/*****************************************************************************
* dec.c : audio output API towards decoders
*****************************************************************************
- * Copyright (C) 2002-2007 the VideoLAN team
+ * Copyright (C) 2002-2007 VLC authors and VideoLAN
* $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
*
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
#include <vlc_common.h>
#include <vlc_aout.h>
#include <vlc_input.h>
-#include <vlc_atomic.h>
#include "aout_internal.h"
#include "libvlc.h"
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 -1;
- }
- if( p_format->i_channels <= 0 )
- {
- msg_Err( p_aout, "no audio channels" );
- return -1;
- }
if( p_format->i_channels != aout_FormatNbChannels( p_format ) )
{
msg_Err( p_aout, "incompatible audio channels count with layout mask" );
return -1;
}
- if( p_format->i_rate > 192000 )
+ if( p_format->i_rate > 352800 )
{
msg_Err( p_aout, "excessive audio sample frequency (%u)",
p_format->i_rate );
return -1;
}
- aout_input_t *p_input = calloc( 1, sizeof(aout_input_t));
- if( !p_input )
- return -1;
-
- p_input->b_error = true;
+ aout_owner_t *owner = aout_owner(p_aout);
- if( p_replay_gain )
- p_input->replay_gain = *p_replay_gain;
+ /* TODO: reduce lock scope depending on decoder's real need */
+ aout_OutputLock (p_aout);
- /* 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 (owner->input == NULL);
+ var_Destroy( p_aout, "stereo-mode" );
- var_Destroy( p_aout, "audio-device" );
- var_Destroy( p_aout, "audio-channels" );
+ /* Create the audio output stream */
+ owner->volume = aout_volume_New (p_aout, p_replay_gain);
- /* Recreate the output using the new format. */
+ atomic_store (&owner->restart, 0);
owner->input_format = *p_format;
- vlc_atomic_set (&owner->restart, 0);
- if( aout_OutputNew( p_aout, p_format ) < 0 )
+ owner->mixer_format = owner->input_format;
+ owner->request_vout = *p_request_vout;
+
+ if (aout_OutputNew (p_aout, &owner->mixer_format))
goto error;
+ aout_volume_SetFormat (owner->volume, owner->mixer_format.i_format);
- assert (owner->volume.mixer == NULL);
- owner->volume.mixer = aout_MixerNew (p_aout, owner->mixer_format.i_format);
+ /* Create the audio filtering "input" pipeline */
+ owner->filters = aout_FiltersNew (p_aout, p_format, &owner->mixer_format,
+ &owner->request_vout);
+ if (owner->filters == NULL)
+ {
+ aout_OutputDelete (p_aout);
+error:
+ aout_volume_Delete (owner->volume);
+ aout_OutputUnlock (p_aout);
+ return -1;
+ }
- date_Init (&owner->sync.date, owner->mixer_format.i_rate, 1);
- date_Set (&owner->sync.date, VLC_TS_INVALID);
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ owner->sync.discontinuity = true;
+ aout_OutputUnlock (p_aout);
- owner->input = p_input;
- aout_InputNew( p_aout, p_format, &owner->mixer_format, p_input,
- p_request_vout );
- aout_unlock( p_aout );
+ atomic_init (&owner->buffers_lost, 0);
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 )
+/**
+ * Stops all plugins involved in the audio output.
+ */
+void aout_DecDelete (audio_output_t *aout)
{
- aout_owner_t *owner = aout_owner (p_aout);
- aout_input_t *input;
- struct audio_mixer *mixer;
-
- aout_lock( p_aout );
- /* Remove the input. */
- input = owner->input;
- aout_InputDelete (p_aout, input);
- owner->input = NULL;
-
- aout_OutputDelete( 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 );
-
- aout_MixerDelete (mixer);
- free (input);
+ aout_owner_t *owner = aout_owner (aout);
+
+ aout_OutputLock (aout);
+ if (owner->mixer_format.i_format)
+ {
+ aout_FiltersDelete (aout, owner->filters);
+ aout_OutputDelete (aout);
+ }
+ aout_volume_Delete (owner->volume);
+ aout_OutputUnlock (aout);
+ var_Destroy (aout, "stereo-mode");
}
-#define AOUT_RESTART_OUTPUT 1
-#define AOUT_RESTART_INPUT 2
-static void aout_CheckRestart (audio_output_t *aout)
+static int aout_CheckReady (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
- aout_input_t *input = owner->input;
-
- aout_assert_locked (aout);
-
- int restart = vlc_atomic_swap (&owner->restart, 0);
- if (likely(restart == 0))
- return;
-
- assert (restart & AOUT_RESTART_INPUT);
- aout_InputDelete (aout, input);
- /* Reinitializes the output */
- if (restart & AOUT_RESTART_OUTPUT)
+ int restart = atomic_exchange (&owner->restart, 0);
+ if (unlikely(restart))
{
- aout_MixerDelete (owner->volume.mixer);
- owner->volume.mixer = NULL;
- aout_OutputDelete (aout);
+ if (owner->mixer_format.i_format)
+ aout_FiltersDelete (aout, owner->filters);
+
+ if (restart & AOUT_RESTART_OUTPUT)
+ { /* Reinitializes the output */
+ msg_Dbg (aout, "restarting output...");
+ if (owner->mixer_format.i_format)
+ aout_OutputDelete (aout);
+ owner->mixer_format = owner->input_format;
+ if (aout_OutputNew (aout, &owner->mixer_format))
+ owner->mixer_format.i_format = 0;
+ aout_volume_SetFormat (owner->volume,
+ owner->mixer_format.i_format);
+ }
+
+ msg_Dbg (aout, "restarting filters...");
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
- if (aout_OutputNew (aout, &owner->input_format))
+ if (owner->mixer_format.i_format)
{
- input->b_error = true;
- return; /* we are officially screwed */
+ owner->filters = aout_FiltersNew (aout, &owner->input_format,
+ &owner->mixer_format,
+ &owner->request_vout);
+ if (owner->filters == NULL)
+ {
+ aout_OutputDelete (aout);
+ owner->mixer_format.i_format = 0;
+ }
}
- owner->volume.mixer = aout_MixerNew (aout,
- owner->mixer_format.i_format);
+ /* TODO: This would be a good time to call clean up any video output
+ * left over by an audio visualization:
+ input_resource_TerminatVout(MAGIC HERE); */
}
-
- if (aout_InputNew (aout, &owner->input_format, &owner->mixer_format, input,
- &input->request_vout))
- assert (input->b_error);
- else
- assert (!input->b_error);
+ return (owner->mixer_format.i_format) ? 0 : -1;
}
/**
* Marks the audio output for restart, to update any parameter of the output
* plug-in (e.g. output device or channel mapping).
*/
-void aout_RequestRestart (audio_output_t *aout)
+void aout_RequestRestart (audio_output_t *aout, unsigned mode)
{
aout_owner_t *owner = aout_owner (aout);
-
- /* DO NOT remove AOUT_RESTART_INPUT. You need to change the atomic ops. */
- vlc_atomic_set (&owner->restart, AOUT_RESTART_OUTPUT|AOUT_RESTART_INPUT);
+ atomic_fetch_or (&owner->restart, mode);
+ msg_Dbg (aout, "restart requested (%u)", mode);
}
-/**
- * This function will safely mark aout input to be restarted as soon as
- * possible to take configuration changes into account
+/*
+ * Buffer management
*/
-void aout_InputRequestRestart (audio_output_t *aout)
+
+static void aout_StopResampling (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
- vlc_atomic_compare_swap (&owner->restart, 0, AOUT_RESTART_INPUT);
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ aout_FiltersAdjustResampling (owner->filters, 0);
}
+static void aout_DecSilence (audio_output_t *aout, mtime_t length, mtime_t pts)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ const audio_sample_format_t *fmt = &owner->mixer_format;
+ size_t frames = (fmt->i_rate * length) / CLOCK_FREQ;
+ block_t *block;
-/*
- * Buffer management
- */
+ if (AOUT_FMT_SPDIF(fmt))
+ block = block_Alloc (4 * frames);
+ else
+ block = block_Alloc (frames * fmt->i_bytes_per_frame);
+ if (unlikely(block == NULL))
+ return; /* uho! */
+
+ msg_Dbg (aout, "inserting %zu zeroes", frames);
+ memset (block->p_buffer, 0, block->i_buffer);
+ block->i_nb_samples = frames;
+ block->i_pts = pts;
+ block->i_dts = pts;
+ block->i_length = length;
+ aout_OutputPlay (aout, block);
+}
-/*****************************************************************************
- * aout_DecNewBuffer : ask for a new empty buffer
- *****************************************************************************/
-block_t *aout_DecNewBuffer (audio_output_t *aout, size_t samples)
+static void aout_DecSynchronize (audio_output_t *aout, mtime_t dec_pts,
+ int input_rate)
{
- /* NOTE: the caller is responsible for serializing input change */
aout_owner_t *owner = aout_owner (aout);
+ mtime_t drift;
+
+ /**
+ * Depending on the drift between the actual and intended playback times,
+ * the audio core may ignore the drift, trigger upsampling or downsampling,
+ * insert silence or even discard samples.
+ * Future VLC versions may instead adjust the input rate.
+ *
+ * The audio output plugin is responsible for estimating its actual
+ * playback time, or rather the estimated time when the next sample will
+ * be played. (The actual playback time is always the current time, that is
+ * to say mdate(). It is not an useful statistic.)
+ *
+ * Most audio output plugins can estimate the delay until playback of
+ * the next sample to be written to the buffer, or equally the time until
+ * all samples in the buffer will have been played. Then:
+ * pts = mdate() + delay
+ */
+ if (aout_OutputTimeGet (aout, &drift) != 0)
+ return; /* nothing can be done if timing is unknown */
+ drift += mdate () - dec_pts;
+
+ /* Late audio output.
+ * This can happen due to insufficient caching, scheduling jitter
+ * or bug in the decoder. Ideally, the output would seek backward. But that
+ * is not portable, not supported by some hardware and often unsafe/buggy
+ * where supported. The other alternative is to flush the buffers
+ * completely. */
+ if (drift > (owner->sync.discontinuity ? 0
+ : +3 * input_rate * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT))
+ {
+ if (!owner->sync.discontinuity)
+ msg_Warn (aout, "playback way too late (%"PRId64"): "
+ "flushing buffers", drift);
+ else
+ msg_Dbg (aout, "playback too late (%"PRId64"): "
+ "flushing buffers", drift);
+ aout_OutputFlush (aout, false);
+
+ aout_StopResampling (aout);
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.discontinuity = true;
+
+ /* Now the output might be too early... Recheck. */
+ if (aout_OutputTimeGet (aout, &drift) != 0)
+ return; /* nothing can be done if timing is unknown */
+ drift += mdate () - dec_pts;
+ }
- size_t length = samples * owner->input_format.i_bytes_per_frame
- / owner->input_format.i_frame_length;
- block_t *block = block_Alloc( length );
- if( likely(block != NULL) )
+ /* Early audio output.
+ * This is rare except at startup when the buffers are still empty. */
+ if (drift < (owner->sync.discontinuity ? 0
+ : -3 * input_rate * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT))
{
- block->i_nb_samples = samples;
- block->i_pts = block->i_length = 0;
+ if (!owner->sync.discontinuity)
+ msg_Warn (aout, "playback way too early (%"PRId64"): "
+ "playing silence", drift);
+ aout_DecSilence (aout, -drift, dec_pts);
+
+ aout_StopResampling (aout);
+ owner->sync.discontinuity = true;
+ drift = 0;
}
- return block;
-}
-/*****************************************************************************
- * aout_DecDeleteBuffer : destroy an undecoded buffer
- *****************************************************************************/
-void aout_DecDeleteBuffer (audio_output_t *aout, block_t *block)
-{
- (void) aout;
- aout_BufferFree (block);
+ /* Resampling */
+ if (drift > +AOUT_MAX_PTS_DELAY
+ && owner->sync.resamp_type != AOUT_RESAMPLING_UP)
+ {
+ msg_Warn (aout, "playback too late (%"PRId64"): up-sampling",
+ drift);
+ owner->sync.resamp_type = AOUT_RESAMPLING_UP;
+ owner->sync.resamp_start_drift = +drift;
+ }
+ if (drift < -AOUT_MAX_PTS_ADVANCE
+ && owner->sync.resamp_type != AOUT_RESAMPLING_DOWN)
+ {
+ msg_Warn (aout, "playback too early (%"PRId64"): down-sampling",
+ drift);
+ owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
+ owner->sync.resamp_start_drift = -drift;
+ }
+
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
+ return; /* Everything is fine. Nothing to do. */
+
+ if (llabs (drift) > 2 * owner->sync.resamp_start_drift)
+ { /* If the drift is ever increasing, then something is seriously wrong.
+ * Cease resampling and hope for the best. */
+ msg_Warn (aout, "timing screwed (drift: %"PRId64" us): "
+ "stopping resampling", drift);
+ aout_StopResampling (aout);
+ return;
+ }
+
+ /* Resampling has been triggered earlier. This checks if it needs to be
+ * increased or decreased. Resampling rate changes must be kept slow for
+ * the comfort of listeners. */
+ int adj = (owner->sync.resamp_type == AOUT_RESAMPLING_UP) ? +2 : -2;
+
+ if (2 * llabs (drift) <= owner->sync.resamp_start_drift)
+ /* If the drift has been reduced from more than half its initial
+ * value, then it is time to switch back the resampling direction. */
+ adj *= -1;
+
+ if (!aout_FiltersAdjustResampling (owner->filters, adj))
+ { /* Everything is back to normal: stop resampling. */
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ msg_Dbg (aout, "resampling stopped (drift: %"PRId64" us)", drift);
+ }
}
/*****************************************************************************
* aout_DecPlay : filter & mix the decoded buffer
*****************************************************************************/
-int aout_DecPlay (audio_output_t *p_aout, block_t *p_buffer, int i_input_rate)
+int aout_DecPlay (audio_output_t *aout, block_t *block, int 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
- / owner->input_format.i_rate;
+ aout_owner_t *owner = aout_owner (aout);
- aout_lock( p_aout );
- if( p_input->b_error )
- {
- aout_unlock( p_aout );
- aout_BufferFree( p_buffer );
- return -1;
+ assert (input_rate >= INPUT_RATE_DEFAULT / AOUT_MAX_INPUT_RATE);
+ assert (input_rate <= INPUT_RATE_DEFAULT * AOUT_MAX_INPUT_RATE);
+ assert (block->i_pts >= VLC_TS_0);
+
+ block->i_length = CLOCK_FREQ * block->i_nb_samples
+ / owner->input_format.i_rate;
+
+ aout_OutputLock (aout);
+ if (unlikely(aout_CheckReady (aout)))
+ goto drop; /* Pipeline is unrecoverably broken :-( */
+
+ const mtime_t now = mdate (), advance = block->i_pts - now;
+ if (advance < -AOUT_MAX_PTS_DELAY)
+ { /* Late buffer can be caused by bugs in the decoder, by scheduling
+ * latency spikes (excessive load, SIGSTOP, etc.) or if buffering is
+ * insufficient. We assume the PTS is wrong and play the buffer anyway:
+ * Hopefully video has encountered a similar PTS problem as audio. */
+ msg_Warn (aout, "buffer too late (%"PRId64" us): dropped", advance);
+ goto drop;
}
+ if (advance > AOUT_MAX_ADVANCE_TIME)
+ { /* Early buffers can only be caused by bugs in the decoder. */
+ msg_Err (aout, "buffer too early (%"PRId64" us): dropped", advance);
+ goto drop;
+ }
+ if (block->i_flags & BLOCK_FLAG_DISCONTINUITY)
+ owner->sync.discontinuity = true;
- aout_CheckRestart( p_aout );
-
- /* 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);
+ block = aout_FiltersPlay (owner->filters, block, input_rate);
+ if (block == NULL)
+ goto lost;
- /* Mixer */
- float amp = owner->volume.multiplier
- * aout_InputGetMultiplier (p_input);
- aout_MixerRun (owner->volume.mixer, p_buffer, amp);
+ /* Software volume */
+ aout_volume_Amplify (owner->volume, block);
- /* Output */
- aout_OutputPlay( p_aout, p_buffer );
- }
+ /* Drift correction */
+ aout_DecSynchronize (aout, block->i_pts, input_rate);
- aout_unlock( p_aout );
+ /* Output */
+ owner->sync.end = block->i_pts + block->i_length + 1;
+ owner->sync.discontinuity = false;
+ aout_OutputPlay (aout, block);
+out:
+ aout_OutputUnlock (aout);
return 0;
+drop:
+ owner->sync.discontinuity = true;
+ block_Release (block);
+lost:
+ atomic_fetch_add(&owner->buffers_lost, 1);
+ goto out;
}
int aout_DecGetResetLost (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
- aout_input_t *input = owner->input;
- int val;
-
- aout_lock (aout);
- val = input->i_buffer_lost;
- input->i_buffer_lost = 0;
- aout_unlock (aout);
-
- return val;
+ return atomic_exchange(&owner->buffers_lost, 0);
}
void aout_DecChangePause (audio_output_t *aout, bool paused, mtime_t date)
{
aout_owner_t *owner = aout_owner (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);
+ aout_OutputLock (aout);
+ if (owner->sync.end != VLC_TS_INVALID)
+ {
+ if (paused)
+ owner->sync.end -= date;
+ else
+ owner->sync.end += date;
+ }
+ if (owner->mixer_format.i_format)
+ aout_OutputPause (aout, paused, date);
+ aout_OutputUnlock (aout);
}
-void aout_DecFlush (audio_output_t *aout)
+void aout_DecFlush (audio_output_t *aout, bool wait)
{
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);
+ aout_OutputLock (aout);
+ owner->sync.end = VLC_TS_INVALID;
+ if (owner->mixer_format.i_format)
+ aout_OutputFlush (aout, wait);
+ aout_OutputUnlock (aout);
}
bool aout_DecIsEmpty (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
- mtime_t end_date;
-
- 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);
- }
+ mtime_t now = mdate ();
+ bool empty = true;
+
+ aout_OutputLock (aout);
+ if (owner->sync.end != VLC_TS_INVALID)
+ empty = owner->sync.end <= now;
+ if (empty && owner->mixer_format.i_format)
+ /* The last PTS has elapsed already. So the underlying audio output
+ * buffer should be empty or almost. Thus draining should be fast
+ * and will not block the caller too long. */
+ aout_OutputFlush (aout, true);
+ aout_OutputUnlock (aout);
+ return empty;
}
projects / vlc / blobdiff