Aegisub/aegisub/audio_player_alsa.cpp

446 lines
11 KiB
C++

// Copyright (c) 2007, Niels Martin Hansen
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// -----------------------------------------------------------------------------
//
// AEGISUB
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:jiifurusu@gmail.com
//
///////////
// Headers
#include <wx/wxprec.h>
#include "audio_player.h"
#include "audio_provider.h"
#include "utils.h"
#include "main.h"
#include "frame_main.h"
#include "audio_player.h"
#include <alsa/asoundlib.h>
#include "options.h"
///////////////
// Alsa player
class AlsaPlayer : public AudioPlayer {
private:
bool open;
volatile bool playing;
volatile float volume;
volatile unsigned long start_frame; // first frame of playback
volatile unsigned long cur_frame; // last written frame + 1
volatile unsigned long end_frame; // last frame to play
unsigned long bpf; // bytes per frame
AudioProvider *provider;
snd_pcm_t *pcm_handle; // device handle
snd_pcm_stream_t stream; // stream direction
snd_async_handler_t *pcm_callback;
snd_pcm_format_t sample_format;
unsigned int rate; // sample rate of audio
unsigned int real_rate; // actual sample rate played back
unsigned int period_len; // length of period in microseconds
unsigned int buflen; // length of buffer in microseconds
snd_pcm_uframes_t period; // size of period in frames
snd_pcm_uframes_t bufsize; // size of buffer in frames
void SetUpHardware();
void SetUpAsync();
static void async_write_handler(snd_async_handler_t *pcm_callback);
public:
AlsaPlayer();
~AlsaPlayer();
void OpenStream();
void CloseStream();
void Play(__int64 start,__int64 count);
void Stop(bool timerToo=true);
bool IsPlaying();
__int64 GetStartPosition();
__int64 GetEndPosition();
__int64 GetCurrentPosition();
void SetEndPosition(__int64 pos);
void SetCurrentPosition(__int64 pos);
void SetVolume(double vol) { volume = vol; }
double GetVolume() { return volume; }
};
///////////
// Factory
class AlsaPlayerFactory : public AudioPlayerFactory {
public:
AudioPlayer *CreatePlayer() { return new AlsaPlayer(); }
AlsaPlayerFactory() : AudioPlayerFactory(_T("alsa")) {}
} registerAlsaPlayer;
///////////////
// Constructor
AlsaPlayer::AlsaPlayer()
{
volume = 1.0f;
open = false;
playing = false;
start_frame = cur_frame = end_frame = bpf = 0;
provider = 0;
}
//////////////
// Destructor
AlsaPlayer::~AlsaPlayer()
{
CloseStream();
}
///////////////
// Open stream
void AlsaPlayer::OpenStream()
{
CloseStream();
// Get provider
provider = GetProvider();
bpf = provider->GetChannels() * provider->GetBytesPerSample();
// We want playback
stream = SND_PCM_STREAM_PLAYBACK;
// And get a device name
wxString device = Options.AsText(_T("Audio Alsa Device"));
// Open device for blocking access
if (snd_pcm_open(&pcm_handle, device.mb_str(wxConvUTF8), stream, 0) < 0) { // supposedly we don't want SND_PCM_ASYNC even for async playback
throw _T("Error opening specified PCM device");
}
SetUpHardware();
// Register async handler
SetUpAsync();
// Now ready
open = true;
}
void AlsaPlayer::SetUpHardware()
{
int dir;
// Allocate params structure
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_malloc(&hwparams);
// Get hardware params
if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
throw _T("Error setting up default PCM device");
}
// Set stream format
if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
throw _T("Could not set interleaved stream format");
}
// Set sample format
switch (provider->GetBytesPerSample()) {
case 1:
sample_format = SND_PCM_FORMAT_S8;
break;
case 2:
sample_format = SND_PCM_FORMAT_S16_LE;
break;
default:
throw _T("Can only handle 8 and 16 bit sound");
}
if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, sample_format) < 0) {
throw _T("Could not set sample format");
}
// Ask for resampling
if (snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1) < 0) {
throw _T("Couldn't enable resampling");
}
// Set sample rate
rate = provider->GetSampleRate();
real_rate = rate;
if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0) < 0) {
throw _T("Could not set sample rate");
}
if (rate != real_rate) {
wxLogDebug(_T("Could not set ideal sample rate %d, using %d instead"), rate, real_rate);
}
// Set number of channels
if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, provider->GetChannels()) < 0) {
throw _T("Could not set number of channels");
}
printf("Set sample rate %u (wanted %u)\n", real_rate, rate);
// Set buffer size
unsigned int wanted_buflen = 1000000; // microseconds
buflen = wanted_buflen;
if (snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buflen, &dir) < 0) {
throw _T("Couldn't set buffer length");
}
if (buflen != wanted_buflen) {
wxLogDebug(_T("Couldn't get wanted buffer size of %u, got %u instead"), wanted_buflen, buflen);
}
if (snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize) < 0) {
throw _T("Couldn't get buffer size");
}
printf("Buffer size: %lu\n", bufsize);
// Set period (number of frames ideally written at a time)
// Somewhat arbitrary for now
unsigned int wanted_period = bufsize / 4;
period_len = wanted_period; // microseconds
if (snd_pcm_hw_params_set_period_time_near(pcm_handle, hwparams, &period_len, &dir) < 0) {
throw _T("Couldn't set period length");
}
if (period_len != wanted_period) {
wxLogDebug(_T("Couldn't get wanted period size of %d, got %d instead"), wanted_period, period_len);
}
if (snd_pcm_hw_params_get_period_size(hwparams, &period, &dir) < 0) {
throw _T("Couldn't get period size");
}
printf("Period size: %lu\n", period);
// Apply parameters
if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) {
throw _T("Failed applying sound hardware settings");
}
// And free memory again
snd_pcm_hw_params_free(hwparams);
}
void AlsaPlayer::SetUpAsync()
{
// Prepare software params struct
snd_pcm_sw_params_t *sw_params;
snd_pcm_sw_params_malloc (&sw_params);
// Get current parameters
if (snd_pcm_sw_params_current(pcm_handle, sw_params) < 0) {
throw _T("Couldn't get current SW params");
}
// How full the buffer must be before playback begins
if (snd_pcm_sw_params_set_start_threshold(pcm_handle, sw_params, bufsize - period) < 0) {
throw _T("Failed setting start threshold");
}
// The the largest write guaranteed never to block
if (snd_pcm_sw_params_set_avail_min(pcm_handle, sw_params, period) < 0) {
throw _T("Failed setting min available buffer");
}
// Apply settings
if (snd_pcm_sw_params(pcm_handle, sw_params) < 0) {
throw _T("Failed applying SW params");
}
// And free struct again
snd_pcm_sw_params_free(sw_params);
// Attach async handler
if (snd_async_add_pcm_handler(&pcm_callback, pcm_handle, async_write_handler, this) < 0) {
throw _T("Failed attaching async handler");
}
}
////////////////
// Close stream
void AlsaPlayer::CloseStream()
{
if (!open) return;
Stop();
// Remove async handler
snd_async_del_handler(pcm_callback);
// Close device
snd_pcm_close(pcm_handle);
// No longer working
open = false;
}
////////
// Play
void AlsaPlayer::Play(__int64 start,__int64 count)
{
if (playing) {
// Quick reset
playing = false;
snd_pcm_drop(pcm_handle);
}
// Set params
start_frame = start;
cur_frame = start;
end_frame = start + count;
playing = true;
// Prepare a bit
snd_pcm_prepare (pcm_handle);
async_write_handler(pcm_callback);
// And go!
snd_pcm_start(pcm_handle);
// Update timer
if (displayTimer && !displayTimer->IsRunning()) displayTimer->Start(15);
}
////////
// Stop
void AlsaPlayer::Stop(bool timerToo)
{
if (!open) return;
if (!playing) return;
// Reset data
playing = false;
start_frame = 0;
cur_frame = 0;
end_frame = 0;
// Then drop the playback
snd_pcm_drop(pcm_handle);
if (timerToo && displayTimer) {
displayTimer->Stop();
}
}
bool AlsaPlayer::IsPlaying()
{
return playing;
}
///////////
// Set end
void AlsaPlayer::SetEndPosition(__int64 pos)
{
end_frame = pos;
}
////////////////////////
// Set current position
void AlsaPlayer::SetCurrentPosition(__int64 pos)
{
cur_frame = pos;
}
__int64 AlsaPlayer::GetStartPosition()
{
return start_frame;
}
__int64 AlsaPlayer::GetEndPosition()
{
return end_frame;
}
////////////////////////
// Get current position
__int64 AlsaPlayer::GetCurrentPosition()
{
// FIXME: this should be based on not duration played but actual sample being heard
// (during vidoeo playback, cur_frame might get changed to resync)
snd_pcm_sframes_t delay = 0;
snd_pcm_delay(pcm_handle, &delay); // don't bother catching errors here
return cur_frame - delay;
}
void AlsaPlayer::async_write_handler(snd_async_handler_t *pcm_callback)
{
// TODO: check for broken pipes in here and restore as needed
AlsaPlayer *player = (AlsaPlayer*)snd_async_handler_get_callback_private(pcm_callback);
if (player->cur_frame >= player->end_frame + player->rate) {
// More than a second past end of stream
snd_pcm_drain(player->pcm_handle);
player->playing = false;
return;
}
snd_pcm_sframes_t frames = snd_pcm_avail_update(player->pcm_handle);
// TODO: handle underrun
if (player->cur_frame >= player->end_frame) {
// Past end of stream, add some silence
void *buf = calloc(frames, player->bpf);
snd_pcm_writei(player->pcm_handle, buf, frames);
free(buf);
player->cur_frame += frames;
return;
}
void *buf = malloc(player->period * player->bpf);
while (frames >= player->period) {
unsigned long start = player->cur_frame;
player->provider->GetAudioWithVolume(buf, player->cur_frame, player->period, player->volume);
snd_pcm_writei(player->pcm_handle, buf, player->period);
player->cur_frame += player->period;
frames = snd_pcm_avail_update(player->pcm_handle);
}
free(buf);
}