LCOV - code coverage report
Current view: top level - media/webrtc/trunk/webrtc/modules/audio_coding/codecs - legacy_encoded_audio_frame.h (source / functions) Hit Total Coverage
Test: output.info Lines: 0 1 0.0 %
Date: 2017-07-14 16:53:18 Functions: 0 2 0.0 %
Legend: Lines: hit not hit

          Line data    Source code
       1             : /*
       2             :  *  Copyright (c) 2016 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_LEGACY_ENCODED_AUDIO_FRAME_H_
      12             : #define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
      13             : 
      14             : #include <vector>
      15             : 
      16             : #include "webrtc/base/array_view.h"
      17             : #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
      18             : 
      19             : namespace webrtc {
      20             : 
      21           0 : class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
      22             :  public:
      23             :   LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);
      24             :   ~LegacyEncodedAudioFrame() override;
      25             : 
      26             :   static std::vector<AudioDecoder::ParseResult> SplitBySamples(
      27             :       AudioDecoder* decoder,
      28             :       rtc::Buffer&& payload,
      29             :       uint32_t timestamp,
      30             :       size_t bytes_per_ms,
      31             :       uint32_t timestamps_per_ms);
      32             : 
      33             :   size_t Duration() const override;
      34             : 
      35             :   rtc::Optional<DecodeResult> Decode(
      36             :       rtc::ArrayView<int16_t> decoded) const override;
      37             : 
      38             :   // For testing:
      39             :   const rtc::Buffer& payload() const { return payload_; }
      40             : 
      41             :  private:
      42             :   AudioDecoder* const decoder_;
      43             :   const rtc::Buffer payload_;
      44             : };
      45             : 
      46             : }  // namespace webrtc
      47             : 
      48             : #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_

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