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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/ilbc/audio_encoder_ilbc.h"
#include <algorithm>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
namespace webrtc {
namespace {
const int kSampleRateHz = 8000;
AudioEncoderIlbc::Config CreateConfig(const CodecInst& codec_inst) {
AudioEncoderIlbc::Config config;
config.frame_size_ms = codec_inst.pacsize / 8;
config.payload_type = codec_inst.pltype;
return config;
}
} // namespace
// static
const size_t AudioEncoderIlbc::kMaxSamplesPerPacket;
bool AudioEncoderIlbc::Config::IsOk() const {
return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
frame_size_ms == 60) &&
static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
kMaxSamplesPerPacket;
}
AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
: config_(config),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
encoder_(nullptr) {
Reset();
}
AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst)
: AudioEncoderIlbc(CreateConfig(codec_inst)) {}
AudioEncoderIlbc::~AudioEncoderIlbc() {
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
}
size_t AudioEncoderIlbc::MaxEncodedBytes() const {
return RequiredOutputSizeBytes();
}
int AudioEncoderIlbc::SampleRateHz() const {
return kSampleRateHz;
}
size_t AudioEncoderIlbc::NumChannels() const {
return 1;
}
size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
return num_10ms_frames_per_packet_;
}
size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const {
return num_10ms_frames_per_packet_;
}
int AudioEncoderIlbc::GetTargetBitrate() const {
switch (num_10ms_frames_per_packet_) {
case 2: case 4:
// 38 bytes per frame of 20 ms => 15200 bits/s.
return 15200;
case 3: case 6:
// 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
return 13333;
default:
FATAL();
}
}
AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
// Save timestamp if starting a new packet.
if (num_10ms_frames_buffered_ == 0)
first_timestamp_in_buffer_ = rtp_timestamp;
// Buffer input.
RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
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;
const int output_len = WebRtcIlbcfix_Encode(
encoder_,
input_buffer_,
kSampleRateHz / 100 * num_10ms_frames_per_packet_,
encoded);
RTC_CHECK_GE(output_len, 0);
EncodedInfo info;
info.encoded_bytes = static_cast<size_t>(output_len);
RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = config_.payload_type;
return info;
}
void AudioEncoderIlbc::Reset() {
if (encoder_)
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
RTC_CHECK(config_.IsOk());
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
const int encoder_frame_size_ms = config_.frame_size_ms > 30
? config_.frame_size_ms / 2
: config_.frame_size_ms;
RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
num_10ms_frames_buffered_ = 0;
}
size_t AudioEncoderIlbc::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: FATAL();
}
}
} // namespace webrtc
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