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Change channels number and bytes per second to hardcoded value

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Because these audio players can only handle 16bit mono anyway.
This commit is contained in:
wangqr 2019-10-30 01:16:28 -04:00
parent 620033915a
commit b86238456f
7 changed files with 25 additions and 20 deletions
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7
src/audio_player_alsa.cpp
View file

@ -127,7 +127,7 @@ void AlsaPlayer::PlaybackThread()
do_setup:
snd_pcm_format_t pcm_format;
switch (provider->GetBytesPerSample())
switch (/*provider->GetBytesPerSample()*/ sizeof(int16_t))
{
case 1:
LOG_D("audio/player/alsa") << "format U8";
@ -143,7 +143,7 @@ do_setup:
if (snd_pcm_set_params(pcm,
pcm_format,
SND_PCM_ACCESS_RW_INTERLEAVED,
provider->GetChannels(),
/*provider->GetChannels()*/ 1,
provider->GetSampleRate(),
1, // allow resample
100*1000 // 100 milliseconds latency
@ -151,7 +151,8 @@ do_setup:
return;
LOG_D("audio/player/alsa") << "set pcm params";
size_t framesize = provider->GetChannels() * provider->GetBytesPerSample();
//size_t framesize = provider->GetChannels() * provider->GetBytesPerSample();
size_t framesize = sizeof(int16_t);
while (true)
{

10
src/audio_player_dsound.cpp
View file

@ -112,8 +112,10 @@ DirectSoundPlayer::DirectSoundPlayer(agi::AudioProvider *provider, wxWindow *par
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
//waveFormat.nChannels = provider->GetChannels();
//waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nChannels = 1;
waveFormat.wBitsPerSample = sizeof(int16_t) * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = sizeof(waveFormat);
@ -161,7 +163,7 @@ bool DirectSoundPlayer::FillBuffer(bool fill) {
HRESULT res;
void *ptr1, *ptr2;
unsigned long int size1, size2;
int bytesps = provider->GetBytesPerSample();
int bytesps = /*provider->GetBytesPerSample()*/ sizeof(int16_t);
// To write length
int toWrite = 0;
@ -255,7 +257,7 @@ void DirectSoundPlayer::Play(int64_t start,int64_t count) {
FillBuffer(true);
DWORD play_flag = 0;
if (count*provider->GetBytesPerSample() > bufSize) {
if (count*/*provider->GetBytesPerSample()*/sizeof(int16_t) > bufSize) {
// Start thread
thread = new DirectSoundPlayerThread(this);
thread->Create();

14
src/audio_player_dsound2.cpp
View file

@ -320,8 +320,10 @@ void DirectSoundPlayer2Thread::Run()
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
//waveFormat.nChannels = provider->GetChannels();
//waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nChannels = 1;
waveFormat.wBitsPerSample = sizeof(int16_t) * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = sizeof(waveFormat);
@ -370,7 +372,7 @@ void DirectSoundPlayer2Thread::Run()
DWORD buffer_offset = 0;
bool playback_should_be_running = false;
int current_latency = wanted_latency;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*provider->GetBytesPerSample()/1000;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t)/1000;
while (running)
{
@ -423,7 +425,7 @@ void DirectSoundPlayer2Thread::Run()
if (bytes_filled < wanted_latency_bytes)
{
// Very short playback length, do without streaming playback
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
if (FAILED(bfr->Play(0, 0, 0)))
REPORT_ERROR("Could not start single-buffer playback.")
}
@ -554,7 +556,7 @@ do_fill_buffer:
else if (bytes_filled < wanted_latency_bytes)
{
// Didn't fill as much as we wanted to, let's get back to filling sooner than normal
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
}
else
{
@ -578,7 +580,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
{
// Assume buffers have been locked and are ready to be filled
DWORD bytes_per_frame = provider->GetChannels() * provider->GetBytesPerSample();
DWORD bytes_per_frame = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
DWORD buf1szf = buf1sz / bytes_per_frame;
DWORD buf2szf = buf2sz / bytes_per_frame;

2
src/audio_player_openal.cpp
View file

@ -125,7 +125,7 @@ public:
OpenALPlayer::OpenALPlayer(agi::AudioProvider *provider)
: AudioPlayer(provider)
, samplerate(provider->GetSampleRate())
, bpf(provider->GetChannels() * provider->GetBytesPerSample())
, bpf(/*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t))
{
device = alcOpenDevice(nullptr);
if (!device) throw AudioPlayerOpenError("Failed opening default OpenAL device");

6
src/audio_player_oss.cpp
View file

@ -146,7 +146,7 @@ public:
void OSSPlayer::OpenStream()
{
bpf = provider->GetChannels() * provider->GetBytesPerSample();
bpf = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
// Open device
wxString device = to_wx(OPT_GET("Player/Audio/OSS/Device")->GetString());
@ -162,14 +162,14 @@ void OSSPlayer::OpenStream()
#endif
// Set number of channels
int channels = provider->GetChannels();
int channels = /*provider->GetChannels()*/1;
if (ioctl(dspdev, SNDCTL_DSP_CHANNELS, &channels) < 0) {
throw AudioPlayerOpenError("OSS player: setting channels failed");
}
// Set sample format
int sample_format;
switch (provider->GetBytesPerSample()) {
switch (/*provider->GetBytesPerSample()*/sizeof(int16_t)) {
case 1:
sample_format = AFMT_S8;
break;

2
src/audio_player_portaudio.cpp
View file

@ -140,7 +140,7 @@ void PortAudioPlayer::OpenStream() {
const PaDeviceInfo *device_info = Pa_GetDeviceInfo((*device_ids)[i]);
PaStreamParameters pa_output_p;
pa_output_p.device = (*device_ids)[i];
pa_output_p.channelCount = provider->GetChannels();
pa_output_p.channelCount = /*provider->GetChannels()*/ 1;
pa_output_p.sampleFormat = paInt16;
pa_output_p.suggestedLatency = device_info->defaultLowOutputLatency;
pa_output_p.hostApiSpecificStreamInfo = nullptr;

4
src/audio_player_pulse.cpp
View file

@ -133,11 +133,11 @@ PulseAudioPlayer::PulseAudioPlayer(agi::AudioProvider *provider) : AudioPlayer(p
}
// Set up stream
bpf = provider->GetChannels() * provider->GetBytesPerSample();
bpf = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
pa_sample_spec ss;
ss.format = PA_SAMPLE_S16LE; // FIXME
ss.rate = provider->GetSampleRate();
ss.channels = provider->GetChannels();
ss.channels = /*provider->GetChannels()*/1;
pa_channel_map map;
pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_DEFAULT);