Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
      12             : #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
      13             : 
      14             : #include <vector>
      15             : 
      16             : #include "webrtc/base/constructormagic.h"
      17             : #include "webrtc/base/scoped_ref_ptr.h"
      18             : #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
      19             : #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
      20             : 
      21             : namespace webrtc {
      22             : 
      23             : struct CodecInst;
      24             : 
      25             : template <typename T>
      26             : class AudioEncoderIsacT final : public AudioEncoder {
      27             :  public:
      28             :   // Allowed combinations of sample rate, frame size, and bit rate are
      29             :   //  - 16000 Hz, 30 ms, 10000-32000 bps
      30             :   //  - 16000 Hz, 60 ms, 10000-32000 bps
      31             :   //  - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
      32           0 :   struct Config {
      33             :     bool IsOk() const;
      34             : 
      35             :     rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo;
      36             : 
      37             :     int payload_type = 103;
      38             :     int sample_rate_hz = 16000;
      39             :     int frame_size_ms = 30;
      40             :     int bit_rate = kDefaultBitRate;  // Limit on the short-term average bit
      41             :                                      // rate, in bits/s.
      42             :     int max_payload_size_bytes = -1;
      43             :     int max_bit_rate = -1;
      44             : 
      45             :     // If true, the encoder will dynamically adjust frame size and bit rate;
      46             :     // the configured values are then merely the starting point.
      47             :     bool adaptive_mode = false;
      48             : 
      49             :     // In adaptive mode, prevent adaptive changes to the frame size. (Not used
      50             :     // in nonadaptive mode.)
      51             :     bool enforce_frame_size = false;
      52             :   };
      53             : 
      54             :   explicit AudioEncoderIsacT(const Config& config);
      55             :   explicit AudioEncoderIsacT(
      56             :       const CodecInst& codec_inst,
      57             :       const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
      58             :   ~AudioEncoderIsacT() override;
      59             : 
      60             :   int SampleRateHz() const override;
      61             :   size_t NumChannels() const override;
      62             :   size_t Num10MsFramesInNextPacket() const override;
      63             :   size_t Max10MsFramesInAPacket() const override;
      64             :   int GetTargetBitrate() const override;
      65             :   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
      66             :                          rtc::ArrayView<const int16_t> audio,
      67             :                          rtc::Buffer* encoded) override;
      68             :   void Reset() override;
      69             : 
      70             :  private:
      71             :   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
      72             :   // STREAM_MAXW16_60MS for iSAC fix (60 ms).
      73             :   static const size_t kSufficientEncodeBufferSizeBytes = 400;
      74             : 
      75             :   static const int kDefaultBitRate = 32000;
      76             : 
      77             :   // Recreate the iSAC encoder instance with the given settings, and save them.
      78             :   void RecreateEncoderInstance(const Config& config);
      79             : 
      80             :   Config config_;
      81             :   typename T::instance_type* isac_state_ = nullptr;
      82             :   rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
      83             : 
      84             :   // Have we accepted input but not yet emitted it in a packet?
      85             :   bool packet_in_progress_ = false;
      86             : 
      87             :   // Timestamp of the first input of the currently in-progress packet.
      88             :   uint32_t packet_timestamp_;
      89             : 
      90             :   // Timestamp of the previously encoded packet.
      91             :   uint32_t last_encoded_timestamp_;
      92             : 
      93             :   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
      94             : };
      95             : 
      96             : }  // namespace webrtc
      97             : 
      98             : #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_