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/*
* 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.
*/
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include <cstddef>
#include <cstdint>
#include <optional>
#include <utility>
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/time_delta.h"
#include "modules/audio_coding/codecs/g711/g711_interface.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
bool AudioEncoderPcm::Config::IsOk() const {
return (frame_size_ms % 10 == 0) && (num_channels >= 1) &&
(num_channels <= AudioEncoder::kMaxNumberOfChannels);
}
AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
: sample_rate_hz_(sample_rate_hz),
num_channels_(config.num_channels),
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
full_frame_samples_(config.num_channels * config.frame_size_ms *
sample_rate_hz / 1000),
first_timestamp_in_buffer_(0) {
RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
speech_buffer_.reserve(full_frame_samples_);
}
AudioEncoderPcm::~AudioEncoderPcm() = default;
int AudioEncoderPcm::SampleRateHz() const {
return sample_rate_hz_;
}
size_t AudioEncoderPcm::NumChannels() const {
return num_channels_;
}
size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderPcm::GetTargetBitrate() const {
return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
NumChannels());
}
AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
uint32_t rtp_timestamp,
ArrayView<const int16_t> audio,
Buffer* encoded) {
if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = rtp_timestamp;
}
speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
if (speech_buffer_.size() < full_frame_samples_) {
return EncodedInfo();
}
RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
EncodedInfo info;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.encoded_bytes = encoded->AppendData(
full_frame_samples_ * BytesPerSample(), [&](ArrayView<uint8_t> encoded) {
return EncodeCall(&speech_buffer_[0], full_frame_samples_,
encoded.data());
});
speech_buffer_.clear();
info.encoder_type = GetCodecType();
return info;
}
void AudioEncoderPcm::Reset() {
speech_buffer_.clear();
}
std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderPcm::GetFrameLengthRange() const {
return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
}
size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeA(audio, input_len, encoded);
}
size_t AudioEncoderPcmA::BytesPerSample() const {
return 1;
}
AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
return AudioEncoder::CodecType::kPcmA;
}
size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) {
return WebRtcG711_EncodeU(audio, input_len, encoded);
}
size_t AudioEncoderPcmU::BytesPerSample() const {
return 1;
}
AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
return AudioEncoder::CodecType::kPcmU;
}
} // namespace webrtc
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