Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 95af8f5..3a0ad24 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -19,6 +19,7 @@
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
+
class AudioBuffer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
@@ -39,6 +40,44 @@
} // namespace audioproc
#endif
+class AudioRate {
+ public:
+ explicit AudioRate(int sample_rate_hz)
+ : rate_(sample_rate_hz),
+ samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {}
+ virtual ~AudioRate() {}
+
+ void set(int rate) {
+ rate_ = rate;
+ samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
+ }
+
+ int rate() const { return rate_; }
+ int samples_per_channel() const { return samples_per_channel_; }
+
+ private:
+ int rate_;
+ int samples_per_channel_;
+};
+
+class AudioFormat : public AudioRate {
+ public:
+ AudioFormat(int sample_rate_hz, int num_channels)
+ : AudioRate(sample_rate_hz),
+ num_channels_(num_channels) {}
+ virtual ~AudioFormat() {}
+
+ void set(int rate, int num_channels) {
+ AudioRate::set(rate);
+ num_channels_ = num_channels;
+ }
+
+ int num_channels() const { return num_channels_; }
+
+ private:
+ int num_channels_;
+};
+
class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
@@ -46,33 +85,34 @@
// AudioProcessing methods.
virtual int Initialize() OVERRIDE;
- virtual int Initialize(int sample_rate_hz,
+ 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) OVERRIDE;
+ ChannelLayout input_layout,
+ ChannelLayout output_layout,
+ ChannelLayout reverse_layout) OVERRIDE;
virtual void SetExtraOptions(const Config& config) OVERRIDE;
virtual int EnableExperimentalNs(bool enable) OVERRIDE;
virtual bool experimental_ns_enabled() const OVERRIDE {
return false;
}
virtual int set_sample_rate_hz(int rate) OVERRIDE;
- virtual int sample_rate_hz() const OVERRIDE;
- virtual int split_sample_rate_hz() const OVERRIDE;
- virtual int set_num_channels(int input_channels,
- int output_channels) OVERRIDE;
+ virtual int input_sample_rate_hz() const OVERRIDE;
+ virtual int proc_sample_rate_hz() const OVERRIDE;
+ virtual int proc_split_sample_rate_hz() const OVERRIDE;
virtual int num_input_channels() const OVERRIDE;
virtual int num_output_channels() const OVERRIDE;
- virtual int set_num_reverse_channels(int channels) OVERRIDE;
virtual int num_reverse_channels() const OVERRIDE;
virtual void set_output_will_be_muted(bool muted) OVERRIDE;
virtual bool output_will_be_muted() const OVERRIDE;
virtual int ProcessStream(AudioFrame* frame) OVERRIDE;
- virtual int ProcessStream(float* const* data,
+ 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) OVERRIDE;
+ int output_sample_rate_hz,
+ ChannelLayout output_layout,
+ float* const* dest) OVERRIDE;
virtual int AnalyzeReverseStream(AudioFrame* frame) OVERRIDE;
virtual int AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
@@ -102,12 +142,14 @@
virtual int InitializeLocked();
private:
- int InitializeLocked(int sample_rate_hz,
+ int InitializeLocked(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);
- int MaybeInitializeLocked(int sample_rate_hz,
+ int MaybeInitializeLocked(int input_sample_rate_hz,
+ int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
@@ -130,8 +172,8 @@
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
- AudioBuffer* render_audio_;
- AudioBuffer* capture_audio_;
+ scoped_ptr<AudioBuffer> render_audio_;
+ scoped_ptr<AudioBuffer> capture_audio_;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
@@ -142,22 +184,22 @@
std::string event_str_; // Memory for protobuf serialization.
#endif
- int sample_rate_hz_;
- int reverse_sample_rate_hz_;
- int split_sample_rate_hz_;
- int samples_per_channel_;
- int reverse_samples_per_channel_;
+ AudioFormat fwd_in_format_;
+ AudioFormat fwd_proc_format_;
+ AudioRate fwd_out_format_;
+ AudioFormat rev_in_format_;
+ AudioFormat rev_proc_format_;
+ int split_rate_;
+
int stream_delay_ms_;
int delay_offset_ms_;
bool was_stream_delay_set_;
- int num_reverse_channels_;
- int num_input_channels_;
- int num_output_channels_;
bool output_will_be_muted_;
bool key_pressed_;
};
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_