mirror of
https://github.com/jojo61/vdr-plugin-softhdcuvid.git
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cf1a661c7b
- remove libavresample from source, deprecated since 2017 - make ALSA and libswresample mandatory for now
2458 lines
75 KiB
C
2458 lines
75 KiB
C
///
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/// @file audio.c @brief Audio module
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///
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/// Copyright (c) 2009 - 2014 by Johns. All Rights Reserved.
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///
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/// Contributor(s):
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///
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/// License: AGPLv3
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///
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/// This program is free software: you can redistribute it and/or modify
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/// it under the terms of the GNU Affero General Public License as
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/// published by the Free Software Foundation, either version 3 of the
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/// License.
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///
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/// This program is distributed in the hope that it will be useful,
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/// but WITHOUT ANY WARRANTY; without even the implied warranty of
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/// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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/// GNU Affero General Public License for more details.
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///
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/// $Id: 77fa65030b179e78c13d0bf69a7cc417dae89e1a $
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//////////////////////////////////////////////////////////////////////////////
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///
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/// @defgroup Audio The audio module.
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///
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/// This module contains all audio output functions.
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///
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/// ALSA PCM/Mixer api is supported.
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/// @see http://www.alsa-project.org/alsa-doc/alsa-lib
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///
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/// @note alsa async playback is broken, don't use it!
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///
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///
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/// @todo FIXME: there can be problems with little/big endian.
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///
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#ifdef DEBUG
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#undef DEBUG
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#endif
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#define USE_AUDIO_THREAD ///< use thread for audio playback
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#define USE_AUDIO_MIXER ///< use audio module mixer
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#include <inttypes.h>
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#include <math.h>
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#include <sched.h>
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#include <stdint.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/prctl.h>
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#include <libintl.h>
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#define _(str) gettext(str) ///< gettext shortcut
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#define _N(str) str ///< gettext_noop shortcut
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#include <alsa/asoundlib.h>
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#ifdef USE_AUDIO_THREAD
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#include <pthread.h>
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#include <sys/resource.h>
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#include <sys/syscall.h>
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#endif
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#include "iatomic.h" // portable atomic_t
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#include "audio.h"
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#include "misc.h"
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#include "ringbuffer.h"
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//----------------------------------------------------------------------------
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// Declarations
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//----------------------------------------------------------------------------
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/**
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** Audio output module structure and typedef.
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*/
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typedef struct _audio_module_ {
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const char *Name; ///< audio output module name
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int (*const Thread)(void); ///< module thread handler
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void (*const FlushBuffers)(void); ///< flush sample buffers
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int64_t (*const GetDelay)(void); ///< get current audio delay
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void (*const SetVolume)(int); ///< set output volume
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int (*const Setup)(int *, int *, int); ///< setup channels, samplerate
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void (*const Play)(void); ///< play audio
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void (*const Pause)(void); ///< pause audio
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void (*const Init)(void); ///< initialize audio output module
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void (*const Exit)(void); ///< cleanup audio output module
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} AudioModule;
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static const AudioModule NoopModule; ///< forward definition of noop module
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//----------------------------------------------------------------------------
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// Variables
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//----------------------------------------------------------------------------
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char AudioAlsaDriverBroken; ///< disable broken driver message
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char AudioAlsaNoCloseOpen; ///< disable alsa close/open fix
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char AudioAlsaCloseOpenDelay; ///< enable alsa close/open delay fix
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static const char *AudioModuleName; ///< which audio module to use
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/// Selected audio module.
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static const AudioModule *AudioUsedModule = &NoopModule;
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static const char *AudioPCMDevice; ///< PCM device name
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static const char *AudioPassthroughDevice; ///< Passthrough device name
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static char AudioAppendAES; ///< flag automatic append AES
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static const char *AudioMixerDevice; ///< mixer device name
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static const char *AudioMixerChannel; ///< mixer channel name
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static char AudioDoingInit; ///> flag in init, reduce error
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static volatile char AudioRunning; ///< thread running / stopped
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static volatile char AudioPaused; ///< audio paused
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static volatile char AudioVideoIsReady; ///< video ready start early
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static int AudioSkip; ///< skip audio to sync to video
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static const int AudioBytesProSample = 2; ///< number of bytes per sample
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static int AudioBufferTime = 336; ///< audio buffer time in ms
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#ifdef USE_AUDIO_THREAD
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static pthread_t AudioThread; ///< audio play thread
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static pthread_mutex_t AudioMutex; ///< audio condition mutex
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static pthread_cond_t AudioStartCond; ///< condition variable
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static char AudioThreadStop; ///< stop audio thread
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#else
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static const int AudioThread; ///< dummy audio thread
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#endif
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static char AudioSoftVolume; ///< flag use soft volume
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static char AudioNormalize; ///< flag use volume normalize
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static char AudioCompression; ///< flag use compress volume
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static char AudioMute; ///< flag muted
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static int AudioAmplifier; ///< software volume factor
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static int AudioNormalizeFactor; ///< current normalize factor
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static const int AudioMinNormalize = 100; ///< min. normalize factor
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static int AudioMaxNormalize; ///< max. normalize factor
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static int AudioCompressionFactor; ///< current compression factor
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static int AudioMaxCompression; ///< max. compression factor
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static int AudioStereoDescent; ///< volume descent for stereo
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static int AudioVolume; ///< current volume (0 .. 1000)
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extern int VideoAudioDelay; ///< import audio/video delay
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/// default ring buffer size ~2s 8ch 16bit (3 * 5 * 7 * 8)
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static const unsigned AudioRingBufferSize = 3 * 5 * 7 * 8 * 2 * 1000;
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static int AudioChannelsInHw[9]; ///< table which channels are supported
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enum _audio_rates { ///< sample rates enumeration
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// HW: 32000 44100 48000 88200 96000 176400 192000
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// Audio32000, ///< 32.0Khz
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Audio44100, ///< 44.1Khz
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Audio48000, ///< 48.0Khz
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// Audio88200, ///< 88.2Khz
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// Audio96000, ///< 96.0Khz
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// Audio176400, ///< 176.4Khz
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Audio192000, ///< 192.0Khz
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AudioRatesMax ///< max index
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};
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/// table which rates are supported
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static int AudioRatesInHw[AudioRatesMax];
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/// input to hardware channel matrix
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static int AudioChannelMatrix[AudioRatesMax][9];
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/// rates tables (must be sorted by frequency)
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static const unsigned AudioRatesTable[AudioRatesMax] = {44100, 48000, 192000};
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//----------------------------------------------------------------------------
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// filter
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//----------------------------------------------------------------------------
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static const int AudioNormSamples = 4096; ///< number of samples
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#define AudioNormMaxIndex 128 ///< number of average values
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/// average of n last sample blocks
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static uint32_t AudioNormAverage[AudioNormMaxIndex];
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static int AudioNormIndex; ///< index into average table
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static int AudioNormReady; ///< index counter
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static int AudioNormCounter; ///< sample counter
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/**
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** Audio normalizer.
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**
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** @param samples sample buffer
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** @param count number of bytes in sample buffer
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*/
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static void AudioNormalizer(int16_t *samples, int count) {
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int i;
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int l;
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int n;
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uint32_t avg;
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int factor;
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int16_t *data;
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// average samples
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l = count / AudioBytesProSample;
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data = samples;
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do {
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n = l;
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if (AudioNormCounter + n > AudioNormSamples) {
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n = AudioNormSamples - AudioNormCounter;
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}
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avg = AudioNormAverage[AudioNormIndex];
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for (i = 0; i < n; ++i) {
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int t;
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t = data[i];
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avg += (t * t) / AudioNormSamples;
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}
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AudioNormAverage[AudioNormIndex] = avg;
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AudioNormCounter += n;
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if (AudioNormCounter >= AudioNormSamples) {
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if (AudioNormReady < AudioNormMaxIndex) {
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AudioNormReady++;
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} else {
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avg = 0;
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for (i = 0; i < AudioNormMaxIndex; ++i) {
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avg += AudioNormAverage[i] / AudioNormMaxIndex;
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}
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// calculate normalize factor
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if (avg > 0) {
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factor = ((INT16_MAX / 8) * 1000U) / (uint32_t)sqrt(avg);
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// smooth normalize
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AudioNormalizeFactor = (AudioNormalizeFactor * 500 + factor * 500) / 1000;
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if (AudioNormalizeFactor < AudioMinNormalize) {
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AudioNormalizeFactor = AudioMinNormalize;
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}
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if (AudioNormalizeFactor > AudioMaxNormalize) {
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AudioNormalizeFactor = AudioMaxNormalize;
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}
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} else {
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factor = 1000;
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}
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Debug(4, "audio/noramlize: avg %8d, fac=%6.3f, norm=%6.3f\n", avg, factor / 1000.0,
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AudioNormalizeFactor / 1000.0);
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}
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AudioNormIndex = (AudioNormIndex + 1) % AudioNormMaxIndex;
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AudioNormCounter = 0;
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AudioNormAverage[AudioNormIndex] = 0U;
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}
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data += n;
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l -= n;
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} while (l > 0);
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// apply normalize factor
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for (i = 0; i < count / AudioBytesProSample; ++i) {
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int t;
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t = (samples[i] * AudioNormalizeFactor) / 1000;
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if (t < INT16_MIN) {
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t = INT16_MIN;
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} else if (t > INT16_MAX) {
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t = INT16_MAX;
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}
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samples[i] = t;
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}
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}
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/**
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** Reset normalizer.
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*/
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static void AudioResetNormalizer(void) {
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int i;
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AudioNormCounter = 0;
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AudioNormReady = 0;
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for (i = 0; i < AudioNormMaxIndex; ++i) {
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AudioNormAverage[i] = 0U;
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}
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AudioNormalizeFactor = 1000;
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}
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/**
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** Audio compression.
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**
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** @param samples sample buffer
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** @param count number of bytes in sample buffer
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*/
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static void AudioCompressor(int16_t *samples, int count) {
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int max_sample;
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int i;
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int factor;
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// find loudest sample
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max_sample = 0;
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for (i = 0; i < count / AudioBytesProSample; ++i) {
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int t;
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t = abs(samples[i]);
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if (t > max_sample) {
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max_sample = t;
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}
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}
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// calculate compression factor
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if (max_sample > 0) {
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factor = (INT16_MAX * 1000) / max_sample;
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// smooth compression (FIXME: make configurable?)
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AudioCompressionFactor = (AudioCompressionFactor * 950 + factor * 50) / 1000;
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if (AudioCompressionFactor > factor) {
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AudioCompressionFactor = factor; // no clipping
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}
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if (AudioCompressionFactor > AudioMaxCompression) {
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AudioCompressionFactor = AudioMaxCompression;
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}
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} else {
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return; // silent nothing todo
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}
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Debug(4, "audio/compress: max %5d, fac=%6.3f, com=%6.3f\n", max_sample, factor / 1000.0,
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AudioCompressionFactor / 1000.0);
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// apply compression factor
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for (i = 0; i < count / AudioBytesProSample; ++i) {
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int t;
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t = (samples[i] * AudioCompressionFactor) / 1000;
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if (t < INT16_MIN) {
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t = INT16_MIN;
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} else if (t > INT16_MAX) {
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t = INT16_MAX;
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}
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samples[i] = t;
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}
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}
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/**
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** Reset compressor.
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*/
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static void AudioResetCompressor(void) {
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AudioCompressionFactor = 2000;
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if (AudioCompressionFactor > AudioMaxCompression) {
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AudioCompressionFactor = AudioMaxCompression;
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}
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}
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/**
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** Audio software amplifier.
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**
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** @param samples sample buffer
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** @param count number of bytes in sample buffer
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**
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** @todo FIXME: this does hard clipping
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*/
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static void AudioSoftAmplifier(int16_t *samples, int count) {
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int i;
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// silence
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if (AudioMute || !AudioAmplifier) {
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memset(samples, 0, count);
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return;
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}
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for (i = 0; i < count / AudioBytesProSample; ++i) {
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int t;
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t = (samples[i] * AudioAmplifier) / 1000;
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if (t < INT16_MIN) {
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t = INT16_MIN;
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} else if (t > INT16_MAX) {
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t = INT16_MAX;
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}
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samples[i] = t;
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}
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}
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#ifdef USE_AUDIO_MIXER
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/**
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** Upmix mono to stereo.
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**
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** @param in input sample buffer
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** @param frames number of frames in sample buffer
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** @param out output sample buffer
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*/
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static void AudioMono2Stereo(const int16_t *in, int frames, int16_t *out) {
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int i;
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for (i = 0; i < frames; ++i) {
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int t;
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t = in[i];
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out[i * 2 + 0] = t;
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out[i * 2 + 1] = t;
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}
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}
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/**
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** Downmix stereo to mono.
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**
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** @param in input sample buffer
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** @param frames number of frames in sample buffer
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** @param out output sample buffer
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*/
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static void AudioStereo2Mono(const int16_t *in, int frames, int16_t *out) {
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int i;
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for (i = 0; i < frames; i += 2) {
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out[i / 2] = (in[i + 0] + in[i + 1]) / 2;
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}
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}
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/**
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** Downmix surround to stereo.
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**
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** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
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** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
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** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
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**
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** @param in input sample buffer
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** @param in_chan nr. of input channels
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** @param frames number of frames in sample buffer
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** @param out output sample buffer
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*/
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static void AudioSurround2Stereo(const int16_t *in, int in_chan, int frames, int16_t *out) {
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while (frames--) {
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int l;
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int r;
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switch (in_chan) {
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case 3: // stereo or surround? =>stereo
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l = in[0] * 600; // L
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r = in[1] * 600; // R
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l += in[2] * 400; // C
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r += in[2] * 400;
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break;
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case 4: // quad or surround? =>quad
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l = in[0] * 600; // L
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r = in[1] * 600; // R
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l += in[2] * 400; // Ls
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r += in[3] * 400; // Rs
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break;
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case 5: // 5.0
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l = in[0] * 500; // L
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r = in[1] * 500; // R
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l += in[2] * 200; // Ls
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r += in[3] * 200; // Rs
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l += in[4] * 300; // C
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r += in[4] * 300;
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break;
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case 6: // 5.1
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l = in[0] * 400; // L
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r = in[1] * 400; // R
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l += in[2] * 200; // Ls
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r += in[3] * 200; // Rs
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l += in[4] * 300; // C
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r += in[4] * 300;
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l += in[5] * 100; // LFE
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r += in[5] * 100;
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break;
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case 7: // 7.0
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l = in[0] * 400; // L
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r = in[1] * 400; // R
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l += in[2] * 200; // Ls
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r += in[3] * 200; // Rs
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l += in[4] * 300; // C
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r += in[4] * 300;
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l += in[5] * 100; // RL
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r += in[6] * 100; // RR
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break;
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case 8: // 7.1
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l = in[0] * 400; // L
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r = in[1] * 400; // R
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l += in[2] * 150; // Ls
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r += in[3] * 150; // Rs
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l += in[4] * 250; // C
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r += in[4] * 250;
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l += in[5] * 100; // LFE
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r += in[5] * 100;
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l += in[6] * 100; // RL
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r += in[7] * 100; // RR
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break;
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default:
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abort();
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}
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in += in_chan;
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out[0] = l / 1000;
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out[1] = r / 1000;
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out += 2;
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}
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}
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/**
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** Upmix @a in_chan channels to @a out_chan.
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**
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** @param in input sample buffer
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** @param in_chan nr. of input channels
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** @param frames number of frames in sample buffer
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** @param out output sample buffer
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** @param out_chan nr. of output channels
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*/
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static void AudioUpmix(const int16_t *in, int in_chan, int frames, int16_t *out, int out_chan) {
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while (frames--) {
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int i;
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for (i = 0; i < in_chan; ++i) { // copy existing channels
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*out++ = *in++;
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}
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for (; i < out_chan; ++i) { // silents missing channels
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*out++ = 0;
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}
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}
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}
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/**
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** Resample ffmpeg sample format to hardware format.
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**
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** FIXME: use libswresample for this and move it to codec.
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** FIXME: ffmpeg to alsa conversion is already done in codec.c.
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**
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** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
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** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
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** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
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**
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** @param in input sample buffer
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** @param in_chan nr. of input channels
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** @param frames number of frames in sample buffer
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** @param out output sample buffer
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** @param out_chan nr. of output channels
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*/
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static void AudioResample(const int16_t *in, int in_chan, int frames, int16_t *out, int out_chan) {
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switch (in_chan * 8 + out_chan) {
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case 1 * 8 + 1:
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case 2 * 8 + 2:
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case 3 * 8 + 3:
|
|
case 4 * 8 + 4:
|
|
case 5 * 8 + 5:
|
|
case 6 * 8 + 6:
|
|
case 7 * 8 + 7:
|
|
case 8 * 8 + 8: // input = output channels
|
|
memcpy(out, in, frames * in_chan * AudioBytesProSample);
|
|
break;
|
|
case 2 * 8 + 1:
|
|
AudioStereo2Mono(in, frames, out);
|
|
break;
|
|
case 1 * 8 + 2:
|
|
AudioMono2Stereo(in, frames, out);
|
|
break;
|
|
case 3 * 8 + 2:
|
|
case 4 * 8 + 2:
|
|
case 5 * 8 + 2:
|
|
case 6 * 8 + 2:
|
|
case 7 * 8 + 2:
|
|
case 8 * 8 + 2:
|
|
AudioSurround2Stereo(in, in_chan, frames, out);
|
|
break;
|
|
case 5 * 8 + 6:
|
|
case 3 * 8 + 8:
|
|
case 5 * 8 + 8:
|
|
case 6 * 8 + 8:
|
|
AudioUpmix(in, in_chan, frames, out, out_chan);
|
|
break;
|
|
|
|
default:
|
|
Error("audio: unsupported %d -> %d channels resample\n", in_chan, out_chan);
|
|
// play silence
|
|
memset(out, 0, frames * out_chan * AudioBytesProSample);
|
|
break;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
// ring buffer
|
|
//----------------------------------------------------------------------------
|
|
|
|
#define AUDIO_RING_MAX 8 ///< number of audio ring buffers
|
|
|
|
/**
|
|
** Audio ring buffer.
|
|
*/
|
|
typedef struct _audio_ring_ring_ {
|
|
char FlushBuffers; ///< flag: flush buffers
|
|
char Passthrough; ///< flag: use pass-through (AC-3, ...)
|
|
int16_t PacketSize; ///< packet size
|
|
unsigned HwSampleRate; ///< hardware sample rate in Hz
|
|
unsigned HwChannels; ///< hardware number of channels
|
|
unsigned InSampleRate; ///< input sample rate in Hz
|
|
unsigned InChannels; ///< input number of channels
|
|
int64_t PTS; ///< pts clock
|
|
RingBuffer *RingBuffer; ///< sample ring buffer
|
|
} AudioRingRing;
|
|
|
|
/// ring of audio ring buffers
|
|
static AudioRingRing AudioRing[AUDIO_RING_MAX];
|
|
static int AudioRingWrite; ///< audio ring write pointer
|
|
static int AudioRingRead; ///< audio ring read pointer
|
|
static atomic_t AudioRingFilled; ///< how many of the ring is used
|
|
static unsigned AudioStartThreshold; ///< start play, if filled
|
|
|
|
/**
|
|
** Add sample-rate, number of channels change to ring.
|
|
**
|
|
** @param sample_rate sample-rate frequency
|
|
** @param channels number of channels
|
|
** @param passthrough use /pass-through (AC-3, ...) device
|
|
**
|
|
** @retval -1 error
|
|
** @retval 0 okay
|
|
**
|
|
** @note this function shouldn't fail. Checks are done during AudoInit.
|
|
*/
|
|
static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough) {
|
|
unsigned u;
|
|
|
|
// search supported sample-rates
|
|
for (u = 0; u < AudioRatesMax; ++u) {
|
|
if (AudioRatesTable[u] == sample_rate) {
|
|
goto found;
|
|
}
|
|
if (AudioRatesTable[u] > sample_rate) {
|
|
break;
|
|
}
|
|
}
|
|
Error(_("audio: %dHz sample-rate unsupported\n"), sample_rate);
|
|
return -1; // unsupported sample-rate
|
|
|
|
found:
|
|
if (!AudioChannelMatrix[u][channels]) {
|
|
Error(_("audio: %d channels unsupported\n"), channels);
|
|
return -1; // unsupported nr. of channels
|
|
}
|
|
|
|
if (atomic_read(&AudioRingFilled) == AUDIO_RING_MAX) { // no free slot
|
|
// FIXME: can wait for ring buffer empty
|
|
Error(_("audio: out of ring buffers\n"));
|
|
return -1;
|
|
}
|
|
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
|
|
|
|
AudioRing[AudioRingWrite].FlushBuffers = 0;
|
|
AudioRing[AudioRingWrite].Passthrough = passthrough;
|
|
AudioRing[AudioRingWrite].PacketSize = 0;
|
|
AudioRing[AudioRingWrite].InSampleRate = sample_rate;
|
|
AudioRing[AudioRingWrite].InChannels = channels;
|
|
AudioRing[AudioRingWrite].HwSampleRate = sample_rate;
|
|
AudioRing[AudioRingWrite].HwChannels = AudioChannelMatrix[u][channels];
|
|
AudioRing[AudioRingWrite].PTS = AV_NOPTS_VALUE;
|
|
RingBufferReset(AudioRing[AudioRingWrite].RingBuffer);
|
|
|
|
Debug(3, "audio: %d ring buffer prepared\n", atomic_read(&AudioRingFilled) + 1);
|
|
|
|
atomic_inc(&AudioRingFilled);
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
if (AudioThread) {
|
|
// tell thread, that there is something todo
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
Debug(3, "Start on AudioRingAdd\n");
|
|
}
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
** Setup audio ring.
|
|
*/
|
|
static void AudioRingInit(void) {
|
|
int i;
|
|
|
|
for (i = 0; i < AUDIO_RING_MAX; ++i) {
|
|
// ~2s 8ch 16bit
|
|
AudioRing[i].RingBuffer = RingBufferNew(AudioRingBufferSize);
|
|
}
|
|
atomic_set(&AudioRingFilled, 0);
|
|
}
|
|
|
|
/**
|
|
** Cleanup audio ring.
|
|
*/
|
|
static void AudioRingExit(void) {
|
|
int i;
|
|
|
|
for (i = 0; i < AUDIO_RING_MAX; ++i) {
|
|
if (AudioRing[i].RingBuffer) {
|
|
RingBufferDel(AudioRing[i].RingBuffer);
|
|
AudioRing[i].RingBuffer = NULL;
|
|
}
|
|
AudioRing[i].HwSampleRate = 0; // checked for valid setup
|
|
AudioRing[i].InSampleRate = 0;
|
|
}
|
|
AudioRingRead = 0;
|
|
AudioRingWrite = 0;
|
|
}
|
|
|
|
//============================================================================
|
|
// A L S A
|
|
//============================================================================
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Alsa variables
|
|
//----------------------------------------------------------------------------
|
|
|
|
static snd_pcm_t *AlsaPCMHandle; ///< alsa pcm handle
|
|
static char AlsaCanPause; ///< hw supports pause
|
|
static int AlsaUseMmap; ///< use mmap
|
|
|
|
static snd_mixer_t *AlsaMixer; ///< alsa mixer handle
|
|
static snd_mixer_elem_t *AlsaMixerElem; ///< alsa pcm mixer element
|
|
static int AlsaRatio; ///< internal -> mixer ratio * 1000
|
|
|
|
//----------------------------------------------------------------------------
|
|
// alsa pcm
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Play samples from ringbuffer.
|
|
**
|
|
** Fill the kernel buffer, as much as possible.
|
|
**
|
|
** @retval 0 ok
|
|
** @retval 1 ring buffer empty
|
|
** @retval -1 underrun error
|
|
*/
|
|
static int AlsaPlayRingbuffer(void) {
|
|
int first;
|
|
|
|
first = 1;
|
|
for (;;) { // loop for ring buffer wrap
|
|
int avail;
|
|
int n;
|
|
int err;
|
|
int frames;
|
|
const void *p;
|
|
|
|
// how many bytes can be written?
|
|
n = snd_pcm_avail_update(AlsaPCMHandle);
|
|
if (n < 0) {
|
|
if (n == -EAGAIN) {
|
|
continue;
|
|
}
|
|
Warning(_("audio/alsa: avail underrun error? '%s'\n"), snd_strerror(n));
|
|
err = snd_pcm_recover(AlsaPCMHandle, n, 0);
|
|
if (err >= 0) {
|
|
continue;
|
|
}
|
|
Error(_("audio/alsa: snd_pcm_avail_update(): %s\n"), snd_strerror(n));
|
|
return -1;
|
|
}
|
|
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, n);
|
|
if (avail < 256) { // too much overhead
|
|
if (first) {
|
|
// happens with broken alsa drivers
|
|
if (AudioThread) {
|
|
if (!AudioAlsaDriverBroken) {
|
|
Error(_("audio/alsa: broken driver %d state '%s'\n"), avail,
|
|
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
|
|
}
|
|
// try to recover
|
|
if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PREPARED) {
|
|
if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio/alsa: snd_pcm_start(): %s\n"), snd_strerror(err));
|
|
}
|
|
}
|
|
usleep(5 * 1000);
|
|
}
|
|
}
|
|
Debug(4, "audio/alsa: break state '%s'\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
|
|
break;
|
|
}
|
|
|
|
n = RingBufferGetReadPointer(AudioRing[AudioRingRead].RingBuffer, &p);
|
|
if (!n) { // ring buffer empty
|
|
if (first) { // only error on first loop
|
|
Debug(4, "audio/alsa: empty buffers %d\n", avail);
|
|
// ring buffer empty
|
|
// AlsaLowWaterMark = 1;
|
|
return 1;
|
|
}
|
|
return 0;
|
|
}
|
|
if (n < avail) { // not enough bytes in ring buffer
|
|
avail = n;
|
|
}
|
|
if (!avail) { // full or buffer empty
|
|
break;
|
|
}
|
|
// muting pass-through AC-3, can produce disturbance
|
|
if (AudioMute || (AudioSoftVolume && !AudioRing[AudioRingRead].Passthrough)) {
|
|
// FIXME: quick&dirty cast
|
|
AudioSoftAmplifier((int16_t *)p, avail);
|
|
// FIXME: if not all are written, we double amplify them
|
|
}
|
|
frames = snd_pcm_bytes_to_frames(AlsaPCMHandle, avail);
|
|
#ifdef DEBUG
|
|
if (avail != snd_pcm_frames_to_bytes(AlsaPCMHandle, frames)) {
|
|
Error(_("audio/alsa: bytes lost -> out of sync\n"));
|
|
}
|
|
#endif
|
|
|
|
for (;;) {
|
|
if (AlsaUseMmap) {
|
|
err = snd_pcm_mmap_writei(AlsaPCMHandle, p, frames);
|
|
} else {
|
|
err = snd_pcm_writei(AlsaPCMHandle, p, frames);
|
|
}
|
|
// Debug(3, "audio/alsa: wrote %d/%d frames\n", err, frames);
|
|
if (err != frames) {
|
|
if (err < 0) {
|
|
if (err == -EAGAIN) {
|
|
continue;
|
|
}
|
|
/*
|
|
if (err == -EBADFD) {
|
|
goto again;
|
|
}
|
|
*/
|
|
Warning(_("audio/alsa: writei underrun error? '%s'\n"), snd_strerror(err));
|
|
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
|
|
if (err >= 0) {
|
|
continue;
|
|
}
|
|
Error(_("audio/alsa: snd_pcm_writei failed: %s\n"), snd_strerror(err));
|
|
return -1;
|
|
}
|
|
// this could happen, if underrun happened
|
|
Warning(_("audio/alsa: not all frames written\n"));
|
|
avail = snd_pcm_frames_to_bytes(AlsaPCMHandle, err);
|
|
}
|
|
break;
|
|
}
|
|
RingBufferReadAdvance(AudioRing[AudioRingRead].RingBuffer, avail);
|
|
first = 0;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
** Flush alsa buffers.
|
|
*/
|
|
static void AlsaFlushBuffers(void) {
|
|
if (AlsaPCMHandle) {
|
|
int err;
|
|
snd_pcm_state_t state;
|
|
|
|
state = snd_pcm_state(AlsaPCMHandle);
|
|
Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state));
|
|
if (state != SND_PCM_STATE_OPEN) {
|
|
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
|
|
}
|
|
// ****ing alsa crash, when in open state here
|
|
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
|
|
//----------------------------------------------------------------------------
|
|
// thread playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Alsa thread
|
|
**
|
|
** Play some samples and return.
|
|
**
|
|
** @retval -1 error
|
|
** @retval 0 underrun
|
|
** @retval 1 running
|
|
*/
|
|
static int AlsaThread(void) {
|
|
int err;
|
|
|
|
if (!AlsaPCMHandle) {
|
|
usleep(24 * 1000);
|
|
return -1;
|
|
}
|
|
for (;;) {
|
|
if (AudioPaused) {
|
|
return 1;
|
|
}
|
|
// wait for space in kernel buffers
|
|
if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) {
|
|
Warning(_("audio/alsa: wait underrun error? '%s'\n"), snd_strerror(err));
|
|
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
|
|
if (err >= 0) {
|
|
continue;
|
|
}
|
|
Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
|
|
usleep(24 * 1000);
|
|
return -1;
|
|
}
|
|
break;
|
|
}
|
|
if (!err || AudioPaused) { // timeout or some commands
|
|
return 1;
|
|
}
|
|
|
|
if ((err = AlsaPlayRingbuffer())) { // empty or error
|
|
snd_pcm_state_t state;
|
|
|
|
if (err < 0) { // underrun error
|
|
return -1;
|
|
}
|
|
|
|
state = snd_pcm_state(AlsaPCMHandle);
|
|
if (state != SND_PCM_STATE_RUNNING) {
|
|
Debug(3, "audio/alsa: stopping play '%s'\n", snd_pcm_state_name(state));
|
|
return 0;
|
|
}
|
|
|
|
usleep(24 * 1000); // let fill/empty the buffers
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Open alsa pcm device.
|
|
**
|
|
** @param passthrough use pass-through (AC-3, ...) device
|
|
*/
|
|
static snd_pcm_t *AlsaOpenPCM(int passthrough) {
|
|
const char *device;
|
|
snd_pcm_t *handle;
|
|
int err;
|
|
|
|
// &&|| hell
|
|
if (!(passthrough && ((device = AudioPassthroughDevice) || (device = getenv("ALSA_PASSTHROUGH_DEVICE")))) &&
|
|
!(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
|
|
device = "default";
|
|
}
|
|
if (!AudioDoingInit) { // reduce blabla during init
|
|
Info(_("audio/alsa: using %sdevice '%s'\n"), passthrough ? "pass-through " : "", device);
|
|
}
|
|
//
|
|
// for AC3 pass-through try to set the non-audio bit, use AES0=6
|
|
//
|
|
if (passthrough && AudioAppendAES) {
|
|
#if 0
|
|
// FIXME: not yet finished
|
|
char *buf;
|
|
const char *s;
|
|
int n;
|
|
|
|
n = strlen(device);
|
|
buf = alloca(n + sizeof(":AES0=6") + 1);
|
|
strcpy(buf, device);
|
|
if (!(s = strchr(buf, ':'))) {
|
|
// no alsa parameters
|
|
strcpy(buf + n, ":AES=6");
|
|
}
|
|
Debug(3, "audio/alsa: try '%s'\n", buf);
|
|
#endif
|
|
}
|
|
// open none blocking; if device is already used, we don't want wait
|
|
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) {
|
|
Error(_("audio/alsa: playback open '%s' error: %s\n"), device, snd_strerror(err));
|
|
return NULL;
|
|
}
|
|
|
|
if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
|
|
Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err));
|
|
}
|
|
return handle;
|
|
}
|
|
|
|
/**
|
|
** Initialize alsa pcm device.
|
|
**
|
|
** @see AudioPCMDevice
|
|
*/
|
|
static void AlsaInitPCM(void) {
|
|
snd_pcm_t *handle;
|
|
snd_pcm_hw_params_t *hw_params;
|
|
int err;
|
|
|
|
if (!(handle = AlsaOpenPCM(0))) {
|
|
return;
|
|
}
|
|
// FIXME: pass-through and pcm out can support different features
|
|
snd_pcm_hw_params_alloca(&hw_params);
|
|
// choose all parameters
|
|
if ((err = snd_pcm_hw_params_any(handle, hw_params)) < 0) {
|
|
Error(_("audio: snd_pcm_hw_params_any: no configurations available: %s\n"), snd_strerror(err));
|
|
}
|
|
AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
|
|
Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
|
|
|
|
AlsaPCMHandle = handle;
|
|
}
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Alsa Mixer
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Set alsa mixer volume (0-1000)
|
|
**
|
|
** @param volume volume (0 .. 1000)
|
|
*/
|
|
static void AlsaSetVolume(int volume) {
|
|
int v;
|
|
|
|
if (AlsaMixer && AlsaMixerElem) {
|
|
v = (volume * AlsaRatio) / (1000 * 1000);
|
|
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 0, v);
|
|
snd_mixer_selem_set_playback_volume(AlsaMixerElem, 1, v);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Initialize alsa mixer.
|
|
*/
|
|
static void AlsaInitMixer(void) {
|
|
const char *device;
|
|
const char *channel;
|
|
snd_mixer_t *alsa_mixer;
|
|
snd_mixer_elem_t *alsa_mixer_elem;
|
|
long alsa_mixer_elem_min;
|
|
long alsa_mixer_elem_max;
|
|
|
|
if (!(device = AudioMixerDevice)) {
|
|
if (!(device = getenv("ALSA_MIXER"))) {
|
|
device = "default";
|
|
}
|
|
}
|
|
if (!(channel = AudioMixerChannel)) {
|
|
if (!(channel = getenv("ALSA_MIXER_CHANNEL"))) {
|
|
channel = "PCM";
|
|
}
|
|
}
|
|
Debug(3, "audio/alsa: mixer %s - %s open\n", device, channel);
|
|
snd_mixer_open(&alsa_mixer, 0);
|
|
if (alsa_mixer && snd_mixer_attach(alsa_mixer, device) >= 0 &&
|
|
snd_mixer_selem_register(alsa_mixer, NULL, NULL) >= 0 && snd_mixer_load(alsa_mixer) >= 0) {
|
|
|
|
const char *const alsa_mixer_elem_name = channel;
|
|
|
|
alsa_mixer_elem = snd_mixer_first_elem(alsa_mixer);
|
|
while (alsa_mixer_elem) {
|
|
const char *name;
|
|
|
|
name = snd_mixer_selem_get_name(alsa_mixer_elem);
|
|
if (!strcasecmp(name, alsa_mixer_elem_name)) {
|
|
snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem, &alsa_mixer_elem_min, &alsa_mixer_elem_max);
|
|
AlsaRatio = 1000 * (alsa_mixer_elem_max - alsa_mixer_elem_min);
|
|
Debug(3, "audio/alsa: PCM mixer found %ld - %ld ratio %d\n", alsa_mixer_elem_min, alsa_mixer_elem_max,
|
|
AlsaRatio);
|
|
break;
|
|
}
|
|
|
|
alsa_mixer_elem = snd_mixer_elem_next(alsa_mixer_elem);
|
|
}
|
|
|
|
AlsaMixer = alsa_mixer;
|
|
AlsaMixerElem = alsa_mixer_elem;
|
|
} else {
|
|
Error(_("audio/alsa: can't open mixer '%s'\n"), device);
|
|
}
|
|
}
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Alsa API
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Get alsa audio delay in time-stamps.
|
|
**
|
|
** @returns audio delay in time-stamps.
|
|
**
|
|
** @todo FIXME: handle the case no audio running
|
|
*/
|
|
static int64_t AlsaGetDelay(void) {
|
|
int err;
|
|
snd_pcm_sframes_t delay;
|
|
int64_t pts;
|
|
|
|
// setup error
|
|
if (!AlsaPCMHandle || !AudioRing[AudioRingRead].HwSampleRate) {
|
|
return 0L;
|
|
}
|
|
// delay in frames in alsa + kernel buffers
|
|
if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) {
|
|
// Debug(3, "audio/alsa: no hw delay\n");
|
|
delay = 0L;
|
|
#ifdef DEBUG
|
|
} else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) {
|
|
// Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay);
|
|
#endif
|
|
}
|
|
Debug(4, "audio/alsa: %ld frames hw delay\n", delay);
|
|
|
|
// delay can be negative, when underrun occur
|
|
if (delay < 0) {
|
|
delay = 0L;
|
|
}
|
|
|
|
pts = ((int64_t)delay * 90 * 1000) / AudioRing[AudioRingRead].HwSampleRate;
|
|
|
|
return pts;
|
|
}
|
|
|
|
/**
|
|
** Setup alsa audio for requested format.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param passthrough use pass-through (AC-3, ...) device
|
|
**
|
|
** @retval 0 everything ok
|
|
** @retval 1 didn't support frequency/channels combination
|
|
** @retval -1 something gone wrong
|
|
**
|
|
** @todo FIXME: remove pointer for freq + channels
|
|
*/
|
|
static int AlsaSetup(int *freq, int *channels, int passthrough) {
|
|
snd_pcm_uframes_t buffer_size;
|
|
snd_pcm_uframes_t period_size;
|
|
int err;
|
|
int delay;
|
|
|
|
if (!AlsaPCMHandle) { // alsa not running yet
|
|
// FIXME: if open fails for fe. pass-through, we never recover
|
|
return -1;
|
|
}
|
|
if (!AudioAlsaNoCloseOpen) { // close+open to fix HDMI no sound bug
|
|
snd_pcm_t *handle;
|
|
|
|
handle = AlsaPCMHandle;
|
|
// no lock needed, thread exit in main loop only
|
|
// Debug(3, "audio: %s [\n", __FUNCTION__);
|
|
AlsaPCMHandle = NULL; // other threads should check handle
|
|
snd_pcm_close(handle);
|
|
if (AudioAlsaCloseOpenDelay) {
|
|
usleep(50 * 1000); // 50ms delay for alsa recovery
|
|
}
|
|
// FIXME: can use multiple retries
|
|
if (!(handle = AlsaOpenPCM(passthrough))) {
|
|
return -1;
|
|
}
|
|
AlsaPCMHandle = handle;
|
|
// Debug(3, "audio: %s ]\n", __FUNCTION__);
|
|
}
|
|
|
|
for (;;) {
|
|
if ((err = snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
|
|
AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED : SND_PCM_ACCESS_RW_INTERLEAVED,
|
|
*channels, *freq, 1, 96 * 1000))) {
|
|
// try reduced buffer size (needed for sunxi)
|
|
// FIXME: alternativ make this configurable
|
|
if ((err =
|
|
snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
|
|
AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED : SND_PCM_ACCESS_RW_INTERLEAVED,
|
|
*channels, *freq, 1, 72 * 1000))) {
|
|
|
|
/*
|
|
if ( err == -EBADFD ) {
|
|
snd_pcm_close(AlsaPCMHandle);
|
|
AlsaPCMHandle = NULL;
|
|
continue;
|
|
}
|
|
*/
|
|
|
|
if (!AudioDoingInit) {
|
|
Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));
|
|
}
|
|
// FIXME: must stop sound, AudioChannels ... invalid
|
|
return -1;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
// this is disabled, no advantages!
|
|
if (0) { // no underruns allowed, play silence
|
|
snd_pcm_sw_params_t *sw_params;
|
|
snd_pcm_uframes_t boundary;
|
|
|
|
snd_pcm_sw_params_alloca(&sw_params);
|
|
err = snd_pcm_sw_params_current(AlsaPCMHandle, sw_params);
|
|
if (err < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_current failed: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err = snd_pcm_sw_params_get_boundary(sw_params, &boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_get_boundary failed: %s\n"), snd_strerror(err));
|
|
}
|
|
Debug(4, "audio/alsa: boundary %lu frames\n", boundary);
|
|
if ((err = snd_pcm_sw_params_set_stop_threshold(AlsaPCMHandle, sw_params, boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err = snd_pcm_sw_params_set_silence_size(AlsaPCMHandle, sw_params, boundary)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params_set_silence_size failed: %s\n"), snd_strerror(err));
|
|
}
|
|
if ((err = snd_pcm_sw_params(AlsaPCMHandle, sw_params)) < 0) {
|
|
Error(_("audio: snd_pcm_sw_params failed: %s\n"), snd_strerror(err));
|
|
}
|
|
}
|
|
// update buffer
|
|
|
|
snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
|
|
Debug(3, "audio/alsa: buffer size %lu %zdms, period size %lu %zdms\n", buffer_size,
|
|
snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size) * 1000 / (*freq * *channels * AudioBytesProSample),
|
|
period_size,
|
|
snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size) * 1000 / (*freq * *channels * AudioBytesProSample));
|
|
Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
|
|
|
|
AudioStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
|
|
// buffer time/delay in ms
|
|
delay = AudioBufferTime;
|
|
if (VideoAudioDelay > 0) {
|
|
delay += VideoAudioDelay / 90;
|
|
}
|
|
if (AudioStartThreshold < (*freq * *channels * AudioBytesProSample * delay) / 1000U) {
|
|
AudioStartThreshold = (*freq * *channels * AudioBytesProSample * delay) / 1000U;
|
|
}
|
|
// no bigger, than 1/3 the buffer
|
|
if (AudioStartThreshold > AudioRingBufferSize / 3) {
|
|
AudioStartThreshold = AudioRingBufferSize / 3;
|
|
}
|
|
if (!AudioDoingInit) {
|
|
Info(_("audio/alsa: start delay %ums\n"),
|
|
(AudioStartThreshold * 1000) / (*freq * *channels * AudioBytesProSample));
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
** Play audio.
|
|
*/
|
|
static void AlsaPlay(void) {
|
|
int err;
|
|
|
|
if (AlsaCanPause) {
|
|
if ((err = snd_pcm_pause(AlsaPCMHandle, 0))) {
|
|
Error(_("audio/alsa: snd_pcm_pause(): %s\n"), snd_strerror(err));
|
|
}
|
|
} else {
|
|
if ((err = snd_pcm_prepare(AlsaPCMHandle)) < 0) {
|
|
Error(_("audio/alsa: snd_pcm_prepare(): %s\n"), snd_strerror(err));
|
|
}
|
|
}
|
|
#ifdef DEBUG
|
|
if (snd_pcm_state(AlsaPCMHandle) == SND_PCM_STATE_PAUSED) {
|
|
Error(_("audio/alsa: still paused\n"));
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/**
|
|
** Pause audio.
|
|
*/
|
|
static void AlsaPause(void) {
|
|
int err;
|
|
|
|
if (AlsaCanPause) {
|
|
if ((err = snd_pcm_pause(AlsaPCMHandle, 1))) {
|
|
Error(_("snd_pcm_pause(): %s\n"), snd_strerror(err));
|
|
}
|
|
} else {
|
|
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
|
|
Error(_("snd_pcm_drop(): %s\n"), snd_strerror(err));
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Empty log callback
|
|
*/
|
|
static void AlsaNoopCallback(__attribute__((unused)) const char *file, __attribute__((unused)) int line,
|
|
__attribute__((unused)) const char *function, __attribute__((unused)) int err,
|
|
__attribute__((unused)) const char *fmt, ...) {}
|
|
|
|
/**
|
|
** Initialize alsa audio output module.
|
|
*/
|
|
static void AlsaInit(void) {
|
|
#ifdef DEBUG
|
|
(void)AlsaNoopCallback;
|
|
#else
|
|
// disable display of alsa error messages
|
|
snd_lib_error_set_handler(AlsaNoopCallback);
|
|
#endif
|
|
|
|
AlsaInitPCM();
|
|
AlsaInitMixer();
|
|
}
|
|
|
|
/**
|
|
** Cleanup alsa audio output module.
|
|
*/
|
|
static void AlsaExit(void) {
|
|
if (AlsaPCMHandle) {
|
|
snd_pcm_close(AlsaPCMHandle);
|
|
AlsaPCMHandle = NULL;
|
|
}
|
|
if (AlsaMixer) {
|
|
snd_mixer_close(AlsaMixer);
|
|
AlsaMixer = NULL;
|
|
AlsaMixerElem = NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Alsa module.
|
|
*/
|
|
static const AudioModule AlsaModule = {
|
|
.Name = "alsa",
|
|
#ifdef USE_AUDIO_THREAD
|
|
.Thread = AlsaThread,
|
|
#endif
|
|
.FlushBuffers = AlsaFlushBuffers,
|
|
.GetDelay = AlsaGetDelay,
|
|
.SetVolume = AlsaSetVolume,
|
|
.Setup = AlsaSetup,
|
|
.Play = AlsaPlay,
|
|
.Pause = AlsaPause,
|
|
.Init = AlsaInit,
|
|
.Exit = AlsaExit,
|
|
};
|
|
|
|
//============================================================================
|
|
// Noop
|
|
//============================================================================
|
|
|
|
/**
|
|
** Get audio delay in time stamps.
|
|
**
|
|
** @returns audio delay in time stamps.
|
|
*/
|
|
static int64_t NoopGetDelay(void) { return 0L; }
|
|
|
|
/**
|
|
** Set mixer volume (0-1000)
|
|
**
|
|
** @param volume volume (0 .. 1000)
|
|
*/
|
|
static void NoopSetVolume(__attribute__((unused)) int volume) {}
|
|
|
|
/**
|
|
** Noop setup.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param passthrough use pass-through (AC-3, ...) device
|
|
*/
|
|
static int NoopSetup(__attribute__((unused)) int *channels, __attribute__((unused)) int *freq,
|
|
__attribute__((unused)) int passthrough) {
|
|
return -1;
|
|
}
|
|
|
|
/**
|
|
** Noop void
|
|
*/
|
|
static void NoopVoid(void) {}
|
|
|
|
/**
|
|
** Noop module.
|
|
*/
|
|
static const AudioModule NoopModule = {
|
|
.Name = "noop",
|
|
.FlushBuffers = NoopVoid,
|
|
.GetDelay = NoopGetDelay,
|
|
.SetVolume = NoopSetVolume,
|
|
.Setup = NoopSetup,
|
|
.Play = NoopVoid,
|
|
.Pause = NoopVoid,
|
|
.Init = NoopVoid,
|
|
.Exit = NoopVoid,
|
|
};
|
|
|
|
//----------------------------------------------------------------------------
|
|
// thread playback
|
|
//----------------------------------------------------------------------------
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
|
|
/**
|
|
** Prepare next ring buffer.
|
|
*/
|
|
static int AudioNextRing(void) {
|
|
int passthrough;
|
|
int sample_rate;
|
|
int channels;
|
|
size_t used;
|
|
|
|
// update audio format
|
|
// not always needed, but check if needed is too complex
|
|
passthrough = AudioRing[AudioRingRead].Passthrough;
|
|
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
|
|
channels = AudioRing[AudioRingRead].HwChannels;
|
|
if (AudioUsedModule->Setup(&sample_rate, &channels, passthrough)) {
|
|
Error(_("audio: can't set channels %d sample-rate %dHz\n"), channels, sample_rate);
|
|
// FIXME: handle error
|
|
AudioRing[AudioRingRead].HwSampleRate = 0;
|
|
AudioRing[AudioRingRead].InSampleRate = 0;
|
|
return -1;
|
|
}
|
|
|
|
AudioSetVolume(AudioVolume); // update channel delta
|
|
AudioResetCompressor();
|
|
AudioResetNormalizer();
|
|
|
|
Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled),
|
|
(RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000) /
|
|
(AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
|
|
|
|
// stop, if not enough in next buffer
|
|
used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer);
|
|
if (AudioStartThreshold * 4 < used || (AudioVideoIsReady && AudioStartThreshold < used)) {
|
|
return 0;
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
/**
|
|
** Audio play thread.
|
|
**
|
|
** @param dummy unused thread argument
|
|
*/
|
|
static void *AudioPlayHandlerThread(void *dummy) {
|
|
Debug(3, "audio: play thread started\n");
|
|
prctl(PR_SET_NAME, "cuvid audio", 0, 0, 0);
|
|
|
|
for (;;) {
|
|
// check if we should stop the thread
|
|
if (AudioThreadStop) {
|
|
Debug(3, "audio: play thread stopped\n");
|
|
return PTHREAD_CANCELED;
|
|
}
|
|
|
|
Debug(3, "audio: wait on start condition\n");
|
|
pthread_mutex_lock(&AudioMutex);
|
|
AudioRunning = 0;
|
|
do {
|
|
pthread_cond_wait(&AudioStartCond, &AudioMutex);
|
|
// cond_wait can return, without signal!
|
|
} while (!AudioRunning);
|
|
pthread_mutex_unlock(&AudioMutex);
|
|
|
|
Debug(
|
|
3, "audio: ----> %dms %d start\n",
|
|
(AudioUsedBytes() * 1000) /
|
|
(!AudioRing[AudioRingWrite].HwSampleRate + !AudioRing[AudioRingWrite].HwChannels +
|
|
AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
|
|
AudioUsedBytes());
|
|
|
|
do {
|
|
int filled;
|
|
int read;
|
|
int flush;
|
|
int err;
|
|
int i;
|
|
|
|
// check if we should stop the thread
|
|
if (AudioThreadStop) {
|
|
Debug(3, "audio: play thread stopped\n");
|
|
return PTHREAD_CANCELED;
|
|
}
|
|
// look if there is a flush command in the queue
|
|
flush = 0;
|
|
filled = atomic_read(&AudioRingFilled);
|
|
read = AudioRingRead;
|
|
i = filled;
|
|
while (i--) {
|
|
read = (read + 1) % AUDIO_RING_MAX;
|
|
if (AudioRing[read].FlushBuffers) {
|
|
AudioRing[read].FlushBuffers = 0;
|
|
AudioRingRead = read;
|
|
// handle all flush in queue
|
|
flush = filled - i;
|
|
}
|
|
}
|
|
|
|
if (flush) {
|
|
Debug(3, "audio: flush %d ring buffer(s)\n", flush);
|
|
AudioUsedModule->FlushBuffers();
|
|
atomic_sub(flush, &AudioRingFilled);
|
|
if (AudioNextRing()) {
|
|
break;
|
|
}
|
|
}
|
|
// try to play some samples
|
|
err = 0;
|
|
if (RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) {
|
|
err = AudioUsedModule->Thread();
|
|
}
|
|
// underrun, check if new ring buffer is available
|
|
if (!err) {
|
|
int passthrough;
|
|
int sample_rate;
|
|
int channels;
|
|
int old_passthrough;
|
|
int old_sample_rate;
|
|
int old_channels;
|
|
|
|
// underrun, and no new ring buffer, goto sleep.
|
|
if (!atomic_read(&AudioRingFilled)) {
|
|
Debug(3, "audio: HandlerThread Underrun with no new data\n");
|
|
break;
|
|
}
|
|
|
|
Debug(3, "audio: next ring buffer\n");
|
|
old_passthrough = AudioRing[AudioRingRead].Passthrough;
|
|
old_sample_rate = AudioRing[AudioRingRead].HwSampleRate;
|
|
old_channels = AudioRing[AudioRingRead].HwChannels;
|
|
|
|
atomic_dec(&AudioRingFilled);
|
|
AudioRingRead = (AudioRingRead + 1) % AUDIO_RING_MAX;
|
|
|
|
passthrough = AudioRing[AudioRingRead].Passthrough;
|
|
sample_rate = AudioRing[AudioRingRead].HwSampleRate;
|
|
channels = AudioRing[AudioRingRead].HwChannels;
|
|
Debug(3, "audio: thread channels %d frequency %dHz %s\n", channels, sample_rate,
|
|
passthrough ? "pass-through" : "");
|
|
// audio config changed?
|
|
if (old_passthrough != passthrough || old_sample_rate != sample_rate || old_channels != channels) {
|
|
// FIXME: wait for buffer drain
|
|
if (AudioNextRing()) {
|
|
Debug(3, "audio: HandlerThread break on nextring");
|
|
break;
|
|
}
|
|
} else {
|
|
AudioResetCompressor();
|
|
AudioResetNormalizer();
|
|
}
|
|
}
|
|
// FIXME: check AudioPaused ...Thread()
|
|
if (AudioPaused) {
|
|
Debug(3, "audio: HandlerThread break on paused");
|
|
break;
|
|
}
|
|
} while (AudioRing[AudioRingRead].HwSampleRate);
|
|
}
|
|
return dummy;
|
|
}
|
|
|
|
/**
|
|
** Initialize audio thread.
|
|
*/
|
|
static void AudioInitThread(void) {
|
|
AudioThreadStop = 0;
|
|
pthread_mutex_init(&AudioMutex, NULL);
|
|
pthread_cond_init(&AudioStartCond, NULL);
|
|
pthread_create(&AudioThread, NULL, AudioPlayHandlerThread, NULL);
|
|
pthread_setname_np(AudioThread, "softhddev audio");
|
|
}
|
|
|
|
/**
|
|
** Cleanup audio thread.
|
|
*/
|
|
static void AudioExitThread(void) {
|
|
void *retval;
|
|
|
|
Debug(3, "audio: %s\n", __FUNCTION__);
|
|
|
|
if (AudioThread) {
|
|
AudioThreadStop = 1;
|
|
AudioRunning = 1; // wakeup thread, if needed
|
|
pthread_cond_signal(&AudioStartCond);
|
|
if (pthread_join(AudioThread, &retval) || retval != PTHREAD_CANCELED) {
|
|
Error(_("audio: can't cancel play thread\n"));
|
|
}
|
|
pthread_cond_destroy(&AudioStartCond);
|
|
pthread_mutex_destroy(&AudioMutex);
|
|
AudioThread = 0;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
//----------------------------------------------------------------------------
|
|
//----------------------------------------------------------------------------
|
|
|
|
/**
|
|
** Table of all audio modules.
|
|
*/
|
|
static const AudioModule *AudioModules[] = {
|
|
&AlsaModule,
|
|
&NoopModule,
|
|
};
|
|
|
|
void AudioDelayms(int delayms) {
|
|
int count;
|
|
unsigned char *p;
|
|
|
|
#ifdef DEBUG
|
|
printf("Try Delay Audio for %d ms Samplerate %d Channels %d bps %d\n", delayms,
|
|
AudioRing[AudioRingWrite].HwSampleRate, AudioRing[AudioRingWrite].HwChannels, AudioBytesProSample);
|
|
#endif
|
|
|
|
count = delayms * AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
|
|
AudioBytesProSample / 1000;
|
|
|
|
if (delayms < 5000 && delayms > 0) { // not more than 5seconds
|
|
p = calloc(1, count);
|
|
RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, p, count);
|
|
free(p);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Place samples in audio output queue.
|
|
**
|
|
** @param samples sample buffer
|
|
** @param count number of bytes in sample buffer
|
|
*/
|
|
void AudioEnqueue(const void *samples, int count) {
|
|
size_t n;
|
|
int16_t *buffer;
|
|
|
|
#ifdef noDEBUG
|
|
static uint32_t last_tick;
|
|
uint32_t tick;
|
|
|
|
tick = GetMsTicks();
|
|
if (tick - last_tick > 101) {
|
|
Debug(3, "audio: enqueue %4d %dms\n", count, tick - last_tick);
|
|
}
|
|
last_tick = tick;
|
|
#endif
|
|
|
|
if (!AudioRing[AudioRingWrite].HwSampleRate) {
|
|
Debug(3, "audio: enqueue not ready\n");
|
|
return; // no setup yet
|
|
}
|
|
// save packet size
|
|
if (!AudioRing[AudioRingWrite].PacketSize) {
|
|
AudioRing[AudioRingWrite].PacketSize = count;
|
|
Debug(3, "audio: a/v packet size %d bytes\n", count);
|
|
}
|
|
// audio sample modification allowed and needed?
|
|
buffer = (void *)samples;
|
|
if (!AudioRing[AudioRingWrite].Passthrough &&
|
|
(AudioCompression || AudioNormalize ||
|
|
AudioRing[AudioRingWrite].InChannels != AudioRing[AudioRingWrite].HwChannels)) {
|
|
int frames;
|
|
|
|
// resample into ring-buffer is too complex in the case of a roundabout
|
|
// just use a temporary buffer
|
|
frames = count / (AudioRing[AudioRingWrite].InChannels * AudioBytesProSample);
|
|
buffer = alloca(frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
|
|
#ifdef USE_AUDIO_MIXER
|
|
// Convert / resample input to hardware format
|
|
AudioResample(samples, AudioRing[AudioRingWrite].InChannels, frames, buffer,
|
|
AudioRing[AudioRingWrite].HwChannels);
|
|
#else
|
|
#ifdef DEBUG
|
|
if (AudioRing[AudioRingWrite].InChannels != AudioRing[AudioRingWrite].HwChannels) {
|
|
Debug(3, "audio: internal failure channels mismatch\n");
|
|
return;
|
|
}
|
|
#endif
|
|
memcpy(buffer, samples, count);
|
|
#endif
|
|
count = frames * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample;
|
|
|
|
if (AudioCompression) { // in place operation
|
|
AudioCompressor(buffer, count);
|
|
}
|
|
if (AudioNormalize) { // in place operation
|
|
AudioNormalizer(buffer, count);
|
|
}
|
|
}
|
|
|
|
n = RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, buffer, count);
|
|
if (n != (size_t)count) {
|
|
Error(_("audio: can't place %d samples in ring buffer\n"), count);
|
|
// too many bytes are lost
|
|
// FIXME: caller checks buffer full.
|
|
// FIXME: should skip more, longer skip, but less often?
|
|
// FIXME: round to channel + sample border
|
|
}
|
|
|
|
if (!AudioRunning) { // check, if we can start the thread
|
|
int skip;
|
|
|
|
n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
|
|
skip = AudioSkip;
|
|
// FIXME: round to packet size
|
|
|
|
Debug(4, "audio: start? %4zdms skip %dms\n",
|
|
(n * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
|
|
AudioBytesProSample),
|
|
(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
|
|
AudioBytesProSample));
|
|
|
|
if (skip) {
|
|
if (n < (unsigned)skip) {
|
|
skip = n;
|
|
}
|
|
AudioSkip -= skip;
|
|
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
|
|
n = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
|
|
}
|
|
// forced start or enough video + audio buffered
|
|
// for some exotic channels * 4 too small
|
|
if (AudioStartThreshold * 4 < n || (AudioVideoIsReady
|
|
// if ((AudioVideoIsReady
|
|
&& AudioStartThreshold < n)) {
|
|
// restart play-back
|
|
// no lock needed, can wakeup next time
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
Debug(3, "Start on AudioEnque Threshold %d n %d\n", AudioStartThreshold, n);
|
|
}
|
|
}
|
|
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
|
|
if (AudioRing[AudioRingWrite].PTS != (int64_t)AV_NOPTS_VALUE) {
|
|
AudioRing[AudioRingWrite].PTS +=
|
|
((int64_t)count * 90 * 1000) /
|
|
(AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Video is ready.
|
|
**
|
|
** @param pts video presentation timestamp
|
|
*/
|
|
void AudioVideoReady(int64_t pts) {
|
|
int64_t audio_pts;
|
|
size_t used;
|
|
|
|
if (pts == (int64_t)AV_NOPTS_VALUE) {
|
|
Debug(3, "audio: a/v start, no valid video\n");
|
|
return;
|
|
}
|
|
// no valid audio known
|
|
if (!AudioRing[AudioRingWrite].HwSampleRate || !AudioRing[AudioRingWrite].HwChannels ||
|
|
AudioRing[AudioRingWrite].PTS == (int64_t)AV_NOPTS_VALUE) {
|
|
Debug(3, "audio: a/v start, no valid audio\n");
|
|
AudioVideoIsReady = 1;
|
|
return;
|
|
}
|
|
// Audio.PTS = next written sample time stamp
|
|
|
|
used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
|
|
audio_pts = AudioRing[AudioRingWrite].PTS -
|
|
(used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
|
|
AudioBytesProSample);
|
|
|
|
Debug(3, "audio: a/v sync buf(%d,%4zdms) %s | %s = %dms %s\n", atomic_read(&AudioRingFilled),
|
|
(used * 1000) /
|
|
(AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
|
|
Timestamp2String(pts), Timestamp2String(audio_pts), (int)(pts - audio_pts) / 90,
|
|
AudioRunning ? "running" : "ready");
|
|
|
|
if (!AudioRunning) {
|
|
int skip;
|
|
|
|
// buffer ~15 video frames
|
|
// FIXME: HDTV can use smaller video buffer
|
|
skip = pts - 0 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay;
|
|
#ifdef DEBUG
|
|
// fprintf(stderr, "a/v-diff %dms a/v-delay %dms skip %dms Audiobuffer
|
|
//%d\n", (int)(pts - audio_pts) / 90, VideoAudioDelay / 90, skip /
|
|
// 90,AudioBufferTime);
|
|
#endif
|
|
// guard against old PTS
|
|
if (skip > 0 && skip < 4000 * 90) {
|
|
skip = (((int64_t)skip * AudioRing[AudioRingWrite].HwSampleRate) / (1000 * 90)) *
|
|
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample;
|
|
// FIXME: round to packet size
|
|
if ((unsigned)skip > used) {
|
|
AudioSkip = skip - used;
|
|
skip = used;
|
|
}
|
|
Debug(3, "audio: sync advance %dms %d/%zd Rest %d\n",
|
|
(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
|
|
AudioBytesProSample),
|
|
skip, used, AudioSkip);
|
|
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);
|
|
|
|
used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
|
|
} else {
|
|
Debug(3, "No audio skip -> should skip %d\n", skip / 90);
|
|
}
|
|
// FIXME: skip<0 we need bigger audio buffer
|
|
// enough video + audio buffered
|
|
if (AudioStartThreshold < used) {
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
Debug(3, "Start on AudioVideoReady\n");
|
|
}
|
|
}
|
|
|
|
AudioVideoIsReady = 1;
|
|
}
|
|
|
|
/**
|
|
** Flush audio buffers.
|
|
*/
|
|
void AudioFlushBuffers(void) {
|
|
int old;
|
|
int i;
|
|
|
|
if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) {
|
|
// wait for space in ring buffer, should never happen
|
|
for (i = 0; i < 24 * 2; ++i) {
|
|
if (atomic_read(&AudioRingFilled) < AUDIO_RING_MAX) {
|
|
break;
|
|
}
|
|
Debug(3, "audio: flush out of ring buffers\n");
|
|
usleep(1 * 1000); // avoid hot polling
|
|
}
|
|
if (atomic_read(&AudioRingFilled) >= AUDIO_RING_MAX) {
|
|
// FIXME: We can set the flush flag in the last wrote ring buffer
|
|
Error(_("audio: flush out of ring buffers\n"));
|
|
return;
|
|
}
|
|
}
|
|
|
|
old = AudioRingWrite;
|
|
AudioRingWrite = (AudioRingWrite + 1) % AUDIO_RING_MAX;
|
|
AudioRing[AudioRingWrite].FlushBuffers = 1;
|
|
AudioRing[AudioRingWrite].Passthrough = AudioRing[old].Passthrough;
|
|
AudioRing[AudioRingWrite].HwSampleRate = AudioRing[old].HwSampleRate;
|
|
AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
|
|
AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
|
|
AudioRing[AudioRingWrite].InChannels = AudioRing[old].InChannels;
|
|
AudioRing[AudioRingWrite].PTS = AV_NOPTS_VALUE;
|
|
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer,
|
|
RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer));
|
|
Debug(3, "audio: reset video ready\n");
|
|
AudioVideoIsReady = 0;
|
|
AudioSkip = 0;
|
|
|
|
atomic_inc(&AudioRingFilled);
|
|
|
|
// FIXME: wait for flush complete needed?
|
|
for (i = 0; i < 24 * 2; ++i) {
|
|
if (!AudioRunning) { // wakeup thread to flush buffers
|
|
AudioRunning = 1;
|
|
pthread_cond_signal(&AudioStartCond);
|
|
Debug(3, "Start on Flush\n");
|
|
}
|
|
// FIXME: waiting on zero isn't correct, but currently works
|
|
if (!atomic_read(&AudioRingFilled)) {
|
|
break;
|
|
}
|
|
usleep(1 * 1000); // avoid hot polling
|
|
}
|
|
Debug(3, "audio: audio flush %dms\n", i);
|
|
}
|
|
|
|
/**
|
|
** Call back to play audio polled.
|
|
*/
|
|
void AudioPoller(void) {
|
|
// FIXME: write poller
|
|
}
|
|
|
|
/**
|
|
** Get free bytes in audio output.
|
|
*/
|
|
int AudioFreeBytes(void) {
|
|
return AudioRing[AudioRingWrite].RingBuffer ? RingBufferFreeBytes(AudioRing[AudioRingWrite].RingBuffer)
|
|
: INT32_MAX;
|
|
}
|
|
|
|
/**
|
|
** Get used bytes in audio output.
|
|
*/
|
|
int AudioUsedBytes(void) {
|
|
// FIXME: not correct, if multiple buffer are in use
|
|
return AudioRing[AudioRingWrite].RingBuffer ? RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) : 0;
|
|
}
|
|
|
|
/**
|
|
** Get audio delay in time stamps.
|
|
**
|
|
** @returns audio delay in time stamps.
|
|
*/
|
|
int64_t AudioGetDelay(void) {
|
|
int64_t pts;
|
|
|
|
if (!AudioRunning) {
|
|
return 0L; // audio not running
|
|
}
|
|
if (!AudioRing[AudioRingRead].HwSampleRate) {
|
|
return 0L; // audio not setup
|
|
}
|
|
if (atomic_read(&AudioRingFilled)) {
|
|
return 0L; // multiple buffers, invalid delay
|
|
}
|
|
pts = AudioUsedModule->GetDelay();
|
|
pts += ((int64_t)RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 90 * 1000) /
|
|
(AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels * AudioBytesProSample);
|
|
Debug(4, "audio: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer),
|
|
pts / 90);
|
|
|
|
return pts;
|
|
}
|
|
|
|
/**
|
|
** Set audio clock base.
|
|
**
|
|
** @param pts audio presentation timestamp
|
|
*/
|
|
void AudioSetClock(int64_t pts) {
|
|
if (AudioRing[AudioRingWrite].PTS != pts) {
|
|
Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioRing[AudioRingWrite].PTS),
|
|
Timestamp2String(pts));
|
|
}
|
|
// printf("Audiosetclock pts %#012" PRIx64 "
|
|
// %d\n",pts,RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer));
|
|
AudioRing[AudioRingWrite].PTS = pts;
|
|
}
|
|
|
|
/**
|
|
** Get current audio clock.
|
|
**
|
|
** @returns the audio clock in time stamps.
|
|
*/
|
|
int64_t AudioGetClock(void) {
|
|
// (cast) needed for the evil gcc
|
|
if (AudioRing[AudioRingRead].PTS != (int64_t)AV_NOPTS_VALUE) {
|
|
int64_t delay;
|
|
|
|
// delay zero, if no valid time stamp
|
|
if ((delay = AudioGetDelay())) {
|
|
if (AudioRing[AudioRingRead].Passthrough) {
|
|
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
|
|
}
|
|
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
|
|
}
|
|
}
|
|
return AV_NOPTS_VALUE;
|
|
}
|
|
|
|
/**
|
|
** Set mixer volume (0-1000)
|
|
**
|
|
** @param volume volume (0 .. 1000)
|
|
*/
|
|
void AudioSetVolume(int volume) {
|
|
AudioVolume = volume;
|
|
AudioMute = !volume;
|
|
// reduce loudness for stereo output
|
|
if (AudioStereoDescent && AudioRing[AudioRingRead].InChannels == 2 && !AudioRing[AudioRingRead].Passthrough) {
|
|
volume -= AudioStereoDescent;
|
|
if (volume < 0) {
|
|
volume = 0;
|
|
} else if (volume > 1000) {
|
|
volume = 1000;
|
|
}
|
|
}
|
|
AudioAmplifier = volume;
|
|
if (!AudioSoftVolume) {
|
|
AudioUsedModule->SetVolume(volume);
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Setup audio for requested format.
|
|
**
|
|
** @param freq sample frequency
|
|
** @param channels number of channels
|
|
** @param passthrough use pass-through (AC-3, ...) device
|
|
**
|
|
** @retval 0 everything ok
|
|
** @retval 1 didn't support frequency/channels combination
|
|
** @retval -1 something gone wrong
|
|
**
|
|
** @todo add support to report best fitting format.
|
|
*/
|
|
int AudioSetup(int *freq, int *channels, int passthrough) {
|
|
Debug(3, "audio: setup channels %d frequency %dHz %s\n", *channels, *freq, passthrough ? "pass-through" : "");
|
|
|
|
// invalid parameter
|
|
if (!freq || !channels || !*freq || !*channels) {
|
|
Debug(3, "audio: bad channels or frequency parameters\n");
|
|
// FIXME: set flag invalid setup
|
|
return -1;
|
|
}
|
|
return AudioRingAdd(*freq, *channels, passthrough);
|
|
}
|
|
|
|
/**
|
|
** Play audio.
|
|
*/
|
|
void AudioPlay(void) {
|
|
if (!AudioPaused) {
|
|
Debug(3, "audio: not paused, check the code\n");
|
|
return;
|
|
}
|
|
Debug(3, "audio: resumed\n");
|
|
AudioPaused = 0;
|
|
AudioEnqueue(NULL, 0); // wakeup thread
|
|
}
|
|
|
|
/**
|
|
** Pause audio.
|
|
*/
|
|
void AudioPause(void) {
|
|
if (AudioPaused) {
|
|
Debug(3, "audio: already paused, check the code\n");
|
|
return;
|
|
}
|
|
Debug(3, "audio: paused\n");
|
|
AudioPaused = 1;
|
|
}
|
|
|
|
/**
|
|
** Set audio buffer time.
|
|
**
|
|
** PES audio packets have a max distance of 300 ms.
|
|
** TS audio packet have a max distance of 100 ms.
|
|
** The period size of the audio buffer is 24 ms.
|
|
** With streamdev sometimes extra +100ms are needed.
|
|
*/
|
|
void AudioSetBufferTime(int delay) {
|
|
if (!delay) {
|
|
delay = 336;
|
|
}
|
|
AudioBufferTime = delay;
|
|
}
|
|
|
|
/**
|
|
** Enable/disable software volume.
|
|
**
|
|
** @param onoff -1 toggle, true turn on, false turn off
|
|
*/
|
|
void AudioSetSoftvol(int onoff) {
|
|
if (onoff < 0) {
|
|
AudioSoftVolume ^= 1;
|
|
} else {
|
|
AudioSoftVolume = onoff;
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Set normalize volume parameters.
|
|
**
|
|
** @param onoff -1 toggle, true turn on, false turn off
|
|
** @param maxfac max. factor of normalize /1000
|
|
*/
|
|
void AudioSetNormalize(int onoff, int maxfac) {
|
|
if (onoff < 0) {
|
|
AudioNormalize ^= 1;
|
|
} else {
|
|
AudioNormalize = onoff;
|
|
}
|
|
AudioMaxNormalize = maxfac;
|
|
}
|
|
|
|
/**
|
|
** Set volume compression parameters.
|
|
**
|
|
** @param onoff -1 toggle, true turn on, false turn off
|
|
** @param maxfac max. factor of compression /1000
|
|
*/
|
|
void AudioSetCompression(int onoff, int maxfac) {
|
|
if (onoff < 0) {
|
|
AudioCompression ^= 1;
|
|
} else {
|
|
AudioCompression = onoff;
|
|
}
|
|
AudioMaxCompression = maxfac;
|
|
if (!AudioCompressionFactor) {
|
|
AudioCompressionFactor = 1000;
|
|
}
|
|
if (AudioCompressionFactor > AudioMaxCompression) {
|
|
AudioCompressionFactor = AudioMaxCompression;
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Set stereo loudness descent.
|
|
**
|
|
** @param delta value (/1000) to reduce stereo volume
|
|
*/
|
|
void AudioSetStereoDescent(int delta) {
|
|
AudioStereoDescent = delta;
|
|
AudioSetVolume(AudioVolume); // update channel delta
|
|
}
|
|
|
|
/**
|
|
** Set pcm audio device.
|
|
**
|
|
** @param device name of pcm device (fe. "hw:0,9" or "/dev/dsp")
|
|
**
|
|
** @note this is currently used to select alsa/OSS output module.
|
|
*/
|
|
void AudioSetDevice(const char *device) {
|
|
if (!AudioModuleName) {
|
|
AudioModuleName = "alsa"; // detect alsa/OSS
|
|
if (!device[0]) {
|
|
AudioModuleName = "noop";
|
|
} else if (device[0] == '/') {
|
|
AudioModuleName = "oss";
|
|
}
|
|
}
|
|
AudioPCMDevice = device;
|
|
}
|
|
|
|
/**
|
|
** Set pass-through audio device.
|
|
**
|
|
** @param device name of pass-through device (fe. "hw:0,1")
|
|
**
|
|
** @note this is currently usable with alsa only.
|
|
*/
|
|
void AudioSetPassthroughDevice(const char *device) {
|
|
if (!AudioModuleName) {
|
|
AudioModuleName = "alsa"; // detect alsa/OSS
|
|
if (!device[0]) {
|
|
AudioModuleName = "noop";
|
|
} else if (device[0] == '/') {
|
|
AudioModuleName = "oss";
|
|
}
|
|
}
|
|
AudioPassthroughDevice = device;
|
|
}
|
|
|
|
/**
|
|
** Set pcm audio mixer channel.
|
|
**
|
|
** @param channel name of the mixer channel (fe. PCM or Master)
|
|
**
|
|
** @note this is currently used to select alsa/OSS output module.
|
|
*/
|
|
void AudioSetChannel(const char *channel) { AudioMixerChannel = channel; }
|
|
|
|
/**
|
|
** Set automatic AES flag handling.
|
|
**
|
|
** @param onoff turn setting AES flag on or off
|
|
*/
|
|
void AudioSetAutoAES(int onoff) {
|
|
if (onoff < 0) {
|
|
AudioAppendAES ^= 1;
|
|
} else {
|
|
AudioAppendAES = onoff;
|
|
}
|
|
}
|
|
|
|
/**
|
|
** Initialize audio output module.
|
|
**
|
|
** @todo FIXME: make audio output module selectable.
|
|
*/
|
|
void AudioInit(void) {
|
|
unsigned u;
|
|
const char *name;
|
|
int freq;
|
|
int chan;
|
|
|
|
name = "noop";
|
|
name = "alsa";
|
|
if (AudioModuleName) {
|
|
name = AudioModuleName;
|
|
}
|
|
//
|
|
// search selected audio module.
|
|
//
|
|
for (u = 0; u < sizeof(AudioModules) / sizeof(*AudioModules); ++u) {
|
|
if (!strcasecmp(name, AudioModules[u]->Name)) {
|
|
AudioUsedModule = AudioModules[u];
|
|
Info(_("audio: '%s' output module used\n"), AudioUsedModule->Name);
|
|
goto found;
|
|
}
|
|
}
|
|
Error(_("audio: '%s' output module isn't supported\n"), name);
|
|
AudioUsedModule = &NoopModule;
|
|
return;
|
|
|
|
found:
|
|
AudioDoingInit = 1;
|
|
AudioRingInit();
|
|
AudioUsedModule->Init();
|
|
//
|
|
// Check which channels/rates/formats are supported
|
|
// FIXME: we force 44.1Khz and 48Khz must be supported equal
|
|
// FIXME: should use bitmap of channels supported in RatesInHw
|
|
// FIXME: use loop over sample-rates
|
|
freq = 44100;
|
|
AudioRatesInHw[Audio44100] = 0;
|
|
for (chan = 1; chan < 9; ++chan) {
|
|
int tchan;
|
|
int tfreq;
|
|
|
|
tchan = chan;
|
|
tfreq = freq;
|
|
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
|
|
AudioChannelsInHw[chan] = 0;
|
|
} else {
|
|
AudioChannelsInHw[chan] = chan;
|
|
AudioRatesInHw[Audio44100] |= (1 << chan);
|
|
}
|
|
}
|
|
freq = 48000;
|
|
AudioRatesInHw[Audio48000] = 0;
|
|
for (chan = 1; chan < 9; ++chan) {
|
|
int tchan;
|
|
int tfreq;
|
|
|
|
if (!AudioChannelsInHw[chan]) {
|
|
continue;
|
|
}
|
|
tchan = chan;
|
|
tfreq = freq;
|
|
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
|
|
// AudioChannelsInHw[chan] = 0;
|
|
} else {
|
|
AudioChannelsInHw[chan] = chan;
|
|
AudioRatesInHw[Audio48000] |= (1 << chan);
|
|
}
|
|
}
|
|
freq = 192000;
|
|
AudioRatesInHw[Audio192000] = 0;
|
|
for (chan = 1; chan < 9; ++chan) {
|
|
int tchan;
|
|
int tfreq;
|
|
|
|
if (!AudioChannelsInHw[chan]) {
|
|
continue;
|
|
}
|
|
tchan = chan;
|
|
tfreq = freq;
|
|
if (AudioUsedModule->Setup(&tfreq, &tchan, 0)) {
|
|
// AudioChannelsInHw[chan] = 0;
|
|
} else {
|
|
AudioChannelsInHw[chan] = chan;
|
|
AudioRatesInHw[Audio192000] |= (1 << chan);
|
|
}
|
|
}
|
|
// build channel support and conversion table
|
|
for (u = 0; u < AudioRatesMax; ++u) {
|
|
for (chan = 1; chan < 9; ++chan) {
|
|
AudioChannelMatrix[u][chan] = 0;
|
|
if (!AudioRatesInHw[u]) { // rate unsupported
|
|
continue;
|
|
}
|
|
if (AudioChannelsInHw[chan]) {
|
|
AudioChannelMatrix[u][chan] = chan;
|
|
} else {
|
|
switch (chan) {
|
|
case 1:
|
|
if (AudioChannelsInHw[2]) {
|
|
AudioChannelMatrix[u][chan] = 2;
|
|
}
|
|
break;
|
|
case 2:
|
|
case 3:
|
|
if (AudioChannelsInHw[4]) {
|
|
AudioChannelMatrix[u][chan] = 4;
|
|
break;
|
|
}
|
|
case 4:
|
|
if (AudioChannelsInHw[5]) {
|
|
AudioChannelMatrix[u][chan] = 5;
|
|
break;
|
|
}
|
|
case 5:
|
|
if (AudioChannelsInHw[6]) {
|
|
AudioChannelMatrix[u][chan] = 6;
|
|
break;
|
|
}
|
|
case 6:
|
|
if (AudioChannelsInHw[7]) {
|
|
AudioChannelMatrix[u][chan] = 7;
|
|
break;
|
|
}
|
|
case 7:
|
|
if (AudioChannelsInHw[8]) {
|
|
AudioChannelMatrix[u][chan] = 8;
|
|
break;
|
|
}
|
|
case 8:
|
|
if (AudioChannelsInHw[6]) {
|
|
AudioChannelMatrix[u][chan] = 6;
|
|
break;
|
|
}
|
|
if (AudioChannelsInHw[2]) {
|
|
AudioChannelMatrix[u][chan] = 2;
|
|
break;
|
|
}
|
|
if (AudioChannelsInHw[1]) {
|
|
AudioChannelMatrix[u][chan] = 1;
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
for (u = 0; u < AudioRatesMax; ++u) {
|
|
Info(_("audio: %6dHz supports %d %d %d %d %d %d %d %d channels\n"), AudioRatesTable[u],
|
|
AudioChannelMatrix[u][1], AudioChannelMatrix[u][2], AudioChannelMatrix[u][3], AudioChannelMatrix[u][4],
|
|
AudioChannelMatrix[u][5], AudioChannelMatrix[u][6], AudioChannelMatrix[u][7], AudioChannelMatrix[u][8]);
|
|
}
|
|
#ifdef USE_AUDIO_THREAD
|
|
if (AudioUsedModule->Thread) { // supports threads
|
|
AudioInitThread();
|
|
}
|
|
#endif
|
|
AudioDoingInit = 0;
|
|
}
|
|
|
|
/**
|
|
** Cleanup audio output module.
|
|
*/
|
|
void AudioExit(void) {
|
|
const AudioModule *module;
|
|
|
|
Debug(3, "audio: %s\n", __FUNCTION__);
|
|
|
|
#ifdef USE_AUDIO_THREAD
|
|
if (AudioUsedModule->Thread) { // supports threads
|
|
AudioExitThread();
|
|
}
|
|
#endif
|
|
module = AudioUsedModule;
|
|
AudioUsedModule = &NoopModule;
|
|
module->Exit();
|
|
AudioRingExit();
|
|
AudioRunning = 0;
|
|
AudioPaused = 0;
|
|
}
|
|
|
|
#ifdef AUDIO_TEST
|
|
|
|
//----------------------------------------------------------------------------
|
|
// Test
|
|
//----------------------------------------------------------------------------
|
|
|
|
void AudioTest(void) {
|
|
for (;;) {
|
|
unsigned u;
|
|
uint8_t buffer[16 * 1024]; // some random data
|
|
int i;
|
|
|
|
for (u = 0; u < sizeof(buffer); u++) {
|
|
buffer[u] = random() & 0xffff;
|
|
}
|
|
|
|
Debug(3, "audio/test: loop\n");
|
|
for (i = 0; i < 100; ++i) {
|
|
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
|
|
AlsaEnqueue(buffer, sizeof(buffer));
|
|
}
|
|
usleep(20 * 1000);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
#include <getopt.h>
|
|
|
|
int SysLogLevel; ///< show additional debug informations
|
|
|
|
/**
|
|
** Print version.
|
|
*/
|
|
static void PrintVersion(void) {
|
|
printf("audio_test: audio tester Version " VERSION
|
|
#ifdef GIT_REV
|
|
"(GIT-" GIT_REV ")"
|
|
#endif
|
|
",\n\t(c) 2009 - 2013 by Johns\n"
|
|
"\tLicense AGPLv3: GNU Affero General Public License version 3\n");
|
|
}
|
|
|
|
/**
|
|
** Print usage.
|
|
*/
|
|
static void PrintUsage(void) {
|
|
printf("Usage: audio_test [-?dhv]\n"
|
|
"\t-d\tenable debug, more -d increase the verbosity\n"
|
|
"\t-? -h\tdisplay this message\n"
|
|
"\t-v\tdisplay version information\n"
|
|
"Only idiots print usage on stderr!\n");
|
|
}
|
|
|
|
/**
|
|
** Main entry point.
|
|
**
|
|
** @param argc number of arguments
|
|
** @param argv arguments vector
|
|
**
|
|
** @returns -1 on failures, 0 clean exit.
|
|
*/
|
|
int main(int argc, char *const argv[]) {
|
|
SysLogLevel = 0;
|
|
|
|
//
|
|
// Parse command line arguments
|
|
//
|
|
for (;;) {
|
|
switch (getopt(argc, argv, "hv?-c:d")) {
|
|
case 'd': // enabled debug
|
|
++SysLogLevel;
|
|
continue;
|
|
|
|
case EOF:
|
|
break;
|
|
case 'v': // print version
|
|
PrintVersion();
|
|
return 0;
|
|
case '?':
|
|
case 'h': // help usage
|
|
PrintVersion();
|
|
PrintUsage();
|
|
return 0;
|
|
case '-':
|
|
PrintVersion();
|
|
PrintUsage();
|
|
fprintf(stderr, "\nWe need no long options\n");
|
|
return -1;
|
|
case ':':
|
|
PrintVersion();
|
|
fprintf(stderr, "Missing argument for option '%c'\n", optopt);
|
|
return -1;
|
|
default:
|
|
PrintVersion();
|
|
fprintf(stderr, "Unknown option '%c'\n", optopt);
|
|
return -1;
|
|
}
|
|
break;
|
|
}
|
|
if (optind < argc) {
|
|
PrintVersion();
|
|
while (optind < argc) {
|
|
fprintf(stderr, "Unhandled argument '%s'\n", argv[optind++]);
|
|
}
|
|
return -1;
|
|
}
|
|
//
|
|
// main loop
|
|
//
|
|
AudioInit();
|
|
for (;;) {
|
|
unsigned u;
|
|
uint8_t buffer[16 * 1024]; // some random data
|
|
|
|
for (u = 0; u < sizeof(buffer); u++) {
|
|
buffer[u] = random() & 0xffff;
|
|
}
|
|
|
|
Debug(3, "audio/test: loop\n");
|
|
for (;;) {
|
|
while (RingBufferFreeBytes(AlsaRingBuffer) > sizeof(buffer)) {
|
|
AlsaEnqueue(buffer, sizeof(buffer));
|
|
}
|
|
}
|
|
}
|
|
AudioExit();
|
|
|
|
return 0;
|
|
}
|
|
|
|
#endif
|