webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
Jonas Olsson a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00

76 lines
2.4 KiB
C++

/*
* Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
#include <stddef.h>
#include <stdint.h>
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class AudioDecoderPcmU final : public AudioDecoder {
public:
explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
RTC_DCHECK_GE(num_channels, 1);
}
void Reset() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp) override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
const size_t num_channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
};
class AudioDecoderPcmA final : public AudioDecoder {
public:
explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
RTC_DCHECK_GE(num_channels, 1);
}
void Reset() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp) override;
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
int SampleRateHz() const override;
size_t Channels() const override;
protected:
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
const size_t num_channels_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_