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d83a3d71bc
session-android/jni/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
Moxie Marlinspike d83a3d71bc Support for Signal calls. 
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

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/*
* Copyright (c) 2012 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/main/acm2/acm_isac.h"
#include <assert.h>
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#endif
#if defined (WEBRTC_CODEC_ISAC) || defined (WEBRTC_CODEC_ISACFX)
#include "webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h"
#endif
namespace webrtc {
namespace acm2 {
// we need this otherwise we cannot use forward declaration
// in the header file
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
struct ACMISACInst {
ACM_ISAC_STRUCT* inst;
};
#endif
#define ISAC_MIN_RATE 10000
#define ISAC_MAX_RATE 56000
// Tables for bandwidth estimates
#define NR_ISAC_BANDWIDTHS 24
static const int32_t kIsacRatesWb[NR_ISAC_BANDWIDTHS] = {
10000, 11100, 12300, 13700, 15200, 16900, 18800, 20900, 23300, 25900, 28700,
31900, 10100, 11200, 12400, 13800, 15300, 17000, 18900, 21000, 23400, 26000,
28800, 32000};
static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = {
10000, 11000, 12400, 13800, 15300, 17000, 18900, 21000, 23200, 25400, 27600,
29800, 32000, 34100, 36300, 38500, 40700, 42900, 45100, 47300, 49500, 51700,
53900, 56000 };
#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))
ACMISAC::ACMISAC(int16_t /* codec_id */)
: codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
codec_inst_ptr_(NULL),
is_enc_initialized_(false),
isac_coding_mode_(CHANNEL_INDEPENDENT),
enforce_frame_size_(false),
isac_currentBN_(32000),
samples_in10MsAudio_(160), // Initiates to 16 kHz mode.
decoder_initialized_(false) {
}
ACMISAC::~ACMISAC() {
return;
}
ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; }
int16_t ACMISAC::InternalEncode(uint8_t* /* bitstream */,
int16_t* /* bitstream_len_byte */) {
return -1;
}
int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) {
return -1;
}
int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* /* codec_params */) {
return -1;
}
int16_t ACMISAC::InternalCreateEncoder() { return -1; }
void ACMISAC::DestructEncoderSafe() { return; }
void ACMISAC::InternalDestructEncoderInst(void* /* ptr_inst */) { return; }
int16_t ACMISAC::Transcode(uint8_t* /* bitstream */,
int16_t* /* bitstream_len_byte */,
int16_t /* q_bwe */,
int32_t /* scale */,
bool /* is_red */) {
return -1;
}
int16_t ACMISAC::SetBitRateSafe(int32_t /* bit_rate */) { return -1; }
int32_t ACMISAC::GetEstimatedBandwidthSafe() { return -1; }
int32_t ACMISAC::SetEstimatedBandwidthSafe(int32_t /* estimated_bandwidth */) {
return -1;
}
int32_t ACMISAC::GetRedPayloadSafe(uint8_t* /* red_payload */,
int16_t* /* payload_bytes */) {
return -1;
}
int16_t ACMISAC::UpdateDecoderSampFreq(int16_t /* codec_id */) { return -1; }
int16_t ACMISAC::UpdateEncoderSampFreq(uint16_t /* encoder_samp_freq_hz */) {
return -1;
}
int16_t ACMISAC::EncoderSampFreq(uint16_t* /* samp_freq_hz */) { return -1; }
int32_t ACMISAC::ConfigISACBandwidthEstimator(
const uint8_t /* init_frame_size_msec */,
const uint16_t /* init_rate_bit_per_sec */,
const bool /* enforce_frame_size */) {
return -1;
}
int32_t ACMISAC::SetISACMaxPayloadSize(
const uint16_t /* max_payload_len_bytes */) {
return -1;
}
int32_t ACMISAC::SetISACMaxRate(const uint32_t /* max_rate_bit_per_sec */) {
return -1;
}
void ACMISAC::UpdateFrameLen() { return; }
void ACMISAC::CurrentRate(int32_t* /*rate_bit_per_sec */) { return; }
bool ACMISAC::DecoderParamsSafe(WebRtcACMCodecParams* /* dec_params */,
const uint8_t /* payload_type */) {
return false;
}
int16_t ACMISAC::REDPayloadISAC(const int32_t /* isac_rate */,
const int16_t /* isac_bw_estimate */,
uint8_t* /* payload */,
int16_t* /* payload_len_bytes */) {
return -1;
}
AudioDecoder* ACMISAC::Decoder(int /* codec_id */) { return NULL; }
#else //===================== Actual Implementation =======================
#ifdef WEBRTC_CODEC_ISACFX
// How the scaling is computed. iSAC computes a gain based on the
// bottleneck. It follows the following expression for that
//
// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0)
// / 3.4641;
//
// Where for 30 ms framelength we have,
//
// a = -23; b = 0.48; c = 0;
//
// As the default encoder is operating at 32kbps we have the scale as
//
// S(BN_kbps) = G(BN_kbps) / G(32);
#define ISAC_NUM_SUPPORTED_RATES 9
static const uint16_t kIsacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
32000, 30000, 26000, 23000, 21000, 19000, 17000, 15000, 12000};
static const float kIsacScale[ISAC_NUM_SUPPORTED_RATES] = {
1.0f, 0.8954f, 0.7178f, 0.6081f, 0.5445f,
0.4875f, 0.4365f, 0.3908f, 0.3311f
};
enum IsacSamplingRate {
kIsacWideband = 16,
kIsacSuperWideband = 32
};
static float ACMISACFixTranscodingScale(uint16_t rate) {
// find the scale for transcoding, the scale is rounded
// downward
float scale = -1;
for (int16_t n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) {
if (rate >= kIsacSuportedRates[n]) {
scale = kIsacScale[n];
break;
}
}
return scale;
}
static void ACMISACFixGetSendBitrate(ACM_ISAC_STRUCT* inst,
int32_t* bottleneck) {
*bottleneck = WebRtcIsacfix_GetUplinkBw(inst);
}
static int16_t ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
int16_t bwe_index,
int16_t /* jitter_index */,
int32_t rate,
int16_t* bitstream,
bool is_red) {
if (is_red) {
// RED not supported with iSACFIX
return -1;
}
float scale = ACMISACFixTranscodingScale((uint16_t)rate);
return WebRtcIsacfix_GetNewBitStream(inst, bwe_index, scale, bitstream);
}
static int16_t ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst,
int16_t* rate_index,
int16_t* /* dummy */) {
int16_t local_rate_index;
int16_t status = WebRtcIsacfix_GetDownLinkBwIndex(inst, &local_rate_index);
if (status < 0) {
return -1;
} else {
*rate_index = local_rate_index;
return 0;
}
}
static int16_t ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst,
int32_t rate_bps,
int16_t frame_size_ms,
int16_t enforce_frame_size) {
return WebRtcIsacfix_ControlBwe(
inst, (int16_t)rate_bps, frame_size_ms, enforce_frame_size);
}
static int16_t ACMISACFixControl(ACM_ISAC_STRUCT* inst,
int32_t rate_bps,
int16_t frame_size_ms) {
return WebRtcIsacfix_Control(inst, (int16_t)rate_bps, frame_size_ms);
}
// The following two function should have the same signature as their counter
// part in iSAC floating-point, i.e. WebRtcIsac_EncSampRate &
// WebRtcIsac_DecSampRate.
static uint16_t ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) {
return 16000;
}
static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) {
return 16000;
}
#endif
ACMISAC::ACMISAC(int16_t codec_id)
: AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]),
codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
is_enc_initialized_(false),
isac_coding_mode_(CHANNEL_INDEPENDENT),
enforce_frame_size_(false),
isac_current_bn_(32000),
samples_in_10ms_audio_(160), // Initiates to 16 kHz mode.
decoder_initialized_(false) {
codec_id_ = codec_id;
// Create codec instance.
codec_inst_ptr_ = new ACMISACInst;
if (codec_inst_ptr_ == NULL) {
return;
}
codec_inst_ptr_->inst = NULL;
state_ = codec_inst_ptr_;
}
ACMISAC::~ACMISAC() {
if (codec_inst_ptr_ != NULL) {
if (codec_inst_ptr_->inst != NULL) {
ACM_ISAC_FREE(codec_inst_ptr_->inst);
codec_inst_ptr_->inst = NULL;
}
delete codec_inst_ptr_;
codec_inst_ptr_ = NULL;
}
return;
}
int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
// set decoder sampling frequency.
if (codec_params->codec_inst.plfreq == 32000 ||
codec_params->codec_inst.plfreq == 48000) {
UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
} else {
UpdateDecoderSampFreq(ACMCodecDB::kISAC);
}
// in a one-way communication we may never register send-codec.
// However we like that the BWE to work properly so it has to
// be initialized. The BWE is initialized when iSAC encoder is initialized.
// Therefore, we need this.
if (!encoder_initialized_) {
// Since we don't require a valid rate or a valid packet size when
// initializing the decoder, we set valid values before initializing encoder
codec_params->codec_inst.rate = kIsacWbDefaultRate;
codec_params->codec_inst.pacsize = kIsacPacSize960;
if (InternalInitEncoder(codec_params) < 0) {
return -1;
}
encoder_initialized_ = true;
}
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst);
}
ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; }
int16_t ACMISAC::InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) {
// ISAC takes 10ms audio every time we call encoder, therefore,
// it should be treated like codecs with 'basic coding block'
// non-zero, and the following 'while-loop' should not be necessary.
// However, due to a mistake in the codec the frame-size might change
// at the first 10ms pushed in to iSAC if the bit-rate is low, this is
// sort of a bug in iSAC. to address this we treat iSAC as the
// following.
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (codec_inst_ptr_ == NULL) {
return -1;
}
*bitstream_len_byte = 0;
while ((*bitstream_len_byte == 0) && (in_audio_ix_read_ < frame_len_smpl_)) {
if (in_audio_ix_read_ > in_audio_ix_write_) {
// something is wrong.
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
"The actual frame-size of iSAC appears to be larger that "
"expected. All audio pushed in but no bit-stream is "
"generated.");
return -1;
}
*bitstream_len_byte = ACM_ISAC_ENCODE(
codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_],
reinterpret_cast<int16_t*>(bitstream));
// increment the read index this tell the caller that how far
// we have gone forward in reading the audio buffer
in_audio_ix_read_ += samples_in_10ms_audio_;
}
if (*bitstream_len_byte == 0) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_,
"ISAC Has encoded the whole frame but no bit-stream is "
"generated.");
}
// a packet is generated iSAC, is set in adaptive mode may change
// the frame length and we like to update the bottleneck value as
// well, although updating bottleneck is not crucial
if ((*bitstream_len_byte > 0) && (isac_coding_mode_ == ADAPTIVE)) {
ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
}
UpdateFrameLen();
return *bitstream_len_byte;
}
int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
// if rate is set to -1 then iSAC has to be in adaptive mode
if (codec_params->codec_inst.rate == -1) {
isac_coding_mode_ = ADAPTIVE;
} else if ((codec_params->codec_inst.rate >= ISAC_MIN_RATE) &&
(codec_params->codec_inst.rate <= ISAC_MAX_RATE)) {
// sanity check that rate is in acceptable range
isac_coding_mode_ = CHANNEL_INDEPENDENT;
isac_current_bn_ = codec_params->codec_inst.rate;
} else {
return -1;
}
// we need to set the encoder sampling frequency.
if (UpdateEncoderSampFreq((uint16_t)codec_params->codec_inst.plfreq) < 0) {
return -1;
}
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) {
return -1;
}
// apply the frame-size and rate if operating in
// channel-independent mode
if (isac_coding_mode_ == CHANNEL_INDEPENDENT) {
if (ACM_ISAC_CONTROL(codec_inst_ptr_->inst,
codec_params->codec_inst.rate,
codec_params->codec_inst.pacsize /
(codec_params->codec_inst.plfreq / 1000)) < 0) {
return -1;
}
} else {
// We need this for adaptive case and has to be called
// after initialization
ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
}
frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst);
return 0;
}
int16_t ACMISAC::InternalCreateEncoder() {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (codec_inst_ptr_ == NULL) {
return -1;
}
decoder_initialized_ = false;
int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
if (status < 0)
codec_inst_ptr_->inst = NULL;
return status;
}
int16_t ACMISAC::Transcode(uint8_t* bitstream,
int16_t* bitstream_len_byte,
int16_t q_bwe,
int32_t rate,
bool is_red) {
int16_t jitter_info = 0;
// transcode from a higher rate to lower rate sanity check
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (codec_inst_ptr_ == NULL) {
return -1;
}
*bitstream_len_byte = ACM_ISAC_GETNEWBITSTREAM(
codec_inst_ptr_->inst, q_bwe, jitter_info, rate,
reinterpret_cast<int16_t*>(bitstream), (is_red) ? 1 : 0);
if (*bitstream_len_byte < 0) {
// error happened
*bitstream_len_byte = 0;
return -1;
} else {
return *bitstream_len_byte;
}
}
void ACMISAC::UpdateFrameLen() {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst);
encoder_params_.codec_inst.pacsize = frame_len_smpl_;
}
void ACMISAC::DestructEncoderSafe() {
// codec with shared instance cannot delete.
encoder_initialized_ = false;
return;
}
void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) {
if (ptr_inst != NULL) {
ACM_ISAC_FREE(static_cast<ACM_ISAC_STRUCT *>(ptr_inst));
}
return;
}
int16_t ACMISAC::SetBitRateSafe(int32_t bit_rate) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (codec_inst_ptr_ == NULL) {
return -1;
}
uint16_t encoder_samp_freq;
EncoderSampFreq(&encoder_samp_freq);
bool reinit = false;
// change the BN of iSAC
if (bit_rate == -1) {
// ADAPTIVE MODE
// Check if it was already in adaptive mode
if (isac_coding_mode_ != ADAPTIVE) {
// was not in adaptive, then set the mode to adaptive
// and flag for re-initialization
isac_coding_mode_ = ADAPTIVE;
reinit = true;
}
} else if ((bit_rate >= ISAC_MIN_RATE) && (bit_rate <= ISAC_MAX_RATE)) {
// Sanity check if the rate valid
// check if it was in channel-independent mode before
if (isac_coding_mode_ != CHANNEL_INDEPENDENT) {
// was not in channel independent, set the mode to
// channel-independent and flag for re-initialization
isac_coding_mode_ = CHANNEL_INDEPENDENT;
reinit = true;
}
// store the bottleneck
isac_current_bn_ = (uint16_t)bit_rate;
} else {
// invlaid rate
return -1;
}
int16_t status = 0;
if (reinit) {
// initialize and check if it is successful
if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) {
// failed initialization
return -1;
}
}
if (isac_coding_mode_ == CHANNEL_INDEPENDENT) {
status = ACM_ISAC_CONTROL(
codec_inst_ptr_->inst, isac_current_bn_,
(encoder_samp_freq == 32000 || encoder_samp_freq == 48000) ? 30 :
(frame_len_smpl_ / 16));
if (status < 0) {
status = -1;
}
}
// Update encoder parameters
encoder_params_.codec_inst.rate = bit_rate;
UpdateFrameLen();
return status;
}
int32_t ACMISAC::GetEstimatedBandwidthSafe() {
int16_t bandwidth_index = 0;
int16_t delay_index = 0;
int samp_rate;
// Get bandwidth information
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
ACM_ISAC_GETSENDBWE(codec_inst_ptr_->inst, &bandwidth_index, &delay_index);
// Validy check of index
if ((bandwidth_index < 0) || (bandwidth_index >= NR_ISAC_BANDWIDTHS)) {
return -1;
}
// Check sample frequency
samp_rate = ACM_ISAC_GETDECSAMPRATE(codec_inst_ptr_->inst);
if (samp_rate == 16000) {
return kIsacRatesWb[bandwidth_index];
} else {
return kIsacRatesSwb[bandwidth_index];
}
}
int32_t ACMISAC::SetEstimatedBandwidthSafe(int32_t estimated_bandwidth) {
int samp_rate;
int16_t bandwidth_index;
// Check sample frequency and choose appropriate table
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
samp_rate = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst);
if (samp_rate == 16000) {
// Search through the WB rate table to find the index
bandwidth_index = NR_ISAC_BANDWIDTHS / 2 - 1;
for (int i = 0; i < (NR_ISAC_BANDWIDTHS / 2); i++) {
if (estimated_bandwidth == kIsacRatesWb[i]) {
bandwidth_index = i;
break;
} else if (estimated_bandwidth
== kIsacRatesWb[i + NR_ISAC_BANDWIDTHS / 2]) {
bandwidth_index = i + NR_ISAC_BANDWIDTHS / 2;
break;
} else if (estimated_bandwidth < kIsacRatesWb[i]) {
bandwidth_index = i;
break;
}
}
} else {
// Search through the SWB rate table to find the index
bandwidth_index = NR_ISAC_BANDWIDTHS - 1;
for (int i = 0; i < NR_ISAC_BANDWIDTHS; i++) {
if (estimated_bandwidth <= kIsacRatesSwb[i]) {
bandwidth_index = i;
break;
}
}
}
// Set iSAC Bandwidth Estimate
ACM_ISAC_SETBWE(codec_inst_ptr_->inst, bandwidth_index);
return 0;
}
int32_t ACMISAC::GetRedPayloadSafe(
#if (!defined(WEBRTC_CODEC_ISAC))
uint8_t* /* red_payload */,
int16_t* /* payload_bytes */) {
return -1;
#else
uint8_t* red_payload, int16_t* payload_bytes) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
int16_t bytes =
WebRtcIsac_GetRedPayload(
codec_inst_ptr_->inst, reinterpret_cast<int16_t*>(red_payload));
if (bytes < 0) {
return -1;
}
*payload_bytes = bytes;
return 0;
#endif
}
int16_t ACMISAC::UpdateDecoderSampFreq(
#ifdef WEBRTC_CODEC_ISAC
int16_t codec_id) {
// The decoder supports only wideband and super-wideband.
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (ACMCodecDB::kISAC == codec_id) {
return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 16000);
} else if (ACMCodecDB::kISACSWB == codec_id ||
ACMCodecDB::kISACFB == codec_id) {
return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 32000);
} else {
return -1;
}
#else
int16_t /* codec_id */) {
return 0;
#endif
}
int16_t ACMISAC::UpdateEncoderSampFreq(
#ifdef WEBRTC_CODEC_ISAC
uint16_t encoder_samp_freq_hz) {
uint16_t current_samp_rate_hz;
EncoderSampFreq(&current_samp_rate_hz);
if (current_samp_rate_hz != encoder_samp_freq_hz) {
if ((encoder_samp_freq_hz != 16000) && (encoder_samp_freq_hz != 32000) &&
(encoder_samp_freq_hz != 48000)) {
return -1;
} else {
in_audio_ix_read_ = 0;
in_audio_ix_write_ = 0;
in_timestamp_ix_write_ = 0;
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
if (WebRtcIsac_SetEncSampRate(codec_inst_ptr_->inst,
encoder_samp_freq_hz) < 0) {
return -1;
}
samples_in_10ms_audio_ = encoder_samp_freq_hz / 100;
frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst);
encoder_params_.codec_inst.pacsize = frame_len_smpl_;
encoder_params_.codec_inst.plfreq = encoder_samp_freq_hz;
return 0;
}
}
#else
uint16_t /* codec_id */) {
#endif
return 0;
}
int16_t ACMISAC::EncoderSampFreq(uint16_t* samp_freq_hz) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
*samp_freq_hz = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst);
return 0;
}
int32_t ACMISAC::ConfigISACBandwidthEstimator(
const uint8_t init_frame_size_msec,
const uint16_t init_rate_bit_per_sec,
const bool enforce_frame_size) {
int16_t status;
{
uint16_t samp_freq_hz;
EncoderSampFreq(&samp_freq_hz);
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
// TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce
// the frame-size otherwise we might get error. Revise if
// control-bwe is changed.
if (samp_freq_hz == 32000 || samp_freq_hz == 48000) {
status = ACM_ISAC_CONTROL_BWE(codec_inst_ptr_->inst,
init_rate_bit_per_sec, 30, 1);
} else {
status = ACM_ISAC_CONTROL_BWE(codec_inst_ptr_->inst,
init_rate_bit_per_sec,
init_frame_size_msec,
enforce_frame_size ? 1 : 0);
}
}
if (status < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
"Couldn't config iSAC BWE.");
return -1;
}
{
WriteLockScoped wl(codec_wrapper_lock_);
UpdateFrameLen();
}
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
return 0;
}
int32_t ACMISAC::SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_SETMAXPAYLOADSIZE(codec_inst_ptr_->inst,
max_payload_len_bytes);
}
int32_t ACMISAC::SetISACMaxRate(const uint32_t max_rate_bit_per_sec) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_SETMAXRATE(codec_inst_ptr_->inst, max_rate_bit_per_sec);
}
void ACMISAC::CurrentRate(int32_t* rate_bit_per_sec) {
if (isac_coding_mode_ == ADAPTIVE) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, rate_bit_per_sec);
}
}
int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate,
const int16_t isac_bw_estimate,
uint8_t* payload,
int16_t* payload_len_bytes) {
int16_t status;
ReadLockScoped rl(codec_wrapper_lock_);
status =
Transcode(payload, payload_len_bytes, isac_bw_estimate, isac_rate, true);
return status;
}
int ACMISAC::Decode(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
SpeechType* speech_type) {
int16_t temp_type;
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
int ret =
ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
reinterpret_cast<const uint16_t*>(encoded),
static_cast<int16_t>(encoded_len),
decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int ACMISAC::DecodePlc(int num_frames, int16_t* decoded) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_DECODEPLC(
static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
decoded,
static_cast<int16_t>(num_frames));
}
int ACMISAC::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_DECODE_BWE(
static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
reinterpret_cast<const uint16_t*>(payload),
static_cast<uint32_t>(payload_len),
rtp_sequence_number,
rtp_timestamp,
arrival_timestamp);
}
int ACMISAC::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
int16_t ret =
ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
reinterpret_cast<const uint16_t*>(encoded),
static_cast<int16_t>(encoded_len),
decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
int ACMISAC::ErrorCode() {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
return ACM_ISAC_GETERRORCODE(
static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst));
}
AudioDecoder* ACMISAC::Decoder(int codec_id) {
// Create iSAC instance if it does not exist.
WriteLockScoped wl(codec_wrapper_lock_);
if (!encoder_exist_) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
assert(codec_inst_ptr_->inst == NULL);
encoder_initialized_ = false;
decoder_initialized_ = false;
if (ACM_ISAC_CREATE(&(codec_inst_ptr_->inst)) < 0) {
codec_inst_ptr_->inst = NULL;
return NULL;
}
encoder_exist_ = true;
}
WebRtcACMCodecParams codec_params;
if (!encoder_initialized_ || !decoder_initialized_) {
ACMCodecDB::Codec(codec_id, &codec_params.codec_inst);
// The following three values are not used but we set them to valid values.
codec_params.enable_dtx = false;
codec_params.enable_vad = false;
codec_params.vad_mode = VADNormal;
}
if (!encoder_initialized_) {
// Initialize encoder to make sure bandwidth estimator works.
if (InternalInitEncoder(&codec_params) < 0)
return NULL;
encoder_initialized_ = true;
}
if (!decoder_initialized_) {
if (InternalInitDecoder(&codec_params) < 0)
return NULL;
decoder_initialized_ = true;
}
return this;
}
#endif
} // namespace acm2
} // namespace webrtc
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