Index: webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h |
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h |
deleted file mode 100644 |
index 76eb59432f2e16314fab603847735d205d4c59de..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h |
+++ /dev/null |
@@ -1,117 +0,0 @@ |
-/* |
- * Copyright (c) 2014 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_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_ |
- |
-#include <vector> |
- |
-#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
- |
-namespace webrtc { |
- |
-class AudioEncoderPcm : public AudioEncoder { |
- public: |
- struct Config { |
- public: |
- bool IsOk() const; |
- |
- int frame_size_ms; |
- int num_channels; |
- int payload_type; |
- |
- protected: |
- explicit Config(int pt) |
- : frame_size_ms(20), num_channels(1), payload_type(pt) {} |
- }; |
- |
- ~AudioEncoderPcm() override; |
- |
- size_t MaxEncodedBytes() const override; |
- int SampleRateHz() const override; |
- int NumChannels() const override; |
- size_t Num10MsFramesInNextPacket() const override; |
- size_t Max10MsFramesInAPacket() const override; |
- int GetTargetBitrate() const override; |
- EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
- rtc::ArrayView<const int16_t> audio, |
- size_t max_encoded_bytes, |
- uint8_t* encoded) override; |
- void Reset() override; |
- |
- protected: |
- AudioEncoderPcm(const Config& config, int sample_rate_hz); |
- |
- virtual size_t EncodeCall(const int16_t* audio, |
- size_t input_len, |
- uint8_t* encoded) = 0; |
- |
- virtual int BytesPerSample() const = 0; |
- |
- private: |
- const int sample_rate_hz_; |
- const int num_channels_; |
- const int payload_type_; |
- const size_t num_10ms_frames_per_packet_; |
- const size_t full_frame_samples_; |
- std::vector<int16_t> speech_buffer_; |
- uint32_t first_timestamp_in_buffer_; |
-}; |
- |
-struct CodecInst; |
- |
-class AudioEncoderPcmA final : public AudioEncoderPcm { |
- public: |
- struct Config : public AudioEncoderPcm::Config { |
- Config() : AudioEncoderPcm::Config(8) {} |
- }; |
- |
- explicit AudioEncoderPcmA(const Config& config) |
- : AudioEncoderPcm(config, kSampleRateHz) {} |
- explicit AudioEncoderPcmA(const CodecInst& codec_inst); |
- |
- protected: |
- size_t EncodeCall(const int16_t* audio, |
- size_t input_len, |
- uint8_t* encoded) override; |
- |
- int BytesPerSample() const override; |
- |
- private: |
- static const int kSampleRateHz = 8000; |
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA); |
-}; |
- |
-class AudioEncoderPcmU final : public AudioEncoderPcm { |
- public: |
- struct Config : public AudioEncoderPcm::Config { |
- Config() : AudioEncoderPcm::Config(0) {} |
- }; |
- |
- explicit AudioEncoderPcmU(const Config& config) |
- : AudioEncoderPcm(config, kSampleRateHz) {} |
- explicit AudioEncoderPcmU(const CodecInst& codec_inst); |
- |
- protected: |
- size_t EncodeCall(const int16_t* audio, |
- size_t input_len, |
- uint8_t* encoded) override; |
- |
- int BytesPerSample() const override; |
- |
- private: |
- static const int kSampleRateHz = 8000; |
- RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU); |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_ |