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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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/base/checks.h"
namespace webrtc {
template <typename T>
typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
const CodecInst& codec_inst,
LockedIsacBandwidthInfo* bwinfo) {
typename AudioEncoderIsacT<T>::Config config;
config.bwinfo = bwinfo;
config.payload_type = codec_inst.pltype;
config.sample_rate_hz = codec_inst.plfreq;
config.frame_size_ms =
rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
config.adaptive_mode = (codec_inst.rate == -1);
if (codec_inst.rate != -1)
config.bit_rate = codec_inst.rate;
return config;
}
template <typename T>
bool AudioEncoderIsacT<T>::Config::IsOk() const {
if (max_bit_rate < 32000 && max_bit_rate != -1)
return false;
if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
return false;
if (adaptive_mode && !bwinfo)
return false;
switch (sample_rate_hz) {
case 16000:
if (max_bit_rate > 53400)
return false;
if (max_payload_size_bytes > 400)
return false;
return (frame_size_ms == 30 || frame_size_ms == 60) &&
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000));
case 32000:
if (max_bit_rate > 160000)
return false;
if (max_payload_size_bytes > 600)
return false;
return T::has_swb &&
(frame_size_ms == 30 &&
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000)));
default:
return false;
}
}
template <typename T>
AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
RecreateEncoderInstance(config);
}
template <typename T>
AudioEncoderIsacT<T>::AudioEncoderIsacT(const CodecInst& codec_inst,
LockedIsacBandwidthInfo* bwinfo)
: AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
template <typename T>
AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
RTC_CHECK_EQ(0, T::Free(isac_state_));
}
template <typename T>
size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
return kSufficientEncodeBufferSizeBytes;
}
template <typename T>
int AudioEncoderIsacT<T>::SampleRateHz() const {
return T::EncSampRate(isac_state_);
}
template <typename T>
size_t AudioEncoderIsacT<T>::NumChannels() const {
return 1;
}
template <typename T>
size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
return static_cast<size_t>(
rtc::CheckedDivExact(samples_in_next_packet,
rtc::CheckedDivExact(SampleRateHz(), 100)));
}
template <typename T>
size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
return 6; // iSAC puts at most 60 ms in a packet.
}
template <typename T>
int AudioEncoderIsacT<T>::GetTargetBitrate() const {
if (config_.adaptive_mode)
return -1;
return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
}
template <typename T>
AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
if (!packet_in_progress_) {
// Starting a new packet; remember the timestamp for later.
packet_in_progress_ = true;
packet_timestamp_ = rtp_timestamp;
}
if (bwinfo_) {
IsacBandwidthInfo bwinfo = bwinfo_->Get();
T::SetBandwidthInfo(isac_state_, &bwinfo);
}
int r = T::Encode(isac_state_, audio.data(), encoded);
RTC_CHECK_GE(r, 0) << "Encode failed (error code "
<< T::GetErrorCode(isac_state_) << ")";
// T::Encode doesn't allow us to tell it the size of the output
// buffer. All we can do is check for an overrun after the fact.
RTC_CHECK_LE(static_cast<size_t>(r), max_encoded_bytes);
if (r == 0)
return EncodedInfo();
// Got enough input to produce a packet. Return the saved timestamp from
// the first chunk of input that went into the packet.
packet_in_progress_ = false;
EncodedInfo info;
info.encoded_bytes = r;
info.encoded_timestamp = packet_timestamp_;
info.payload_type = config_.payload_type;
return info;
}
template <typename T>
void AudioEncoderIsacT<T>::Reset() {
RecreateEncoderInstance(config_);
}
template <typename T>
void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
RTC_CHECK(config.IsOk());
packet_in_progress_ = false;
bwinfo_ = config.bwinfo;
if (isac_state_)
RTC_CHECK_EQ(0, T::Free(isac_state_));
RTC_CHECK_EQ(0, T::Create(&isac_state_));
RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
if (config.adaptive_mode) {
RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
config.enforce_frame_size));
} else {
RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
}
if (config.max_payload_size_bytes != -1)
RTC_CHECK_EQ(
0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
if (config.max_bit_rate != -1)
RTC_CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
// Set the decoder sample rate even though we just use the encoder. This
// doesn't appear to be necessary to produce a valid encoding, but without it
// we get an encoding that isn't bit-for-bit identical with what a combined
// encoder+decoder object produces.
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
config_ = config;
}
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
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