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7e5dfdbca3
The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
126 lines
3.9 KiB
C++
126 lines
3.9 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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#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
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#include <cstdint>
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#include "modules/audio_coding/codecs/g711/g711_interface.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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bool AudioEncoderPcm::Config::IsOk() const {
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return (frame_size_ms % 10 == 0) && (num_channels >= 1);
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}
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AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
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: sample_rate_hz_(sample_rate_hz),
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num_channels_(config.num_channels),
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payload_type_(config.payload_type),
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num_10ms_frames_per_packet_(
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static_cast<size_t>(config.frame_size_ms / 10)),
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full_frame_samples_(config.num_channels * config.frame_size_ms *
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sample_rate_hz / 1000),
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first_timestamp_in_buffer_(0) {
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RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
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RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
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<< "Frame size must be an integer multiple of 10 ms.";
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speech_buffer_.reserve(full_frame_samples_);
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}
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AudioEncoderPcm::~AudioEncoderPcm() = default;
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int AudioEncoderPcm::SampleRateHz() const {
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return sample_rate_hz_;
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}
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size_t AudioEncoderPcm::NumChannels() const {
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return num_channels_;
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}
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size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
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return num_10ms_frames_per_packet_;
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}
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size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
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return num_10ms_frames_per_packet_;
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}
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int AudioEncoderPcm::GetTargetBitrate() const {
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return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
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NumChannels());
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}
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AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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if (speech_buffer_.empty()) {
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first_timestamp_in_buffer_ = rtp_timestamp;
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}
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speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
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if (speech_buffer_.size() < full_frame_samples_) {
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return EncodedInfo();
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}
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RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
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EncodedInfo info;
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info.encoded_timestamp = first_timestamp_in_buffer_;
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info.payload_type = payload_type_;
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info.encoded_bytes = encoded->AppendData(
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full_frame_samples_ * BytesPerSample(),
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[&](rtc::ArrayView<uint8_t> encoded) {
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return EncodeCall(&speech_buffer_[0], full_frame_samples_,
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encoded.data());
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});
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speech_buffer_.clear();
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info.encoder_type = GetCodecType();
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return info;
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}
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void AudioEncoderPcm::Reset() {
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speech_buffer_.clear();
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}
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absl::optional<std::pair<TimeDelta, TimeDelta>>
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AudioEncoderPcm::GetFrameLengthRange() const {
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return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
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TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
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}
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size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) {
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return WebRtcG711_EncodeA(audio, input_len, encoded);
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}
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size_t AudioEncoderPcmA::BytesPerSample() const {
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return 1;
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}
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AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
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return AudioEncoder::CodecType::kPcmA;
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}
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size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
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size_t input_len,
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uint8_t* encoded) {
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return WebRtcG711_EncodeU(audio, input_len, encoded);
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}
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size_t AudioEncoderPcmU::BytesPerSample() const {
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return 1;
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}
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AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
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return AudioEncoder::CodecType::kPcmU;
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}
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} // namespace webrtc
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