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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/ilbc/audio_encoder_ilbc.h"
#include <algorithm>
#include <cstdint>
#include "modules/audio_coding/codecs/ilbc/ilbc.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
namespace {
const int kSampleRateHz = 8000;
int GetIlbcBitrate(int ptime) {
switch (ptime) {
case 20:
case 40:
// 38 bytes per frame of 20 ms => 15200 bits/s.
return 15200;
case 30:
case 60:
// 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
return 13333;
default:
RTC_CHECK_NOTREACHED();
}
}
} // namespace
AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
int payload_type)
: frame_size_ms_(config.frame_size_ms),
payload_type_(payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
encoder_(nullptr) {
RTC_CHECK(config.IsOk());
Reset();
}
AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
}
int AudioEncoderIlbcImpl::SampleRateHz() const {
return kSampleRateHz;
}
size_t AudioEncoderIlbcImpl::NumChannels() const {
return 1;
}
size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderIlbcImpl::GetTargetBitrate() const {
return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
10);
}
AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
// Save timestamp if starting a new packet.
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = rtp_timestamp;
// Buffer input.
std::copy(audio.cbegin(), audio.cend(),
input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
// If we don't yet have enough buffered input for a whole packet, we're done
// for now.
if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
return EncodedInfo();
}
// Encode buffered input.
RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
size_t encoded_bytes = encoded->AppendData(
RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
const int r = WebRtcIlbcfix_Encode(
encoder_, input_buffer_,
kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
RTC_CHECK_GE(r, 0);
return static_cast<size_t>(r);
});
RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
EncodedInfo info;
info.encoded_bytes = encoded_bytes;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.encoder_type = CodecType::kIlbc;
return info;
}
void AudioEncoderIlbcImpl::Reset() {
if (encoder_)
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
const int encoder_frame_size_ms =
frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
num_10ms_frames_buffered_ = 0;
}
std::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderIlbcImpl::GetFrameLengthRange() const {
return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
}
size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
switch (num_10ms_frames_per_packet_) {
case 2:
return 38;
case 3:
return 50;
case 4:
return 2 * 38;
case 6:
return 2 * 50;
default:
RTC_CHECK_NOTREACHED();
}
}
} // namespace webrtc
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