| /* |
| * 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 <cstdint> |
| |
| #include "modules/audio_coding/codecs/g711/g711_interface.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| bool AudioEncoderPcm::Config::IsOk() const { |
| return (frame_size_ms % 10 == 0) && (num_channels >= 1); |
| } |
| |
| 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, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::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(), |
| [&](rtc::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(); |
| } |
| |
| absl::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 |