReport() is never called, the core will assume that
+ * there is no drift.
+ *
+ * @param ideal estimated ideal time as defined above.
+ */
+void aout_TimeReport (audio_output_t *aout, mtime_t ideal)
+{
+ mtime_t delta = mdate() - ideal /* = -drift */;
+
+ aout_assert_locked (aout);
+ if (delta < -AOUT_MAX_PTS_ADVANCE || +AOUT_MAX_PTS_DELAY < delta)
+ {
+ aout_owner_t *owner = aout_owner (aout);
+
+ msg_Warn (aout, "not synchronized (%"PRId64" us), resampling",
+ delta);
+ if (date_Get (&owner->sync.date) != VLC_TS_INVALID)
+ date_Move (&owner->sync.date, delta);
+ }
}
projects / vlc / blobdiff