Add new method AcmReceiver::last_packet_sample_rate_hz()

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

 /*
  *  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_MAIN_ACM2_ACM_RECEIVER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
 
 #include <map>
 #include <vector>
 
 #include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module_common_types.h"
@@ skipping 119 matching lines @@
   // this is before applying any user-defined limits (specified by calling
   // (SetMinimumDelay() and/or SetMaximumDelay()).
   //
   int LeastRequiredDelayMs() const;
 
   //
   // Resets the initial delay to zero.
   //
   void ResetInitialDelay();
 
+  // Returns the sample rate of of the codec associated with the last incoming

[kwiberg-webrtc, 2015/11/23 13:03:50]

Even number of ofs; odd number expected.

[hlundin-webrtc, 2015/11/23 13:50:45]

Done.

Took care of of of off-by-one error.

+  // packet. If no packet of a registered non-CNG codec has been received, the
+  // return value is empty. Also, if the codec was unregistered since the last

[kwiberg-webrtc, 2015/11/23 13:03:50]

"codec" -> "decoder", maybe?

[hlundin-webrtc, 2015/11/23 13:50:45]

Done.

+  // packet was inserted, the return value is empty.
+  rtc::Optional<int> last_packet_sample_rate_hz() const;
+
   // Returns last_output_sample_rate_hz from the NetEq instance.
   int last_output_sample_rate_hz() const;
 
   //
   // Get the current network statistics from NetEq.
   //
   // Output:
   //   - statistics           : The current network statistics.
   //
   void GetNetworkStatistics(NetworkStatistics* statistics);
@@ skipping 39 matching lines @@
   //
   void set_id(int id);  // TODO(turajs): can be inline.
 
   //
   // Gets the RTP timestamp of the last sample delivered by GetAudio().
   // Returns true if the RTP timestamp is valid, otherwise false.
   //
   bool GetPlayoutTimestamp(uint32_t* timestamp);
 
   //
-  // Return the index of the codec associated with the last non-CNG/non-DTMF
-  // received payload. If no non-CNG/non-DTMF payload is received -1 is
-  // returned.
-  //
-  int last_audio_codec_id() const;  // TODO(turajs): can be inline.
-
-  //
   // Get the audio codec associated with the last non-CNG/non-DTMF received
   // payload. If no non-CNG/non-DTMF packet is received -1 is returned,
   // otherwise return 0.
   //
   int LastAudioCodec(CodecInst* codec) const;
 
   //
   // Get a decoder given its registered payload-type.
   //
   // Input:
@@ skipping 55 matching lines @@
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
   rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
   std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
+  rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_