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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/modules/audio_coding/codecs/g722/include/g722_interface.h"
namespace webrtc {
namespace {
const int kSampleRateHz = 16000;
} // namespace
AudioEncoderG722::EncoderState::EncoderState() {
CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
CHECK_EQ(0, WebRtcG722_EncoderInit(encoder));
}
AudioEncoderG722::EncoderState::~EncoderState() {
CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
}
AudioEncoderG722::AudioEncoderG722(const Config& config)
: num_channels_(config.num_channels),
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(config.frame_size_ms / 10),
num_10ms_frames_buffered_(0),
first_timestamp_in_buffer_(0),
encoders_(new EncoderState[num_channels_]),
interleave_buffer_(new uint8_t[2 * num_channels_]) {
CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
const int 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.reset(new uint8_t[samples_per_channel / 2]);
}
}
AudioEncoderG722::~AudioEncoderG722() {}
int AudioEncoderG722::sample_rate_hz() const {
return kSampleRateHz;
}
int AudioEncoderG722::num_channels() const {
return num_channels_;
}
int AudioEncoderG722::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderG722::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
bool AudioEncoderG722::EncodeInternal(uint32_t timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
const int samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
CHECK_GE(max_encoded_bytes,
static_cast<size_t>(samples_per_channel) / 2 * num_channels_);
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = timestamp;
// Deinterleave samples and save them in each channel's buffer.
const int start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
for (int 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_) {
info->encoded_bytes = 0;
return true;
}
// Encode each channel separately.
CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
num_10ms_frames_buffered_ = 0;
for (int i = 0; i < num_channels_; ++i) {
const int encoded = WebRtcG722_Encode(
encoders_[i].encoder, encoders_[i].speech_buffer.get(),
samples_per_channel, encoders_[i].encoded_buffer.get());
if (encoded < 0)
return false;
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 (int i = 0; i < samples_per_channel / 2; ++i) {
for (int j = 0; j < num_channels_; ++j) {
uint8_t two_samples = encoders_[j].encoded_buffer[i];
interleave_buffer_[j] = two_samples >> 4;
interleave_buffer_[num_channels_ + j] = two_samples & 0xf;
}
for (int j = 0; j < num_channels_; ++j)
encoded[i * num_channels_ + j] =
interleave_buffer_[2 * j] << 4 | interleave_buffer_[2 * j + 1];
}
info->encoded_bytes = samples_per_channel / 2 * num_channels_;
info->encoded_timestamp = first_timestamp_in_buffer_;
info->payload_type = payload_type_;
return true;
}
} // namespace webrtc
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