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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 "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
namespace webrtc {
namespace {
const size_t kSampleRateHz = 16000;
AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
AudioEncoderG722::Config config;
config.num_channels = codec_inst.channels;
config.frame_size_ms = codec_inst.pacsize / 16;
config.payload_type = codec_inst.pltype;
return config;
}
} // namespace
bool AudioEncoderG722::Config::IsOk() const {
return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
(num_channels >= 1);
}
AudioEncoderG722::AudioEncoderG722(const Config& config)
: num_channels_(config.num_channels),
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
num_10ms_frames_buffered_(0),
first_timestamp_in_buffer_(0),
encoders_(new EncoderState[num_channels_]),
interleave_buffer_(2 * num_channels_) {
RTC_CHECK(config.IsOk());
const size_t samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
for (int i = 0; i < num_channels_; ++i) {
encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
}
Reset();
}
AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
: AudioEncoderG722(CreateConfig(codec_inst)) {}
AudioEncoderG722::~AudioEncoderG722() = default;
size_t AudioEncoderG722::MaxEncodedBytes() const {
return SamplesPerChannel() / 2 * num_channels_;
}
int AudioEncoderG722::SampleRateHz() const {
return kSampleRateHz;
}
int AudioEncoderG722::NumChannels() const {
return num_channels_;
}
int AudioEncoderG722::RtpTimestampRateHz() const {
// The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
// codec.
return kSampleRateHz / 2;
}
size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderG722::GetTargetBitrate() const {
// 4 bits/sample, 16000 samples/s/channel.
return 64000 * NumChannels();
}
AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = rtp_timestamp;
// Deinterleave samples and save them in each channel's buffer.
const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
for (size_t i = 0; i < kSampleRateHz / 100; ++i)
for (int j = 0; j < num_channels_; ++j)
encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
// If we don't yet have enough samples for a packet, we're done for now.
if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
return EncodedInfo();
}
// Encode each channel separately.
RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
const size_t samples_per_channel = SamplesPerChannel();
for (int i = 0; i < num_channels_; ++i) {
const size_t encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.data());
RTC_CHECK_EQ(encoded, samples_per_channel / 2);
}
// Interleave the encoded bytes of the different channels. Each separate
// channel and the interleaved stream encodes two samples per byte, most
// significant half first.
for (size_t i = 0; i < samples_per_channel / 2; ++i) {
for (int j = 0; j < num_channels_; ++j) {
uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
interleave_buffer_.data()[j] = two_samples >> 4;
interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
}
for (int j = 0; j < num_channels_; ++j)
encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
interleave_buffer_.data()[2 * j + 1];
}
EncodedInfo info;
info.encoded_bytes = samples_per_channel / 2 * num_channels_;
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
return info;
}
void AudioEncoderG722::Reset() {
num_10ms_frames_buffered_ = 0;
for (int i = 0; i < num_channels_; ++i)
RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
}
AudioEncoderG722::EncoderState::EncoderState() {
RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
}
AudioEncoderG722::EncoderState::~EncoderState() {
RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
}
size_t AudioEncoderG722::SamplesPerChannel() const {
return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
}
} // namespace webrtc
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