/*
* Close out audio sync - stop and disconnect from SDL
*/
CSDLAudioSync::~CSDLAudioSync (void)
{
Our_SDL_PauseAudio(1);
Our_SDL_CloseAudio();
for (int ix = 0; ix < DECODE_BUFFERS_MAX; ix++) {
if (m_sample_buffer[ix] != NULL)
free(m_sample_buffer[ix]);
m_sample_buffer[ix] = NULL;
}
audio_message(LOG_NOTICE,
"Audio sync skipped %u buffers",
m_skipped_buffers);
audio_message(LOG_NOTICE, "didn't fill %u buffers", m_didnt_fill_buffers);
SDL_DestroySemaphore(m_audio_waiting);
}
/*
* Create an CPortAudioSync for a session. Don't alloc any buffers until
* config is called by codec
*/
CPortAudioSync::CPortAudioSync (CPlayerSession *psptr, int volume) :
CBufferAudioSync(psptr, volume)
{
PaError err;
err = Pa_Initialize();
if (err != paNoError) {
audio_message(LOG_CRIT, "resp from Pa_Initialize is %d %s", err,
Pa_GetErrorText(err));
}
m_volume = volume;
m_pa_stream = NULL;
}
int do_we_have_audio (void)
{
PaError err;
err = Pa_Initialize();
if (err != paNoError) {
audio_message(LOG_ERR, "No audio - %d %s", err,
Pa_GetErrorText(err));
return 0;
}
Pa_Terminate();
return (1);
}
/*
* codec api - set up information about the stream
*/
void CSDLAudioSync::set_config (int freq,
int channels,
audio_format_t format,
uint32_t sample_size)
{
if (m_config_set != 0)
return 0;
audio_message(LOG_DEBUG, "configure audio freq %d chan %d format %d",
freq, channels, format);
switch (format) {
case AUDIO_FMT_U8:
case AUDIO_FMT_S8:
case AUDIO_FMT_HW_AC3:
m_bytes_per_sample_input = sizeof(uint8_t);
m_bytes_per_sample_output = sizeof(uint8_t);
break;
case AUDIO_FMT_FLOAT:
// we convert float to S16 (native format)
m_bytes_per_sample_input = sizeof(float);
m_bytes_per_sample_output = sizeof(int16_t);
break;
default:
m_bytes_per_sample_input = sizeof(int16_t);
m_bytes_per_sample_output = sizeof(int16_t);
break;
}
if (sample_size == 0) {
int temp;
temp = freq;
while ((temp & 0x1) == 0) temp >>= 1;
sample_size = temp;
while (sample_size < 1024) sample_size *= 2;
while (((sample_size * 1000) % freq) != 0) sample_size *= 2;
}
// Sync task api - initialize the sucker.
// May need to check out non-standard frequencies, see about conversion.
// returns 1 for initialized, -1 for error
int CPortAudioSync::InitializeHardware (void)
{
uint32_t bytes_per_sample;
uint32_t sample_size;
bool need_convert = false;
// bytes per sample is the # of bytes for the output buffer
bytes_per_sample = 2;
if (m_decode_format == AUDIO_FMT_U8 || m_decode_format == AUDIO_FMT_S8)
bytes_per_sample = 1;
sample_size = m_samples_per_frame;
uint32_t ix;
for (ix = 2; ix <= 0x8000; ix <<= 1) {
if ((sample_size & ~(ix - 1)) == 0) {
break;
}
}
audio_message(LOG_DEBUG, "Sample size is %d %d", ix, m_samples_per_frame);
sample_size = ix;
#if 0
if (m_output_buffer_size_bytes < 4096)
m_output_buffer_size_bytes = 4096;
#endif
if (config.get_config_value(CONFIG_LIMIT_AUDIO_SDL_BUFFER) > 0 &&
sample_size > 4096)
sample_size = 4096;
while (sample_size * m_bytes_per_sample_input > m_sample_buffer_size / 4) {
sample_size /= 2;
}
PaSampleFormat format;
double freq = (double) m_freq;
format = fmt_to_pa_format[m_decode_format];
if (format == 0) {
format = paInt16;
need_convert = true;
}
PaError err;
err = Pa_OpenDefaultStream(&m_pa_stream,
0,
m_channels,
format,
freq,
sample_size,
0,
c_audio_callback,
this);
m_got_channels = m_channels;
if (err != paNoError) {
if (err == paInvalidChannelCount) {
format = paInt16;
err = Pa_OpenDefaultStream(&m_pa_stream,
0,
2,
format,
freq,
sample_size,
0,
c_audio_callback,
this);
m_got_channels = 2;
need_convert = true;
}
if (err != paNoError) {
audio_message(LOG_CRIT, "Couldn't open audio, %s",
Pa_GetErrorText(err));
return (-1);
}
}
audio_message(LOG_DEBUG, "freq %u chans %u samples %u",
m_freq, m_got_channels, sample_size);
m_output_buffer_size_bytes = sample_size * m_bytes_per_sample_input;
if (m_decode_format == AUDIO_FMT_FLOAT || need_convert) {
m_bytes_per_sample_output = sizeof(int16_t) * m_got_channels;
audio_message(LOG_DEBUG, "convert buffer size is %d", m_output_buffer_size_bytes);
audio_convert_init(m_output_buffer_size_bytes, sample_size);
} else {
m_bytes_per_sample_output = m_got_channels * bytes_per_sample;
}
return 1;
}
// Sync task api - initialize the sucker.
// May need to check out non-standard frequencies, see about conversion.
// returns 1 for initialized, -1 for error
int CSDLAudioSync::InitializeHardware (void)
{
uint32_t bytes_per_sample;
SDL_AudioSpec wanted;
memset(&wanted, 0, sizeof(wanted));
wanted.freq = m_freq;
wanted.channels = m_channels;
// bytes per sample is the # of bytes for the output buffer
bytes_per_sample = 2;
switch (m_decode_format) {
case AUDIO_FMT_U8: wanted.format = AUDIO_U8; bytes_per_sample = 1; break;
case AUDIO_FMT_S8: wanted.format = AUDIO_S8; bytes_per_sample = 1; break;
case AUDIO_FMT_U16LSB: wanted.format = AUDIO_U16LSB; break;
case AUDIO_FMT_U16MSB: wanted.format = AUDIO_U16MSB; break;
case AUDIO_FMT_S16LSB: wanted.format = AUDIO_S16LSB; break;
case AUDIO_FMT_S16MSB: wanted.format = AUDIO_S16MSB; break;
case AUDIO_FMT_U16: wanted.format = AUDIO_U16SYS; break;
case AUDIO_FMT_S16: wanted.format = AUDIO_S16SYS; break;
case AUDIO_FMT_FLOAT: wanted.format = AUDIO_S16SYS; break;
case AUDIO_FMT_HW_AC3:
wanted.format = AUDIO_FORMAT_HW_AC3;
bytes_per_sample = 1;
break;
}
uint32_t sample_size;
sample_size = m_samples_per_frame * m_channels;
#ifndef _WIN32
uint32_t ix;
for (ix = 2; ix <= 0x8000; ix <<= 1) {
if ((sample_size & ~(ix - 1)) == 0) {
break;
}
}
ix >>= 1;
audio_message(LOG_DEBUG, "Sample size is %d", ix);
sample_size = ix;
#else
sample_size = 4096;
#endif
#if 0
if (m_output_buffer_size_bytes < 4096)
m_output_buffer_size_bytes = 4096;
#endif
if (config.get_config_value(CONFIG_LIMIT_AUDIO_SDL_BUFFER) > 0 &&
sample_size > 4096)
sample_size = 4096;
if (sample_size < m_freq / 10)
sample_size = 4096;
while (sample_size * m_bytes_per_sample_input > m_sample_buffer_size / 4) {
sample_size /= 2;
}
wanted.samples = sample_size;
wanted.callback = c_audio_callback;
wanted.userdata = this;
#if 1
audio_message(LOG_INFO,
"requested f %d chan %d format %x samples %d",
wanted.freq,
wanted.channels,
wanted.format,
wanted.samples);
#endif
int ret = Our_SDL_OpenAudio(&wanted, &m_obtained);
if (ret < 0) {
if (wanted.channels > 2) {
wanted.channels = 2;
ret = Our_SDL_OpenAudio(&wanted, &m_obtained);
}
if (ret < 0) {
audio_message(LOG_CRIT, "Couldn't open audio, %s", SDL_GetError());
return (-1);
}
}
char buffer[128];
if (Our_SDL_AudioDriverName(buffer, sizeof(buffer)) == NULL) {
strcpy(buffer, "no audio driver");
}
audio_message(LOG_INFO, "got f %d chan %d format %x samples %d size %u %s",
m_obtained.freq,
m_obtained.channels,
m_obtained.format,
m_obtained.samples,
m_obtained.size,
buffer);
#ifdef TEST_MONO_TO_STEREO
#define CHECK_SDL_CHANS_RETURNED TRUE
#define OBTAINED_CHANS 2
#else
#define CHECK_SDL_CHANS_RETURNED m_obtained.channels != m_channels
#define OBTAINED_CHANS m_obtained.channels
#endif
bool need_convert = false;
if (CHECK_SDL_CHANS_RETURNED) {
Our_SDL_CloseAudio();
wanted.channels = OBTAINED_CHANS;
wanted.format = AUDIO_S16SYS; // we're converting, so always choose
m_bytes_per_sample_output = sizeof(int16_t) * wanted.channels;
ret = Our_SDL_OpenAudio(&wanted, &m_obtained);
audio_message(LOG_INFO,
"requested f %d chan %d format %x samples %d",
wanted.freq,
wanted.channels,
wanted.format,
wanted.samples);
if (ret < 0) {
audio_message(LOG_CRIT, "Couldn't reopen audio, %s", SDL_GetError());
return (-1);
}
audio_message(LOG_INFO, "got f %d chan %d format %x samples %d size %u %s",
m_obtained.freq,
m_obtained.channels,
m_obtained.format,
m_obtained.samples,
m_obtained.size,
buffer);
need_convert = true;
} else {
m_bytes_per_sample_output = m_obtained.channels * bytes_per_sample;
}
m_got_channels = m_obtained.channels;
m_output_buffer_size_bytes = m_obtained.size;
if (m_decode_format == AUDIO_FMT_FLOAT || need_convert) {
audio_message(LOG_DEBUG, "convert buffer size is %d", m_obtained.size);
audio_convert_init(m_obtained.size, m_obtained.samples);
}
m_use_SDL_delay = Our_SDL_HasAudioDelay();
if (m_use_SDL_delay)
audio_message(LOG_NOTICE, "Using delay measurement from SDL");
return 1; // check again pretty soon...
}