Aegisub/aegisub/src/audio_player_alsa.cpp

// Copyright (c) 2007, Niels Martin Hansen
// All rights reserved.
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// Aegisub Project http://www.aegisub.org/
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// $Id$

/// @file audio_player_alsa.cpp
/// @brief ALSA-based audio output
/// @ingroup audio_output
///


#include "config.h"


#ifdef WITH_ALSA


///////////
// Headers
#include <libaegisub/log.h>

#include "audio_player_alsa.h"
#include "audio_player_manager.h"
#include "audio_provider_manager.h"
#include "main.h"
#include "compat.h"
#include "frame_main.h"
#include "utils.h"


/// @brief Constructor 
///
AlsaPlayer::AlsaPlayer()
{
	volume = 1.0f;
	open = false;
	playing = false;
	start_frame = cur_frame = end_frame = bpf = 0;
	provider = 0;
}



/// @brief Destructor 
///
AlsaPlayer::~AlsaPlayer()
{
	CloseStream();
}



/// @brief 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 = lagi_wxString(OPT_GET("Player/Audio/ALSA/Device")->GetString());

	// 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("ALSA player: Error opening specified PCM device");
	}

	SetUpHardware();

	// Register async handler
	SetUpAsync();

	// Now ready
	open = true;
}



/// @brief DOCME
///
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("ALSA player: 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("ALSA player: 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("ALSA player: Can only handle 8 and 16 bit sound");
	}
	if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, sample_format) < 0) {
		throw _T("ALSA player: Could not set sample format");
	}

	// Ask for resampling
	if (snd_pcm_hw_params_set_rate_resample(pcm_handle, hwparams, 1) < 0) {
		throw _T("ALSA player: 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("ALSA player: Could not set sample rate");
	}
	LOG_D_IF(rate != real_rate, "player/audio/alsa") << "ALSA player: Could not set ideal sample rate " << rate << "using " << real_rate << "instead";

	// Set number of channels
	if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, provider->GetChannels()) < 0) {
		LOG_E("player/audio/alsa") << "Could not set number of channels";
		throw _T("ALSA player: Could not set number of channels");
	}
	printf("ALSA player: 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) {
		LOG_E("player/audio/alsa") << "Couldn't set buffer length";
		throw _T("ALSA player: Couldn't set buffer length");
	}
	LOG_D_IF(buflen != wanted_buflen, "player/audio/alsa") << "Couldn't get wanted buffer size of " << wanted_buflen << ", got " << buflen << "instead";

	if (snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize) < 0) {
		LOG_E("player/audio/alsa") << "Couldn't get buffer size";
		throw _T("ALSA player: Couldn't get buffer size");
	}
	printf("ALSA player: 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("ALSA player: Couldn't set period length");
	}
	LOG_D_IF(period_len != wanted_period, "player/audio/alsa") << "Couldn't get wanted period size of " << wanted_period << ", got " << period_len << " instead";

	if (snd_pcm_hw_params_get_period_size(hwparams, &period, &dir) < 0) {
		LOG_E("player/audio/alsa") << "Couldn't get period size";
		throw _T("ALSA player: Couldn't get period size");
	}
	printf("ALSA player: Period size: %lu\n", period);

	// Apply parameters
	if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) {
		LOG_E("player/audio/alsa") << "Failed applying sound hardware settings";
		throw _T("ALSA player: Failed applying sound hardware settings");
	}

	// And free memory again
	snd_pcm_hw_params_free(hwparams);
}



/// @brief DOCME
///
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) {
		LOG_E("player/audio/alsa") << "Couldn't get current SW params";
		throw _T("ALSA player: 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) {
		LOG_E("player/audio/alsa") << "Failed setting start threshold";
		throw _T("ALSA player: 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) {
		LOG_E("player/audio/alsa") << "Failed setting min available buffer";
		throw _T("ALSA player: Failed setting min available buffer");
	}

	// Apply settings
	if (snd_pcm_sw_params(pcm_handle, sw_params) < 0) {
		LOG_E("player/audio/alsa") << "Failed applying SW params";
		throw _T("ALSA player: Failed applying SW params");
	}

	// And free struct again
	snd_pcm_sw_params_free(sw_params);

	// Prepare for playback
	snd_pcm_prepare(pcm_handle);

	// Attach async handler
	if (snd_async_add_pcm_handler(&pcm_callback, pcm_handle, async_write_handler, this) < 0) {
		LOG_E("player/audio/alsa") << "Failed attaching async handler";
		throw _T("ALSA player: Failed attaching async handler");
	}
}



/// @brief Close stream 
/// @return 
///
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;
}



/// @brief Play 
/// @param start 
/// @param count 
///
void AlsaPlayer::Play(int64_t start,int64_t 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);
}



/// @brief Stop 
/// @param timerToo 
/// @return 
///
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();
        }
}



/// @brief DOCME
/// @return 
///
bool AlsaPlayer::IsPlaying()
{
	return playing;
}



/// @brief Set end 
/// @param pos 
///
void AlsaPlayer::SetEndPosition(int64_t pos)
{
	end_frame = pos;
}



/// @brief Set current position 
/// @param pos 
///
void AlsaPlayer::SetCurrentPosition(int64_t pos)
{
	cur_frame = pos;
}



/// @brief DOCME
/// @return 
///
int64_t AlsaPlayer::GetStartPosition()
{
	return start_frame;
}



/// @brief DOCME
/// @return 
///
int64_t AlsaPlayer::GetEndPosition()
{
	return end_frame;
}



/// @brief Get current position 
/// @return 
///
int64_t 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;
}



/// @brief DOCME
/// @param pcm_callback 
///
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);
	
	if (frames == -EPIPE) {
		snd_pcm_prepare(player->pcm_handle);
		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);
		int err = snd_pcm_writei(player->pcm_handle, buf, player->period);
		if(err == -EPIPE) {
			snd_pcm_prepare(player->pcm_handle);
		}
		player->cur_frame += player->period;
		frames = snd_pcm_avail_update(player->pcm_handle);
	}
	free(buf);
}


#endif // WITH_ALSA