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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
124 lines
3.5 KiB
C++
124 lines
3.5 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_encoder.h"
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#include "webrtc/rtc_base/constructormagic.h"
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namespace webrtc {
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class AudioEncoderPcm : public AudioEncoder {
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public:
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struct Config {
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public:
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bool IsOk() const;
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int frame_size_ms;
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size_t num_channels;
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int payload_type;
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protected:
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explicit Config(int pt)
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: frame_size_ms(20), num_channels(1), payload_type(pt) {}
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};
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~AudioEncoderPcm() override;
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int SampleRateHz() const override;
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size_t NumChannels() const override;
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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int GetTargetBitrate() const override;
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void Reset() override;
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protected:
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AudioEncoderPcm(const Config& config, int sample_rate_hz);
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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virtual size_t EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) = 0;
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virtual size_t BytesPerSample() const = 0;
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// Used to set EncodedInfoLeaf::encoder_type in
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// AudioEncoderPcm::EncodeImpl
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virtual AudioEncoder::CodecType GetCodecType() const = 0;
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private:
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const int sample_rate_hz_;
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const size_t num_channels_;
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const int payload_type_;
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const size_t num_10ms_frames_per_packet_;
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const size_t full_frame_samples_;
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std::vector<int16_t> speech_buffer_;
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uint32_t first_timestamp_in_buffer_;
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};
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struct CodecInst;
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class AudioEncoderPcmA final : public AudioEncoderPcm {
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public:
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struct Config : public AudioEncoderPcm::Config {
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Config() : AudioEncoderPcm::Config(8) {}
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};
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explicit AudioEncoderPcmA(const Config& config)
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: AudioEncoderPcm(config, kSampleRateHz) {}
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explicit AudioEncoderPcmA(const CodecInst& codec_inst);
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protected:
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size_t EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) override;
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size_t BytesPerSample() const override;
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AudioEncoder::CodecType GetCodecType() const override;
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private:
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static const int kSampleRateHz = 8000;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
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};
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class AudioEncoderPcmU final : public AudioEncoderPcm {
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public:
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struct Config : public AudioEncoderPcm::Config {
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Config() : AudioEncoderPcm::Config(0) {}
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};
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explicit AudioEncoderPcmU(const Config& config)
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: AudioEncoderPcm(config, kSampleRateHz) {}
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explicit AudioEncoderPcmU(const CodecInst& codec_inst);
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protected:
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size_t EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) override;
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size_t BytesPerSample() const override;
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AudioEncoder::CodecType GetCodecType() const override;
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private:
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static const int kSampleRateHz = 8000;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
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