Implemented (seemingly broken) Alsa audio player.
Fixed hunspell-related bugs in autoconf/automake system. Originally committed to SVN as r1114.
This commit is contained in:
Niels Martin Hansen
parent
ba944fd9b2
commit
5851f0561f
4 changed files with 497 additions and 10 deletions
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@ -6,7 +6,7 @@ lua51/src/liblua.a:
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all-recursive: lua51/src/liblua.a
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EXTRA_DIST = \
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EXTRA_DIST = ac/config.rpath m4/ChangeLog \
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lua51/COPYRIGHT \
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lua51/HISTORY \
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lua51/INSTALL \
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@ -76,3 +76,5 @@ EXTRA_DIST = \
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lua51/src/lvm.h \
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lua51/src/lzio.c \
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lua51/src/lzio.h
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ACLOCAL_AMFLAGS = -I m4
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@ -20,10 +20,14 @@ else
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LIBASS=
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endif
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AUDIO_PLAYER=audio_player.cpp
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if HAVE_PORTAUDIO
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PORTAUDIO=audio_player_portaudio.cpp
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else
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PORTAUDIO=
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AUDIO_PLAYER+=audio_player_portaudio.cpp
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LDADD += -lportaudio
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endif
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if HAVE_ALSA
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AUDIO_PLAYER+=audio_player_alsa.cpp
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LDADD += -lasound
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endif
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#if HAVE_RUBY
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@ -36,7 +40,7 @@ if HAVE_HUNSPELL
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HUNSPELL=spellchecker_hunspell.cpp
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else
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HUNSPELL=
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end
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endif
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aegisub_SOURCES = \
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aegisublocale.cpp \
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@ -53,8 +57,7 @@ aegisub_SOURCES = \
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audio_box.cpp \
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audio_display.cpp \
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audio_karaoke.cpp \
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audio_player.cpp \
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$(PORTAUDIO) \
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$(AUDIO_PLAYER) \
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audio_provider.cpp \
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audio_provider_hd.cpp \
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audio_provider_lavc.cpp \
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475
aegisub/audio_player_alsa.cpp
Normal file
475
aegisub/audio_player_alsa.cpp
Normal file
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@ -0,0 +1,475 @@
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// Copyright (c) 2007, 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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// -----------------------------------------------------------------------------
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//
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// AEGISUB
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//
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// Website: http://aegisub.cellosoft.com
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// Contact: mailto:jiifurusu@gmail.com
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//
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///////////
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// Headers
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#include <wx/wxprec.h>
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#include "audio_player.h"
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#include "audio_provider.h"
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#include "utils.h"
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#include "main.h"
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#include "frame_main.h"
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#include "audio_player.h"
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#include <alsa/asoundlib.h>
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//////////////
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// Prototypes
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class AlsaPlayer;
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//////////
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// Thread
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class AlsaPlayerThread : public wxThread {
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private:
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wxMutex play_mutex; // held while playing
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wxSemaphore play_notify; // posted when a playback operation is set up
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wxSemaphore stop_notify; // when set, audio playback should stop asap
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wxSemaphore shutdown_notify; // when set, thread should shutdown asap
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wxMutex parameter_mutex;
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AudioProvider *provider; // provides sample data!
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snd_pcm_t *pcm_handle; // device handle
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snd_pcm_stream_t stream; // stream direction
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snd_pcm_hw_params_t *hwparams; // hardware info
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char *pcm_name; // device name
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snd_pcm_format_t sample_format;
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unsigned int rate; // sample rate of audio
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unsigned int real_rate; // actual sample rate played back
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int periods; // number of bytes in a frame
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snd_pcm_uframes_t bufsize; // size of buffer in frames
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volatile double volume; // volume to get audio at
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volatile unsigned long start_frame; // first frame of playback
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volatile unsigned long cur_frame; // last written frame + 1
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volatile unsigned long end_frame; // last frame to play
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void SetUpHardware();
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void PlaybackLoop();
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public:
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AlsaPlayerThread(AudioProvider *_provider);
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~AlsaPlayerThread();
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wxThread::ExitCode Entry();
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// The following methods are all thread safe
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void Play(unsigned long _start_frame, unsigned long _num_frames); // Notify thread to stop any playback and instead play specified range
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void SetEndPosition(unsigned long _end_frame); // Notify thread to use new end position
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void Stop(); // Notify thread to stop audio playback
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void StopWait(); // Notify thread to stop, and wait for it to do it
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void Shutdown(); // Notify the thread to stop playback and die
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void SetVolume(double new_volume); // Set volume
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unsigned long GetStartPosition(); // Get first played back frame number
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unsigned long GetEndPosition(); // Get last frame number to be played back
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unsigned long GetCurrentPosition(); // Get currently played back frame number
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bool IsPlaying();
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};
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///////////////
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// Alsa player
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class AlsaPlayer : public AudioPlayer {
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private:
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float volume;
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AlsaPlayerThread *thread;
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public:
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AlsaPlayer();
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~AlsaPlayer();
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void OpenStream();
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void CloseStream();
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void Play(__int64 start,__int64 count);
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void Stop(bool timerToo=true);
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bool IsPlaying();
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__int64 GetStartPosition();
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__int64 GetEndPosition();
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__int64 GetCurrentPosition();
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void SetEndPosition(__int64 pos);
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void SetCurrentPosition(__int64 pos);
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void SetVolume(double vol) { volume = vol; }
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double GetVolume() { return volume; }
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};
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///////////
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// Factory
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class AlsaPlayerFactory : public AudioPlayerFactory {
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public:
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AudioPlayer *CreatePlayer() { return new AlsaPlayer(); }
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AlsaPlayerFactory() : AudioPlayerFactory(_T("alsa")) {}
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} registerAlsaPlayer;
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///////////////
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// Constructor
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AlsaPlayer::AlsaPlayer() {
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volume = 1.0f;
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thread = NULL;
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}
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//////////////
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// Destructor
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AlsaPlayer::~AlsaPlayer() {
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CloseStream();
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}
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///////////////
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// Open stream
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void AlsaPlayer::OpenStream() {
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CloseStream();
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// Get provider
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AudioProvider *provider = GetProvider();
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thread = new AlsaPlayerThread(provider);
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}
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////////////////
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// Close stream
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void AlsaPlayer::CloseStream() {
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if (!thread) return;
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thread->Shutdown();
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thread->Wait();
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thread = 0;
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}
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////////
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// Play
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void AlsaPlayer::Play(__int64 start,__int64 count) {
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// Make sure that it's stopped
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thread->Stop();
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thread->Play(start, count);
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}
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////////
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// Stop
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void AlsaPlayer::Stop(bool timerToo) {
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if (thread) thread->Stop();
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if (timerToo && displayTimer) {
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displayTimer->Stop();
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}
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}
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bool AlsaPlayer::IsPlaying()
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{
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return thread && thread->IsPlaying();
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}
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///////////
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// Set end
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void AlsaPlayer::SetEndPosition(__int64 pos) {
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if (thread) thread->SetEndPosition(pos);
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}
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////////////////////////
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// Set current position
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void AlsaPlayer::SetCurrentPosition(__int64 pos) {
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assert(false); // not supported (yet?)
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// I believe this isn't used anywhere. I hope not.
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}
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__int64 AlsaPlayer::GetStartPosition()
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{
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if (thread) return thread->GetStartPosition();
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return 0;
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}
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__int64 AlsaPlayer::GetEndPosition()
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{
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if (thread) return thread->GetEndPosition();
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return 0;
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}
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////////////////////////
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// Get current position
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__int64 AlsaPlayer::GetCurrentPosition() {
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if (thread) return thread->GetCurrentPosition();
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return 0;
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}
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//////////////////////
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// Thread constructor
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AlsaPlayerThread::AlsaPlayerThread(AudioProvider *_provider)
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: wxThread(wxTHREAD_JOINABLE)
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, play_notify(0, 1)
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, stop_notify(0, 1)
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, shutdown_notify(0, 1)
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, provider(_provider)
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{
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// We want playback
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stream = SND_PCM_STREAM_PLAYBACK;
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// Use default device and automatic sample type conversion
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pcm_name = "plughw:0,0";
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// Allocate params structure
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snd_pcm_hw_params_alloca(&hwparams);
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// Open device for blocking access
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if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0) {
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throw _T("Error opening default PCM device");
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}
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SetUpHardware();
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}
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/////////////////////
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// Thread destructor
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AlsaPlayerThread::~AlsaPlayerThread() {
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}
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void AlsaPlayerThread::SetUpHardware()
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{
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// Get hardware params
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if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
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throw _T("Error setting up default PCM device");
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}
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// Set stream format
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if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
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throw _T("Could not set interleaved stream format");
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}
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// Set sample format
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sample_format = SND_PCM_FORMAT_S16_LE; // TODO: support other formats
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if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, sample_format) < 0) {
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throw _T("Could not set sample format");
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}
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// Set sample rate
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rate = provider->GetSampleRate();
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real_rate = rate;
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if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0) < 0) {
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throw _T("Could not set sample rate");
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}
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if (rate != real_rate) {
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wxLogDebug(_T("Could not set ideal sample rate %d, using %d instead"), rate, real_rate);
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}
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// Set number of channels
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if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, provider->GetChannels()) < 0) {
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throw _T("Could not set number of channels");
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}
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// Set periods (size in bytes of one frame?)
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periods = provider->GetChannels() * provider->GetBytesPerSample();
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if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) {
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throw _T("Could not set periods");
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}
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// Set buffer size
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snd_pcm_uframes_t wanted_bufsize = rate;
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bufsize = wanted_bufsize;
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if (snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &bufsize) < 0) {
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throw _T("Could not set buffer size");
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}
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if (bufsize != wanted_bufsize) {
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wxLogDebug(_T("Couldn't get wanted buffer size of %u, got %u instead"), wanted_bufsize, bufsize);
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}
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// Apply parameters
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if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) {
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throw _T("Failed applying sound hardware settings");
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}
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}
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//////////////////////
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// Thread entry point
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wxThread::ExitCode AlsaPlayerThread::Entry()
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{
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// Loop as long as we aren't told to shutdown
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while (shutdown_notify.TryWait() == wxSEMA_BUSY) {
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// Wait for a playback operation
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while (play_notify.WaitTimeout(100) == wxSEMA_NO_ERROR) {
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// So playback was posted... now play something
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play_mutex.Lock();
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PlaybackLoop();
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play_mutex.Unlock();
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}
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}
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return 0;
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}
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void AlsaPlayerThread::PlaybackLoop()
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{
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unsigned long datalen = rate/2;
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void *data = malloc(datalen * periods);
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unsigned long cur_pos = 0;
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while (true) {
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// Get/set parameters
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parameter_mutex.Lock();
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double vol = volume;
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cur_frame = cur_pos;
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unsigned long end_pos = end_frame;
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parameter_mutex.Unlock();
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// Check for stop
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if (stop_notify.TryWait() != wxSEMA_BUSY) {
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snd_pcm_drop(pcm_handle);
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break;
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}
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// Check for end of stream
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if (cur_pos >= end_pos) {
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snd_pcm_drain(pcm_handle);
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break;
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}
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// Write some frames
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provider->GetAudioWithVolume(data, cur_pos, datalen, vol);
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snd_pcm_sframes_t written = snd_pcm_writei(pcm_handle, data, datalen);
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cur_pos += written;
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}
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free(data);
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}
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void AlsaPlayerThread::Play(unsigned long _start_frame, unsigned long _num_frames)
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{
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StopWait();
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cur_frame = start_frame = _start_frame;
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end_frame = start_frame + _num_frames;
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play_notify.Post();
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}
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void AlsaPlayerThread::SetEndPosition(unsigned long _end_frame)
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{
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parameter_mutex.Lock();
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end_frame = _end_frame;
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parameter_mutex.Unlock();
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}
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////////////////////////
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// Stop playback thread
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void AlsaPlayerThread::Stop()
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{
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stop_notify.Post();
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}
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void AlsaPlayerThread::StopWait()
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{
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stop_notify.Post();
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// Now wait for the play mutex to have been freed
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play_mutex.Lock();
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play_mutex.Unlock();
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}
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void AlsaPlayerThread::Shutdown()
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{
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Stop();
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shutdown_notify.Post();
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}
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void AlsaPlayerThread::SetVolume(double new_volume)
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{
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parameter_mutex.Lock();
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volume = new_volume;
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parameter_mutex.Unlock();
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}
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unsigned long AlsaPlayerThread::GetStartPosition()
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{
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wxMutexLocker lock(parameter_mutex);
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return start_frame;
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}
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unsigned long AlsaPlayerThread::GetEndPosition()
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{
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wxMutexLocker lock(parameter_mutex);
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return end_frame;
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}
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unsigned long AlsaPlayerThread::GetCurrentPosition()
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{
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// Step 1: Get last written frame number
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parameter_mutex.Lock();
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unsigned long pos = cur_frame;
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parameter_mutex.Unlock();
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// Step 2: ???
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// Step 3: Profit!
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return pos;
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}
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bool AlsaPlayerThread::IsPlaying()
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{
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return play_mutex.TryLock() == wxMUTEX_BUSY;
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}
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13
configure.ac
13
configure.ac
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@ -66,13 +66,18 @@ ACX_PTHREAD([], [
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AC_MSG_FAILURE([Aegisub needs pthreads])
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])
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AC_CHECK_LIB([portaudio], [Pa_Initialize],, [with_portaudio=no])
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AC_CHECK_LIB([asound], [snd_pcm_open], [found_audio=yes], [with_alsa=no])
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AC_CHECK_LIB([portaudio], [Pa_Initialize], [found_audio=yes], [with_portaudio=no])
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AC_CHECK_FUNCS([Pa_GetStreamTime])
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AC_CHECK_LIB([avcodec], [avcodec_init],, [AC_MSG_ERROR([avcodec not found.])])
|
||||
AC_CHECK_LIB([avformat], [av_read_frame],, [AC_MSG_ERROR([avformat not found.])])
|
||||
AC_CHECK_LIB([hunspell], [main])
|
||||
AC_CHECK_LIB([hunspell], [main],, [with_hunspell=no])
|
||||
# AC_CHECK_LIB([ruby1.9], [ruby_init],, [with_ruby=no])
|
||||
|
||||
if test -z "$found_audio"; then
|
||||
AC_MSG_WARN([No audio interface found, Aegisub will be built without Audio support!])
|
||||
fi
|
||||
|
||||
AC_CHECK_HEADER([wchar.h],,[
|
||||
AC_MSG_FAILURE([aegisub needs wide character support, find a wchar.h])
|
||||
])
|
||||
|
|
@ -205,14 +210,16 @@ AC_HEADER_STDBOOL
|
|||
AC_CONFIG_SUBDIRS([csri])
|
||||
|
||||
AM_CONDITIONAL([USE_LIBASS], [test "$with_libass" != "no" ])
|
||||
AM_CONDITIONAL([HAVE_ALSA], [test "$with_alsa" != "no"])
|
||||
AM_CONDITIONAL([HAVE_PORTAUDIO], [test "$with_portaudio" != "no"])
|
||||
AM_CONDITIONAL([HAVE_HUNSPELL], [test "$with_hunspell" != "no"])
|
||||
#AM_CONDITIONAL([HAVE_RUBY], [test "$with_ruby" != "no"])
|
||||
|
||||
AC_OUTPUT([
|
||||
Makefile
|
||||
ac/Makefile
|
||||
locale/Makefile
|
||||
FexTrackerSource/Makefile
|
||||
locale/Makefile
|
||||
aegisub/Makefile
|
||||
aegisub/posix/Makefile
|
||||
aegisub/bitmaps/Makefile
|
||||
|
|
|
|||
Loading…
Reference in a new issue