Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic.h b/webrtc/modules/audio_coding/neteq/decision_logic.h
new file mode 100644
index 0000000..672ce93
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/decision_logic.h
@@ -0,0 +1,168 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/defines.h"
+#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class BufferLevelFilter;
+class DecoderDatabase;
+class DelayManager;
+class Expand;
+class PacketBuffer;
+class SyncBuffer;
+struct RTPHeader;
+
+// This is the base class for the decision tree implementations. Derived classes
+// must implement the method GetDecisionSpecialized().
+class DecisionLogic {
+ public:
+ // Static factory function which creates different types of objects depending
+ // on the |playout_mode|.
+ static DecisionLogic* Create(int fs_hz,
+ int output_size_samples,
+ NetEqPlayoutMode playout_mode,
+ DecoderDatabase* decoder_database,
+ const PacketBuffer& packet_buffer,
+ DelayManager* delay_manager,
+ BufferLevelFilter* buffer_level_filter);
+
+ // Constructor.
+ DecisionLogic(int fs_hz,
+ int output_size_samples,
+ NetEqPlayoutMode playout_mode,
+ DecoderDatabase* decoder_database,
+ const PacketBuffer& packet_buffer,
+ DelayManager* delay_manager,
+ BufferLevelFilter* buffer_level_filter);
+
+ // Destructor.
+ virtual ~DecisionLogic() {}
+
+ // Resets object to a clean state.
+ void Reset();
+
+ // Resets parts of the state. Typically done when switching codecs.
+ void SoftReset();
+
+ // Sets the sample rate and the output block size.
+ void SetSampleRate(int fs_hz, int output_size_samples);
+
+ // Returns the operation that should be done next. |sync_buffer| and |expand|
+ // are provided for reference. |decoder_frame_length| is the number of samples
+ // obtained from the last decoded frame. If there is a packet available, the
+ // packet header should be supplied in |packet_header|; otherwise it should
+ // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
+ // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
+ // should be set to true. The output variable |reset_decoder| will be set to
+ // true if a reset is required; otherwise it is left unchanged (i.e., it can
+ // remain true if it was true before the call).
+ // This method end with calling GetDecisionSpecialized to get the actual
+ // return value.
+ Operations GetDecision(const SyncBuffer& sync_buffer,
+ const Expand& expand,
+ int decoder_frame_length,
+ const RTPHeader* packet_header,
+ Modes prev_mode,
+ bool play_dtmf,
+ bool* reset_decoder);
+
+ // These methods test the |cng_state_| for different conditions.
+ bool CngRfc3389On() const { return cng_state_ == kCngRfc3389On; }
+ bool CngOff() const { return cng_state_ == kCngOff; }
+
+ // Resets the |cng_state_| to kCngOff.
+ void SetCngOff() { cng_state_ = kCngOff; }
+
+ // Reports back to DecisionLogic whether the decision to do expand remains or
+ // not. Note that this is necessary, since an expand decision can be changed
+ // to kNormal in NetEqImpl::GetDecision if there is still enough data in the
+ // sync buffer.
+ virtual void ExpandDecision(Operations operation);
+
+ // Adds |value| to |sample_memory_|.
+ void AddSampleMemory(int32_t value) {
+ sample_memory_ += value;
+ }
+
+ // Accessors and mutators.
+ void set_sample_memory(int32_t value) { sample_memory_ = value; }
+ int generated_noise_samples() const { return generated_noise_samples_; }
+ void set_generated_noise_samples(int value) {
+ generated_noise_samples_ = value;
+ }
+ int packet_length_samples() const { return packet_length_samples_; }
+ void set_packet_length_samples(int value) {
+ packet_length_samples_ = value;
+ }
+ void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
+ NetEqPlayoutMode playout_mode() const { return playout_mode_; }
+
+ protected:
+ // The value 6 sets maximum time-stretch rate to about 100 ms/s.
+ static const int kMinTimescaleInterval = 6;
+
+ enum CngState {
+ kCngOff,
+ kCngRfc3389On,
+ kCngInternalOn
+ };
+
+ // Returns the operation that should be done next. |sync_buffer| and |expand|
+ // are provided for reference. |decoder_frame_length| is the number of samples
+ // obtained from the last decoded frame. If there is a packet available, the
+ // packet header should be supplied in |packet_header|; otherwise it should
+ // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
+ // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
+ // should be set to true. The output variable |reset_decoder| will be set to
+ // true if a reset is required; otherwise it is left unchanged (i.e., it can
+ // remain true if it was true before the call).
+ // Should be implemented by derived classes.
+ virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
+ const Expand& expand,
+ int decoder_frame_length,
+ const RTPHeader* packet_header,
+ Modes prev_mode,
+ bool play_dtmf,
+ bool* reset_decoder) = 0;
+
+ // Updates the |buffer_level_filter_| with the current buffer level
+ // |buffer_size_packets|.
+ void FilterBufferLevel(int buffer_size_packets, Modes prev_mode);
+
+ DecoderDatabase* decoder_database_;
+ const PacketBuffer& packet_buffer_;
+ DelayManager* delay_manager_;
+ BufferLevelFilter* buffer_level_filter_;
+ int fs_mult_;
+ int output_size_samples_;
+ CngState cng_state_; // Remember if comfort noise is interrupted by other
+ // event (e.g., DTMF).
+ int generated_noise_samples_;
+ int packet_length_samples_;
+ int sample_memory_;
+ bool prev_time_scale_;
+ int timescale_hold_off_;
+ int num_consecutive_expands_;
+ const NetEqPlayoutMode playout_mode_;
+
+ private:
+ DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_