bool enabled;
};
+static const int kAudioProcMaxNativeSampleRateHz = 32000;
+
// The Audio Processing Module (APM) provides a collection of voice processing
// components designed for real-time communications software.
//
// 2. Parameter getters are never called concurrently with the corresponding
// setter.
//
-// APM accepts only 16-bit linear PCM audio data in frames of 10 ms. Multiple
-// channels should be interleaved.
+// APM accepts only linear PCM audio data in chunks of 10 ms. The int16
+// interfaces use interleaved data, while the float interfaces use deinterleaved
+// data.
//
// Usage example, omitting error checking:
// AudioProcessing* apm = AudioProcessing::Create(0);
// Initializes internal states, while retaining all user settings. This
// should be called before beginning to process a new audio stream. However,
// it is not necessary to call before processing the first stream after
- // creation. It is also not necessary to call if the audio parameters (sample
+ // creation.
+ //
+ // It is also not necessary to call if the audio parameters (sample
// rate and number of channels) have changed. Passing updated parameters
// directly to |ProcessStream()| and |AnalyzeReverseStream()| is permissible.
+ // If the parameters are known at init-time though, they may be provided.
virtual int Initialize() = 0;
- virtual int Initialize(int sample_rate_hz,
+
+ // The int16 interfaces require:
+ // - only |NativeRate|s be used
+ // - that the input, output and reverse rates must match
+ // - that |output_layout| matches |input_layout|
+ //
+ // The float interfaces accept arbitrary rates and support differing input
+ // and output layouts, but the output may only remove channels, not add.
+ virtual int Initialize(int input_sample_rate_hz,
+ int output_sample_rate_hz,
int reverse_sample_rate_hz,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_channels) = 0;
+ ChannelLayout input_layout,
+ ChannelLayout output_layout,
+ ChannelLayout reverse_layout) = 0;
// Pass down additional options which don't have explicit setters. This
// ensures the options are applied immediately.
virtual int EnableExperimentalNs(bool enable) = 0;
virtual bool experimental_ns_enabled() const = 0;
- // DEPRECATED: It is now possible to modify the sample rate directly in a call
- // to |ProcessStream|.
- // Sets the sample |rate| in Hz for both the primary and reverse audio
- // streams. 8000, 16000 or 32000 Hz are permitted.
+ // DEPRECATED.
+ // TODO(ajm): Remove after Chromium has upgraded to using Initialize().
virtual int set_sample_rate_hz(int rate) = 0;
+ // TODO(ajm): Remove after voice engine no longer requires it to resample
+ // the reverse stream to the forward rate.
+ virtual int input_sample_rate_hz() const = 0;
+ // TODO(ajm): Remove after Chromium no longer depends on it.
virtual int sample_rate_hz() const = 0;
- virtual int split_sample_rate_hz() const = 0;
-
- // DEPRECATED: It is now possible to modify the number of channels directly in
- // a call to |ProcessStream|.
- // Sets the number of channels for the primary audio stream. Input frames must
- // contain a number of channels given by |input_channels|, while output frames
- // will be returned with number of channels given by |output_channels|.
- virtual int set_num_channels(int input_channels, int output_channels) = 0;
+
+ // TODO(ajm): Only intended for internal use. Make private and friend the
+ // necessary classes?
+ virtual int proc_sample_rate_hz() const = 0;
+ virtual int proc_split_sample_rate_hz() const = 0;
virtual int num_input_channels() const = 0;
virtual int num_output_channels() const = 0;
-
- // DEPRECATED: It is now possible to modify the number of channels directly in
- // a call to |AnalyzeReverseStream|.
- // Sets the number of channels for the reverse audio stream. Input frames must
- // contain a number of channels given by |channels|.
- virtual int set_num_reverse_channels(int channels) = 0;
virtual int num_reverse_channels() const = 0;
// Set to true when the output of AudioProcessing will be muted or in some
virtual int ProcessStream(AudioFrame* frame) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
- // of |data| points to a channel buffer, arranged according to
+ // of |src| points to a channel buffer, arranged according to
// |input_layout|. At output, the channels will be arranged according to
- // |output_layout|.
- // TODO(ajm): Output layout conversion does not yet work.
- virtual int ProcessStream(float* const* data,
+ // |output_layout| at |output_sample_rate_hz| in |dest|.
+ //
+ // The output layout may only remove channels, not add. |src| and |dest|
+ // may use the same memory, if desired.
+ virtual int ProcessStream(const float* const* src,
int samples_per_channel,
- int sample_rate_hz,
+ int input_sample_rate_hz,
ChannelLayout input_layout,
- ChannelLayout output_layout) = 0;
+ int output_sample_rate_hz,
+ ChannelLayout output_layout,
+ float* const* dest) = 0;
// Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame
// will not be modified. On the client-side, this is the far-end (or to be
//
// The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_|
// members of |frame| must be valid. |sample_rate_hz_| must correspond to
- // |sample_rate_hz()|
+ // |input_sample_rate_hz()|
//
// TODO(ajm): add const to input; requires an implementation fix.
virtual int AnalyzeReverseStream(AudioFrame* frame) = 0;
kBadStreamParameterWarning = -13
};
- enum {
+ enum NativeRate {
kSampleRate8kHz = 8000,
kSampleRate16kHz = 16000,
kSampleRate32kHz = 32000
};
+
+ static const int kChunkSizeMs = 10;
};
// The acoustic echo cancellation (AEC) component provides better performance
// render and capture devices are used, particularly with webcams.
//
// This enables a compensation mechanism, and requires that
- // |set_device_sample_rate_hz()| and |set_stream_drift_samples()| be called.
+ // set_stream_drift_samples() be called.
virtual int enable_drift_compensation(bool enable) = 0;
virtual bool is_drift_compensation_enabled() const = 0;
- // Provides the sampling rate of the audio devices. It is assumed the render
- // and capture devices use the same nominal sample rate. Required if and only
- // if drift compensation is enabled.
- virtual int set_device_sample_rate_hz(int rate) = 0;
- virtual int device_sample_rate_hz() const = 0;
-
// Sets the difference between the number of samples rendered and captured by
// the audio devices since the last call to |ProcessStream()|. Must be called
// if drift compensation is enabled, prior to |ProcessStream()|.
// frames since the last call to RMS(). The returned value is positive but
// should be interpreted as negative. It is constrained to [0, 127].
//
- // The computation follows:
- // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-05
+ // The computation follows: https://tools.ietf.org/html/rfc6465
// with the intent that it can provide the RTP audio level indication.
//
// Frames passed to ProcessStream() with an |_energy| of zero are considered
projects / platform / framework / web / crosswalk.git / blobdiff