mirror of https://github.com/odrling/Aegisub
537 lines
13 KiB
C++
537 lines
13 KiB
C++
// Copyright (c) 2011, Niels Martin Hansen
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// All rights reserved.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// Aegisub Project http://www.aegisub.org/
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//
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// $Id$
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/// @file audio_player_alsa.cpp
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/// @brief ALSA-based audio output
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/// @ingroup audio_output
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///
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#include "config.h"
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#ifdef WITH_ALSA
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#include <libaegisub/log.h>
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#include "audio_player_alsa.h"
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#include "audio_controller.h"
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#include "include/aegisub/audio_provider.h"
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#include "compat.h"
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#include "frame_main.h"
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#include "main.h"
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#ifndef AGI_PRE
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#include <algorithm>
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#include <inttypes.h>
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#endif
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class PthreadMutexLocker {
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pthread_mutex_t &mutex;
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PthreadMutexLocker(const PthreadMutexLocker &); // uncopyable
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PthreadMutexLocker(); // no default
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PthreadMutexLocker& operator=(PthreadMutexLocker const&);
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public:
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explicit PthreadMutexLocker(pthread_mutex_t &mutex) : mutex(mutex)
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{
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pthread_mutex_lock(&mutex);
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}
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~PthreadMutexLocker()
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{
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pthread_mutex_unlock(&mutex);
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}
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int WaitCondition(pthread_cond_t &cond)
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{
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return pthread_cond_wait(&cond, &mutex);
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}
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int WaitConditionTimeout(pthread_cond_t &cond, int ms)
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{
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timespec abstime;
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clock_gettime(CLOCK_REALTIME, &abstime);
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abstime.tv_nsec += ms * 1000000;
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return pthread_cond_timedwait(&cond, &mutex, &abstime);
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}
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};
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class ScopedAliveFlag {
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bool &flag;
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ScopedAliveFlag(const ScopedAliveFlag &); // uncopyable
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ScopedAliveFlag(); // no default
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ScopedAliveFlag& operator=(ScopedAliveFlag const&);
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public:
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explicit ScopedAliveFlag(bool &var) : flag(var) { flag = true; }
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~ScopedAliveFlag() { flag = false; }
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};
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struct PlaybackState {
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pthread_mutex_t mutex;
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pthread_cond_t cond;
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bool playing;
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bool alive;
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bool signal_start;
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bool signal_stop;
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bool signal_close;
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bool signal_volume;
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double volume;
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int64_t start_position;
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int64_t end_position;
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AudioProvider *provider;
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std::string device_name;
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int64_t last_position;
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timespec last_position_time;
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PlaybackState()
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{
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pthread_mutex_init(&mutex, 0);
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pthread_cond_init(&cond, 0);
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Reset();
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volume = 1.0;
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}
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~PlaybackState()
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{
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pthread_cond_destroy(&cond);
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pthread_mutex_destroy(&mutex);
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}
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void Reset()
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{
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playing = false;
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alive = false;
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signal_start = false;
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signal_stop = false;
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signal_close = false;
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signal_volume = false;
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start_position = 0;
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end_position = 0;
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last_position = 0;
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provider = 0;
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}
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};
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void *playback_thread(void *arg)
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{
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// This is exception-free territory!
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// Return a pointer to a static string constant describing the error, or 0 on no error
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PlaybackState &ps = *(PlaybackState*)arg;
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PthreadMutexLocker ml(ps.mutex);
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ScopedAliveFlag alive_flag(ps.alive);
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snd_pcm_t *pcm = 0;
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if (snd_pcm_open(&pcm, ps.device_name.c_str(), SND_PCM_STREAM_PLAYBACK, 0) != 0)
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return (void*)"snd_pcm_open";
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LOG_D("audio/player/alsa") << "opened pcm";
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do_setup:
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snd_pcm_format_t pcm_format;
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switch (ps.provider->GetBytesPerSample())
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{
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case 1:
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LOG_D("audio/player/alsa") << "format U8";
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pcm_format = SND_PCM_FORMAT_U8;
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break;
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case 2:
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LOG_D("audio/player/alsa") << "format S16_LE";
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pcm_format = SND_PCM_FORMAT_S16_LE;
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break;
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default:
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snd_pcm_close(pcm);
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return (void*)"snd_pcm_format_t";
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}
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if (snd_pcm_set_params(pcm,
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pcm_format,
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SND_PCM_ACCESS_RW_INTERLEAVED,
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ps.provider->GetChannels(),
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ps.provider->GetSampleRate(),
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1, // allow resample
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100*1000 // 100 milliseconds latency
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) != 0)
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return (void*)"snd_pcm_set_params";
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LOG_D("audio/player/alsa") << "set pcm params";
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size_t framesize = ps.provider->GetChannels() * ps.provider->GetBytesPerSample();
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ps.signal_close = false;
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while (ps.signal_close == false)
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{
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// Wait for condition to trigger
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if (!ps.signal_start)
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ml.WaitCondition(ps.cond);
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LOG_D("audio/player/alsa") << "outer loop, condition happened";
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if (ps.signal_start == false || ps.end_position <= ps.start_position)
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{
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LOG_D("audio/player/alsa") << "nothing to play, rewaiting";
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ps.signal_start = false;
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continue;
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}
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LOG_D("audio/player/alsa") << "starting playback";
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int64_t position = ps.start_position;
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// Playback position
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ps.last_position = position;
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clock_gettime(CLOCK_REALTIME, &ps.last_position_time);
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// Initial buffer-fill
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snd_pcm_sframes_t avail = std::min(snd_pcm_avail(pcm), (snd_pcm_sframes_t)(ps.end_position-position));
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char *buf = new char[avail*framesize];
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ps.provider->GetAudioWithVolume(buf, position, avail, ps.volume);
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snd_pcm_sframes_t written = 0;
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while (written <= 0)
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{
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written = snd_pcm_writei(pcm, buf, avail);
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if (written == -ESTRPIPE)
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{
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snd_pcm_recover(pcm, written, 0);
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}
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else if (written <= 0)
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{
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delete[] buf;
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snd_pcm_close(pcm);
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LOG_D("audio/player/alsa") << "error filling buffer";
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return (void*)"snd_pcm_writei";
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}
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}
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delete[] buf;
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position += written;
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// Start playback
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LOG_D("audio/player/alsa") << "initial buffer filled, hitting start";
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snd_pcm_start(pcm);
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ps.signal_start = false;
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ps.signal_stop = false;
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while (ps.signal_stop == false)
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{
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int64_t orig_position = position;
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int64_t orig_ps_end_position = ps.end_position;
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ScopedAliveFlag playing_flag(ps.playing);
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// Sleep a bit, or until an event
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ml.WaitConditionTimeout(ps.cond, 50);
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//LOG_D("audio/player/alsa") << "playback loop, out of wait";
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// Check for stop signal
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if (ps.signal_stop == true)
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{
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LOG_D("audio/player/alsa") << "playback loop, stop signal";
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snd_pcm_drop(pcm);
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break;
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}
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// Playback position
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snd_pcm_sframes_t delay;
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if (snd_pcm_delay(pcm, &delay) == 0)
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{
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ps.last_position = position - delay;
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clock_gettime(CLOCK_REALTIME, &ps.last_position_time);
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}
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// Fill buffer
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long tmp_pcm_avail = snd_pcm_avail(pcm);
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if (tmp_pcm_avail == -EPIPE)
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{
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if (snd_pcm_recover(pcm, -EPIPE, 1) < 0)
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{
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LOG_D("audio/player/alsa") << "failed to recover from underrun";
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return (void*)"snd_pcm_avail";
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}
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tmp_pcm_avail = snd_pcm_avail(pcm);
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}
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avail = std::min(tmp_pcm_avail, (snd_pcm_sframes_t)(ps.end_position-position));
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if (avail < 0)
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{
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printf("\n--------- avail was less than 0: %" PRId64 "\n", avail);
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printf("snd_pcm_avail(pcm): %" PRId64 "\n", tmp_pcm_avail);
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printf("original position: %" PRId64 "\n", orig_position);
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printf("current position: %" PRId64 "\n", position);
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printf("original ps.end_position: %" PRId64 "\n", orig_ps_end_position);
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printf("current ps.end_position: %" PRId64 "\n", ps.end_position);
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printf("---------\n\n");
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continue;
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}
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buf = new char[avail*framesize];
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ps.provider->GetAudioWithVolume(buf, position, avail, ps.volume);
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written = 0;
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while (written <= 0)
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{
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written = snd_pcm_writei(pcm, buf, avail);
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if (written == -ESTRPIPE || written == -EPIPE)
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{
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snd_pcm_recover(pcm, written, 0);
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}
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else if (written == 0)
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{
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break;
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}
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else if (written < 0)
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{
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delete[] buf;
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snd_pcm_close(pcm);
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LOG_D("audio/player/alsa") << "error filling buffer, written=" << written;
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return (void*)"snd_pcm_writei";
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}
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}
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delete[] buf;
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position += written;
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//LOG_D("audio/player/alsa") << "playback loop, filled buffer";
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// Check for end of playback
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if (position >= ps.end_position)
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{
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LOG_D("audio/player/alsa") << "playback loop, past end, draining";
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snd_pcm_drain(pcm);
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break;
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}
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}
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ps.signal_stop = false;
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LOG_D("audio/player/alsa") << "out of playback loop";
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switch (snd_pcm_state(pcm))
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{
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case SND_PCM_STATE_OPEN:
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// no clue what could have happened here, but start over
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ps.signal_start = false;
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ps.signal_stop = false;
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goto do_setup;
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case SND_PCM_STATE_SETUP:
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// we lost the preparedness?
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snd_pcm_prepare(pcm);
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break;
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case SND_PCM_STATE_DISCONNECTED:
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// lost device, close the handle and return error
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snd_pcm_close(pcm);
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return (void*)"SND_PCM_STATE_DISCONNECTED";
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default:
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// everything else should either be fine or impossible (here)
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break;
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}
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}
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ps.signal_close = false;
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LOG_D("audio/player/alsa") << "out of outer loop";
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snd_pcm_close(pcm);
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return 0;
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}
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AlsaPlayer::AlsaPlayer()
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: ps(new PlaybackState)
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{
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open = false;
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}
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AlsaPlayer::~AlsaPlayer()
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{
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CloseStream();
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}
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void AlsaPlayer::OpenStream()
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{
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if (open) return;
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CloseStream();
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ps->Reset();
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ps->provider = provider;
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wxString device_name = lagi_wxString(OPT_GET("Player/Audio/ALSA/Device")->GetString());
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ps->device_name = std::string(device_name.utf8_str());
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if (pthread_create(&thread, 0, &playback_thread, ps.get()) == 0)
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open = true;
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else
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throw agi::AudioPlayerOpenError("AlsaPlayer: Creating the playback thread failed", 0);
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}
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void AlsaPlayer::CloseStream()
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{
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if (!open) return;
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{
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PthreadMutexLocker ml(ps->mutex);
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ps->signal_stop = true;
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ps->signal_close = true;
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LOG_D("audio/player/alsa") << "close stream, stop+close signal";
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pthread_cond_signal(&ps->cond);
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}
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pthread_join(thread, 0); // FIXME: check for errors
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open = false;
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}
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void AlsaPlayer::Play(int64_t start, int64_t count)
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{
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OpenStream();
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PthreadMutexLocker ml(ps->mutex);
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ps->signal_start = true;
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ps->signal_stop = true; // make sure to stop any ongoing playback first
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ps->start_position = start;
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ps->end_position = start + count;
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pthread_cond_signal(&ps->cond);
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}
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void AlsaPlayer::Stop()
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{
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if (!open) return;
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PthreadMutexLocker ml(ps->mutex);
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ps->signal_stop = true;
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LOG_D("audio/player/alsa") << "stop stream, stop signal";
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pthread_cond_signal(&ps->cond);
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}
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bool AlsaPlayer::IsPlaying()
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{
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PthreadMutexLocker ml(ps->mutex);
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return open && ps->playing;
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}
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void AlsaPlayer::SetEndPosition(int64_t pos)
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{
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if (!open) return;
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PthreadMutexLocker ml(ps->mutex);
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ps->end_position = pos;
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}
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void AlsaPlayer::SetCurrentPosition(int64_t pos)
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{
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if (!open) return;
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PthreadMutexLocker ml(ps->mutex);
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if (!ps->playing) return;
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ps->start_position = pos;
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ps->signal_start = true;
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ps->signal_stop = true;
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LOG_D("audio/player/alsa") << "set position, stop+start signal";
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pthread_cond_signal(&ps->cond);
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}
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int64_t AlsaPlayer::GetStartPosition()
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{
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if (!open) return 0;
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PthreadMutexLocker ml(ps->mutex);
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return ps->start_position;
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}
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int64_t AlsaPlayer::GetEndPosition()
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{
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if (!open) return 0;
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PthreadMutexLocker ml(ps->mutex);
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return ps->end_position;
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}
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int64_t AlsaPlayer::GetCurrentPosition()
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{
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if (!open) return 0;
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int64_t lastpos;
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timespec lasttime;
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int64_t samplerate;
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{
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PthreadMutexLocker ml(ps->mutex);
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lastpos = ps->last_position;
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lasttime = ps->last_position_time;
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samplerate = ps->provider->GetSampleRate();
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}
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timespec now;
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clock_gettime(CLOCK_REALTIME, &now);
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const double NANO = 1000000000; // nano- is 10^-9
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double now_sec = now.tv_sec + now.tv_nsec/NANO;
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double last_sec = lasttime.tv_sec + lasttime.tv_nsec/NANO;
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double diff_sec = now_sec - last_sec;
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int64_t pos = lastpos + (int64_t)(diff_sec * samplerate);
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//printf("AlsaPlayer: current position = %lld\n", pos);
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return pos;
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}
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void AlsaPlayer::SetVolume(double vol)
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{
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if (!open) return;
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PthreadMutexLocker ml(ps->mutex);
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ps->volume = vol;
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ps->signal_volume = true;
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pthread_cond_signal(&ps->cond);
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}
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double AlsaPlayer::GetVolume()
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{
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if (!open) return 1.0;
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PthreadMutexLocker ml(ps->mutex);
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return ps->volume;
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}
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#endif // WITH_ALSA
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