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841 lines
27 KiB
C++
841 lines
27 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/main/acm2/acm_isac.h"
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#include <assert.h>
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#ifdef WEBRTC_CODEC_ISAC
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#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
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#endif
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#ifdef WEBRTC_CODEC_ISACFX
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
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#endif
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#if defined (WEBRTC_CODEC_ISAC) || defined (WEBRTC_CODEC_ISACFX)
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#include "webrtc/modules/audio_coding/main/acm2/acm_isac_macros.h"
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#endif
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namespace webrtc {
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namespace acm2 {
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// we need this otherwise we cannot use forward declaration
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// in the header file
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
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struct ACMISACInst {
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ACM_ISAC_STRUCT* inst;
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};
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#endif
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#define ISAC_MIN_RATE 10000
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#define ISAC_MAX_RATE 56000
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// Tables for bandwidth estimates
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#define NR_ISAC_BANDWIDTHS 24
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static const int32_t kIsacRatesWb[NR_ISAC_BANDWIDTHS] = {
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10000, 11100, 12300, 13700, 15200, 16900, 18800, 20900, 23300, 25900, 28700,
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31900, 10100, 11200, 12400, 13800, 15300, 17000, 18900, 21000, 23400, 26000,
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28800, 32000};
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static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = {
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10000, 11000, 12400, 13800, 15300, 17000, 18900, 21000, 23200, 25400, 27600,
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29800, 32000, 34100, 36300, 38500, 40700, 42900, 45100, 47300, 49500, 51700,
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53900, 56000 };
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#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))
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ACMISAC::ACMISAC(int16_t /* codec_id */)
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: codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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codec_inst_ptr_(NULL),
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is_enc_initialized_(false),
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isac_coding_mode_(CHANNEL_INDEPENDENT),
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enforce_frame_size_(false),
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isac_currentBN_(32000),
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samples_in10MsAudio_(160), // Initiates to 16 kHz mode.
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decoder_initialized_(false) {
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}
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ACMISAC::~ACMISAC() {
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return;
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}
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ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; }
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int16_t ACMISAC::InternalEncode(uint8_t* /* bitstream */,
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int16_t* /* bitstream_len_byte */) {
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return -1;
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}
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int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) {
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return -1;
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}
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int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* /* codec_params */) {
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return -1;
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}
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int16_t ACMISAC::InternalCreateEncoder() { return -1; }
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void ACMISAC::DestructEncoderSafe() { return; }
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void ACMISAC::InternalDestructEncoderInst(void* /* ptr_inst */) { return; }
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int16_t ACMISAC::Transcode(uint8_t* /* bitstream */,
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int16_t* /* bitstream_len_byte */,
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int16_t /* q_bwe */,
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int32_t /* scale */,
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bool /* is_red */) {
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return -1;
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}
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int16_t ACMISAC::SetBitRateSafe(int32_t /* bit_rate */) { return -1; }
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int32_t ACMISAC::GetEstimatedBandwidthSafe() { return -1; }
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int32_t ACMISAC::SetEstimatedBandwidthSafe(int32_t /* estimated_bandwidth */) {
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return -1;
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}
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int32_t ACMISAC::GetRedPayloadSafe(uint8_t* /* red_payload */,
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int16_t* /* payload_bytes */) {
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return -1;
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}
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int16_t ACMISAC::UpdateDecoderSampFreq(int16_t /* codec_id */) { return -1; }
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int16_t ACMISAC::UpdateEncoderSampFreq(uint16_t /* encoder_samp_freq_hz */) {
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return -1;
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}
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int16_t ACMISAC::EncoderSampFreq(uint16_t* /* samp_freq_hz */) { return -1; }
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int32_t ACMISAC::ConfigISACBandwidthEstimator(
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const uint8_t /* init_frame_size_msec */,
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const uint16_t /* init_rate_bit_per_sec */,
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const bool /* enforce_frame_size */) {
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return -1;
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}
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int32_t ACMISAC::SetISACMaxPayloadSize(
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const uint16_t /* max_payload_len_bytes */) {
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return -1;
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}
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int32_t ACMISAC::SetISACMaxRate(const uint32_t /* max_rate_bit_per_sec */) {
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return -1;
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}
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void ACMISAC::UpdateFrameLen() { return; }
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void ACMISAC::CurrentRate(int32_t* /*rate_bit_per_sec */) { return; }
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bool ACMISAC::DecoderParamsSafe(WebRtcACMCodecParams* /* dec_params */,
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const uint8_t /* payload_type */) {
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return false;
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}
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int16_t ACMISAC::REDPayloadISAC(const int32_t /* isac_rate */,
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const int16_t /* isac_bw_estimate */,
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uint8_t* /* payload */,
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int16_t* /* payload_len_bytes */) {
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return -1;
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}
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AudioDecoder* ACMISAC::Decoder(int /* codec_id */) { return NULL; }
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#else //===================== Actual Implementation =======================
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#ifdef WEBRTC_CODEC_ISACFX
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// How the scaling is computed. iSAC computes a gain based on the
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// bottleneck. It follows the following expression for that
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//
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// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0)
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// / 3.4641;
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//
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// Where for 30 ms framelength we have,
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//
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// a = -23; b = 0.48; c = 0;
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//
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// As the default encoder is operating at 32kbps we have the scale as
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//
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// S(BN_kbps) = G(BN_kbps) / G(32);
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#define ISAC_NUM_SUPPORTED_RATES 9
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static const uint16_t kIsacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
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32000, 30000, 26000, 23000, 21000, 19000, 17000, 15000, 12000};
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static const float kIsacScale[ISAC_NUM_SUPPORTED_RATES] = {
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1.0f, 0.8954f, 0.7178f, 0.6081f, 0.5445f,
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0.4875f, 0.4365f, 0.3908f, 0.3311f
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};
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enum IsacSamplingRate {
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kIsacWideband = 16,
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kIsacSuperWideband = 32
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};
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static float ACMISACFixTranscodingScale(uint16_t rate) {
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// find the scale for transcoding, the scale is rounded
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// downward
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float scale = -1;
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for (int16_t n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) {
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if (rate >= kIsacSuportedRates[n]) {
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scale = kIsacScale[n];
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break;
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}
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}
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return scale;
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}
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static void ACMISACFixGetSendBitrate(ACM_ISAC_STRUCT* inst,
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int32_t* bottleneck) {
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*bottleneck = WebRtcIsacfix_GetUplinkBw(inst);
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}
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static int16_t ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
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int16_t bwe_index,
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int16_t /* jitter_index */,
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int32_t rate,
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int16_t* bitstream,
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bool is_red) {
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if (is_red) {
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// RED not supported with iSACFIX
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return -1;
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}
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float scale = ACMISACFixTranscodingScale((uint16_t)rate);
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return WebRtcIsacfix_GetNewBitStream(inst, bwe_index, scale, bitstream);
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}
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static int16_t ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst,
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int16_t* rate_index,
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int16_t* /* dummy */) {
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int16_t local_rate_index;
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int16_t status = WebRtcIsacfix_GetDownLinkBwIndex(inst, &local_rate_index);
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if (status < 0) {
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return -1;
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} else {
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*rate_index = local_rate_index;
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return 0;
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}
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}
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static int16_t ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst,
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int32_t rate_bps,
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int16_t frame_size_ms,
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int16_t enforce_frame_size) {
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return WebRtcIsacfix_ControlBwe(
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inst, (int16_t)rate_bps, frame_size_ms, enforce_frame_size);
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}
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static int16_t ACMISACFixControl(ACM_ISAC_STRUCT* inst,
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int32_t rate_bps,
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int16_t frame_size_ms) {
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return WebRtcIsacfix_Control(inst, (int16_t)rate_bps, frame_size_ms);
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}
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// The following two function should have the same signature as their counter
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// part in iSAC floating-point, i.e. WebRtcIsac_EncSampRate &
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// WebRtcIsac_DecSampRate.
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static uint16_t ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) {
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return 16000;
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}
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static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) {
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return 16000;
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}
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#endif
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ACMISAC::ACMISAC(int16_t codec_id)
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: AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]),
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codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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is_enc_initialized_(false),
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isac_coding_mode_(CHANNEL_INDEPENDENT),
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enforce_frame_size_(false),
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isac_current_bn_(32000),
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samples_in_10ms_audio_(160), // Initiates to 16 kHz mode.
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decoder_initialized_(false) {
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codec_id_ = codec_id;
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// Create codec instance.
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codec_inst_ptr_ = new ACMISACInst;
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if (codec_inst_ptr_ == NULL) {
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return;
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}
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codec_inst_ptr_->inst = NULL;
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state_ = codec_inst_ptr_;
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}
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ACMISAC::~ACMISAC() {
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if (codec_inst_ptr_ != NULL) {
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if (codec_inst_ptr_->inst != NULL) {
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ACM_ISAC_FREE(codec_inst_ptr_->inst);
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codec_inst_ptr_->inst = NULL;
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}
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delete codec_inst_ptr_;
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codec_inst_ptr_ = NULL;
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}
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return;
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}
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int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) {
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// set decoder sampling frequency.
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if (codec_params->codec_inst.plfreq == 32000 ||
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codec_params->codec_inst.plfreq == 48000) {
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UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
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} else {
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UpdateDecoderSampFreq(ACMCodecDB::kISAC);
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}
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// in a one-way communication we may never register send-codec.
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// However we like that the BWE to work properly so it has to
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// be initialized. The BWE is initialized when iSAC encoder is initialized.
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// Therefore, we need this.
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if (!encoder_initialized_) {
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// Since we don't require a valid rate or a valid packet size when
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// initializing the decoder, we set valid values before initializing encoder
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codec_params->codec_inst.rate = kIsacWbDefaultRate;
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codec_params->codec_inst.pacsize = kIsacPacSize960;
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if (InternalInitEncoder(codec_params) < 0) {
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return -1;
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}
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encoder_initialized_ = true;
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}
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst);
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}
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ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; }
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int16_t ACMISAC::InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) {
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// ISAC takes 10ms audio every time we call encoder, therefore,
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// it should be treated like codecs with 'basic coding block'
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// non-zero, and the following 'while-loop' should not be necessary.
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// However, due to a mistake in the codec the frame-size might change
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// at the first 10ms pushed in to iSAC if the bit-rate is low, this is
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// sort of a bug in iSAC. to address this we treat iSAC as the
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// following.
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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if (codec_inst_ptr_ == NULL) {
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return -1;
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}
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*bitstream_len_byte = 0;
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while ((*bitstream_len_byte == 0) && (in_audio_ix_read_ < frame_len_smpl_)) {
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if (in_audio_ix_read_ > in_audio_ix_write_) {
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// something is wrong.
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"The actual frame-size of iSAC appears to be larger that "
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"expected. All audio pushed in but no bit-stream is "
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"generated.");
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return -1;
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}
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*bitstream_len_byte = ACM_ISAC_ENCODE(
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codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_],
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reinterpret_cast<int16_t*>(bitstream));
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// increment the read index this tell the caller that how far
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// we have gone forward in reading the audio buffer
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in_audio_ix_read_ += samples_in_10ms_audio_;
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}
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if (*bitstream_len_byte == 0) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, unique_id_,
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"ISAC Has encoded the whole frame but no bit-stream is "
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"generated.");
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}
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// a packet is generated iSAC, is set in adaptive mode may change
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// the frame length and we like to update the bottleneck value as
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// well, although updating bottleneck is not crucial
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if ((*bitstream_len_byte > 0) && (isac_coding_mode_ == ADAPTIVE)) {
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ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
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}
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UpdateFrameLen();
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return *bitstream_len_byte;
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}
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int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
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// if rate is set to -1 then iSAC has to be in adaptive mode
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if (codec_params->codec_inst.rate == -1) {
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isac_coding_mode_ = ADAPTIVE;
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} else if ((codec_params->codec_inst.rate >= ISAC_MIN_RATE) &&
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(codec_params->codec_inst.rate <= ISAC_MAX_RATE)) {
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// sanity check that rate is in acceptable range
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isac_coding_mode_ = CHANNEL_INDEPENDENT;
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isac_current_bn_ = codec_params->codec_inst.rate;
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} else {
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return -1;
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}
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// we need to set the encoder sampling frequency.
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if (UpdateEncoderSampFreq((uint16_t)codec_params->codec_inst.plfreq) < 0) {
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return -1;
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}
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) {
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return -1;
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}
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// apply the frame-size and rate if operating in
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// channel-independent mode
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if (isac_coding_mode_ == CHANNEL_INDEPENDENT) {
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if (ACM_ISAC_CONTROL(codec_inst_ptr_->inst,
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codec_params->codec_inst.rate,
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codec_params->codec_inst.pacsize /
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(codec_params->codec_inst.plfreq / 1000)) < 0) {
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return -1;
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}
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} else {
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// We need this for adaptive case and has to be called
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// after initialization
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ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_);
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}
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frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst);
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return 0;
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}
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int16_t ACMISAC::InternalCreateEncoder() {
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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if (codec_inst_ptr_ == NULL) {
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return -1;
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}
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decoder_initialized_ = false;
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int16_t status = ACM_ISAC_CREATE(&(codec_inst_ptr_->inst));
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if (status < 0)
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codec_inst_ptr_->inst = NULL;
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return status;
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}
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int16_t ACMISAC::Transcode(uint8_t* bitstream,
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int16_t* bitstream_len_byte,
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int16_t q_bwe,
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int32_t rate,
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bool is_red) {
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int16_t jitter_info = 0;
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// transcode from a higher rate to lower rate sanity check
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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if (codec_inst_ptr_ == NULL) {
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return -1;
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}
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*bitstream_len_byte = ACM_ISAC_GETNEWBITSTREAM(
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codec_inst_ptr_->inst, q_bwe, jitter_info, rate,
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reinterpret_cast<int16_t*>(bitstream), (is_red) ? 1 : 0);
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if (*bitstream_len_byte < 0) {
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// error happened
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*bitstream_len_byte = 0;
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return -1;
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} else {
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return *bitstream_len_byte;
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}
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}
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void ACMISAC::UpdateFrameLen() {
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CriticalSectionScoped lock(codec_inst_crit_sect_.get());
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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(¤t_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
|