nsresult
BufferedAudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioChannelType)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = aNumChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = ConvertChannelToCubebType(aAudioChannelType);
#else
params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
#endif
if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
return NS_ERROR_INVALID_ARG;
}
#endif
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * aNumChannels;
mAudioClock.Init();
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "BufferedAudioStream", params,
GetCubebLatency(), DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
mCubebStream.own(stream);
}
}
if (!mCubebStream) {
return NS_ERROR_FAILURE;
}
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = FramesToBytes(aRate);
NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
return NS_OK;
}
// This code used to live inside AudioStream::Init(), but on Mac (others?)
// it has been known to take 300-800 (or even 8500) ms to execute(!)
nsresult
AudioStream::OpenCubeb(cubeb_stream_params &aParams,
LatencyRequest aLatencyRequest)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext) {
MonitorAutoLock mon(mMonitor);
mState = AudioStream::ERRORED;
return NS_ERROR_FAILURE;
}
// If the latency pref is set, use it. Otherwise, if this stream is intended
// for low latency playback, try to get the lowest latency possible.
// Otherwise, for normal streams, use 100ms.
uint32_t latency;
if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
latency = GetCubebLatency();
}
} else {
latency = GetCubebLatency();
}
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
MonitorAutoLock mon(mMonitor);
mCubebStream.own(stream);
// Make sure we weren't shut down while in flight!
if (mState == SHUTDOWN) {
mCubebStream.reset();
LOG(("AudioStream::OpenCubeb() %p Shutdown while opening cubeb", this));
return NS_ERROR_FAILURE;
}
// We can't cubeb_stream_start() the thread from a transient thread due to
// cubeb API requirements (init can be called from another thread, but
// not start/stop/destroy/etc)
} else {
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
LOG(("AudioStream::OpenCubeb() %p failed to init cubeb", this));
return NS_ERROR_FAILURE;
}
}
if (!mStartTime.IsNull()) {
TimeDuration timeDelta = TimeStamp::Now() - mStartTime;
LOG(("AudioStream creation time %sfirst: %u ms", mIsFirst ? "" : "not ",
(uint32_t) timeDelta.ToMilliseconds()));
Telemetry::Accumulate(mIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
}
return NS_OK;
}
/*static*/ int AudioStream::MaxNumberOfChannels()
{
cubeb* cubebContext = GetCubebContext();
uint32_t maxNumberOfChannels;
if (cubebContext &&
cubeb_get_max_channel_count(cubebContext,
&maxNumberOfChannels) == CUBEB_OK) {
return static_cast<int>(maxNumberOfChannels);
}
return 0;
}
int AudioStream::MaxNumberOfChannels()
{
uint32_t maxNumberOfChannels, rv;
rv = cubeb_get_max_channel_count(GetCubebContext(), &maxNumberOfChannels);
if (rv != CUBEB_OK) {
return 0;
}
return static_cast<int>(maxNumberOfChannels);
}
uint32_t MaxNumberOfChannels()
{
cubeb* cubebContext = GetCubebContext();
uint32_t maxNumberOfChannels;
if (cubebContext &&
cubeb_get_max_channel_count(cubebContext,
&maxNumberOfChannels) == CUBEB_OK) {
return maxNumberOfChannels;
}
return 0;
}
void GetCurrentBackend(nsAString& aBackend)
{
cubeb* cubebContext = GetCubebContext();
if (cubebContext) {
const char* backend = cubeb_get_backend_id(cubebContext);
if (backend) {
aBackend.AssignASCII(backend);
return;
}
}
aBackend.AssignLiteral("unknown");
}
// This code used to live inside AudioStream::Init(), but on Mac (others?)
// it has been known to take 300-800 (or even 8500) ms to execute(!)
nsresult
AudioStream::OpenCubeb(cubeb_stream_params &aParams,
LatencyRequest aLatencyRequest)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext) {
MonitorAutoLock mon(mMonitor);
mState = AudioStream::ERRORED;
return NS_ERROR_FAILURE;
}
// If the latency pref is set, use it. Otherwise, if this stream is intended
// for low latency playback, try to get the lowest latency possible.
// Otherwise, for normal streams, use 100ms.
uint32_t latency;
if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
latency = GetCubebLatency();
}
} else {
latency = GetCubebLatency();
}
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
MonitorAutoLock mon(mMonitor);
mCubebStream.own(stream);
// Make sure we weren't shut down while in flight!
if (mState == SHUTDOWN) {
mCubebStream.reset();
LOG(("AudioStream::OpenCubeb() %p Shutdown while opening cubeb", this));
return NS_ERROR_FAILURE;
}
// We can't cubeb_stream_start() the thread from a transient thread due to
// cubeb API requirements (init can be called from another thread, but
// not start/stop/destroy/etc)
} else {
MonitorAutoLock mon(mMonitor);
mState = ERRORED;
LOG(("AudioStream::OpenCubeb() %p failed to init cubeb", this));
return NS_ERROR_FAILURE;
}
}
return NS_OK;
}
nsresult
nsBufferedAudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioChannelType)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
mRate = aRate;
mChannels = aNumChannels;
cubeb_stream_params params;
params.rate = aRate;
params.channels = aNumChannels;
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * aNumChannels;
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "nsBufferedAudioStream", params,
GetCubebLatency(), DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
mCubebStream.own(stream);
}
}
if (!mCubebStream) {
return NS_ERROR_FAILURE;
}
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = aRate * mBytesPerFrame;
NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
return NS_OK;
}
// NOTE: this must not block a LowLatency stream for any significant amount
// of time, or it will block the entirety of MSG
nsresult
AudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioChannel,
LatencyRequest aLatencyRequest)
{
mStartTime = TimeStamp::Now();
mIsFirst = GetFirstStream();
if (!GetCubebContext() || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
("%s channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
mLatencyRequest = aLatencyRequest;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = mOutChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = ConvertChannelToCubebType(aAudioChannel);
#else
params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
#endif
if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
return NS_ERROR_INVALID_ARG;
}
#endif
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
mAudioClock.Init();
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = FramesToBytes(aRate);
NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
if (aLatencyRequest == LowLatency) {
// Don't block this thread to initialize a cubeb stream.
// When this is done, it will start callbacks from Cubeb. Those will
// cause us to move from INITIALIZED to RUNNING. Until then, we
// can't access any cubeb functions.
// Use a RefPtr to avoid leaks if Dispatch fails
RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
init->Dispatch();
return NS_OK;
}
// High latency - open synchronously
nsresult rv = OpenCubeb(params, aLatencyRequest);
// See if we need to start() the stream, since we must do that from this
// thread for now (cubeb API issue)
{
MonitorAutoLock mon(mMonitor);
CheckForStart();
}
return rv;
}
nsresult
AudioStream::Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioChannelType,
LatencyRequest aLatencyRequest)
{
cubeb* cubebContext = GetCubebContext();
if (!cubebContext || aNumChannels < 0 || aRate < 0) {
return NS_ERROR_FAILURE;
}
PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
("%s channels: %d, rate: %d", __FUNCTION__, aNumChannels, aRate));
mInRate = mOutRate = aRate;
mChannels = aNumChannels;
mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
mLatencyRequest = aLatencyRequest;
mDumpFile = OpenDumpFile(this);
cubeb_stream_params params;
params.rate = aRate;
params.channels = mOutChannels;
#if defined(__ANDROID__)
#if defined(MOZ_B2G)
params.stream_type = ConvertChannelToCubebType(aAudioChannelType);
#else
params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
#endif
if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
return NS_ERROR_INVALID_ARG;
}
#endif
if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
params.format = CUBEB_SAMPLE_S16NE;
} else {
params.format = CUBEB_SAMPLE_FLOAT32NE;
}
mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
mAudioClock.Init();
// If the latency pref is set, use it. Otherwise, if this stream is intended
// for low latency playback, try to get the lowest latency possible.
// Otherwise, for normal streams, use 100ms.
uint32_t latency;
if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
if (cubeb_get_min_latency(cubebContext, params, &latency) != CUBEB_OK) {
latency = GetCubebLatency();
}
} else {
latency = GetCubebLatency();
}
{
cubeb_stream* stream;
if (cubeb_stream_init(cubebContext, &stream, "AudioStream", params,
latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
mCubebStream.own(stream);
}
}
if (!mCubebStream) {
return NS_ERROR_FAILURE;
}
// Size mBuffer for one second of audio. This value is arbitrary, and was
// selected based on the observed behaviour of the existing AudioStream
// implementations.
uint32_t bufferLimit = FramesToBytes(aRate);
NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
mBuffer.SetCapacity(bufferLimit);
// Start the stream right away when low latency has been requested. This means
// that the DataCallback will feed silence to cubeb, until the first frames
// are writtent to this AudioStream.
if (mLatencyRequest == LowLatency) {
Start();
}
return NS_OK;
}