modules/audio_coding/main/source/acm_g7291.cc - src/webrtc - Git at Google

 /*
  *  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/source/acm_g7291.h"

 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
 #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
 #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #ifdef WEBRTC_CODEC_G729_1
 // NOTE! G.729.1 is not included in the open-source package. Modify this file
 // or your codec API to match the function calls and names of used G.729.1 API
 // file.
 #include "g7291_interface.h"
 #endif

 namespace webrtc {

 namespace acm1 {

 #ifndef WEBRTC_CODEC_G729_1

 ACMG729_1::ACMG729_1(int16_t /* codec_id */)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       my_rate_(32000),
       flag_8khz_(0),
       flag_g729_mode_(0) {
   return;
 }

 ACMG729_1::~ACMG729_1() {
   return;
 }

 int16_t ACMG729_1::InternalEncode(
     uint8_t* /* bitstream */,
     int16_t* /* bitstream_len_byte */) {
   return -1;
 }

 int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
                               int16_t /* bitstream_len_byte */,
                               int16_t* /* audio */,
                               int16_t* /* audio_samples */,
                               int8_t* /* speech_type */) {
   return -1;
 }

 int16_t ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }

 int16_t ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   return -1;
 }

 int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
                             const CodecInst& /* codec_inst */) {
   return -1;
 }

 ACMGenericCodec* ACMG729_1::CreateInstance(void) {
   return NULL;
 }

 int16_t ACMG729_1::InternalCreateEncoder() {
   return -1;
 }

 void ACMG729_1::DestructEncoderSafe() {
   return;
 }

 int16_t ACMG729_1::InternalCreateDecoder() {
   return -1;
 }

 void ACMG729_1::DestructDecoderSafe() {
   return;
 }

 void ACMG729_1::InternalDestructEncoderInst(void* /* ptr_inst */) {
   return;
 }

 int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) {
   return -1;
 }

 #else     //===================== Actual Implementation =======================

 struct G729_1_inst_t_;

 ACMG729_1::ACMG729_1(int16_t codec_id)
     : encoder_inst_ptr_(NULL),
       decoder_inst_ptr_(NULL),
       my_rate_(32000),  // Default rate.
       flag_8khz_(0),
       flag_g729_mode_(0) {
   // TODO(tlegrand): We should add codec_id as a input variable to the
   // constructor of ACMGenericCodec.
   codec_id_ = codec_id;
   return;
 }

 ACMG729_1::~ACMG729_1() {
   if (encoder_inst_ptr_ != NULL) {
     WebRtcG7291_Free(encoder_inst_ptr_);
     encoder_inst_ptr_ = NULL;
   }
   if (decoder_inst_ptr_ != NULL) {
     WebRtcG7291_Free(decoder_inst_ptr_);
     decoder_inst_ptr_ = NULL;
   }
   return;
 }

 int16_t ACMG729_1::InternalEncode(uint8_t* bitstream,
                                   int16_t* bitstream_len_byte) {

   // Initialize before entering the loop
   int16_t num_encoded_samples = 0;
   *bitstream_len_byte = 0;

   int16_t byte_length_frame = 0;

   // Derive number of 20ms frames per encoded packet.
   // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
   int16_t num_20ms_frames = (frame_len_smpl_ / 320);
   // Byte length for the frame. +1 is for rate information.
   byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 -
       flag_g729_mode_);

   // The following might be revised if we have G729.1 Annex C (support for DTX);
   do {
     *bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_,
                                              &in_audio_[in_audio_ix_read_],
                                              (int16_t*) bitstream,
                                              my_rate_, num_20ms_frames);

     // increment the read index this tell the caller that how far
     // we have gone forward in reading the audio buffer
     in_audio_ix_read_ += 160;

     // sanity check
     if (*bitstream_len_byte < 0) {
       // error has happened
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                    "InternalEncode: Encode error for G729_1");
       *bitstream_len_byte = 0;
       return -1;
     }

     num_encoded_samples += 160;
   } while (*bitstream_len_byte == 0);

   // This criteria will change if we have Annex C.
   if (*bitstream_len_byte != byte_length_frame) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalEncode: Encode error for G729_1");
     *bitstream_len_byte = 0;
     return -1;
   }

   if (num_encoded_samples != frame_len_smpl_) {
     *bitstream_len_byte = 0;
     return -1;
   }

   return *bitstream_len_byte;
 }

 int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
                               int16_t /* bitstream_len_byte */,
                               int16_t* /* audio */,
                               int16_t* /* audio_samples */,
                               int8_t* /* speech_type */) {
   return 0;
 }

 int16_t ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* codec_params) {
   //set the bit rate and initialize
   my_rate_ = codec_params->codec_inst.rate;
   return SetBitRateSafe((uint32_t) my_rate_);
 }

 int16_t ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codec_params */) {
   if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalInitDecoder: init decoder failed for G729_1");
     return -1;
   }
   return 0;
 }

 int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
                             const CodecInst& codec_inst) {
   if (!decoder_initialized_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "CodeDef: Decoder uninitialized for G729_1");
     return -1;
   }

   // Fill up the structure by calling
   // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
   // Then call NetEQ to add the codec to it's
   // database.
   SET_CODEC_PAR((codec_def), kDecoderG729_1, codec_inst.pltype,
                 decoder_inst_ptr_, 16000);
   SET_G729_1_FUNCTIONS((codec_def));
   return 0;
 }

 ACMGenericCodec* ACMG729_1::CreateInstance(void) {
   return NULL;
 }

 int16_t ACMG729_1::InternalCreateEncoder() {
   if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateEncoder: create encoder failed for G729_1");
     return -1;
   }
   return 0;
 }

 void ACMG729_1::DestructEncoderSafe() {
   encoder_exist_ = false;
   encoder_initialized_ = false;
   if (encoder_inst_ptr_ != NULL) {
     WebRtcG7291_Free(encoder_inst_ptr_);
     encoder_inst_ptr_ = NULL;
   }
 }

 int16_t ACMG729_1::InternalCreateDecoder() {
   if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "InternalCreateDecoder: create decoder failed for G729_1");
     return -1;
   }
   return 0;
 }

 void ACMG729_1::DestructDecoderSafe() {
   decoder_exist_ = false;
   decoder_initialized_ = false;
   if (decoder_inst_ptr_ != NULL) {
     WebRtcG7291_Free(decoder_inst_ptr_);
     decoder_inst_ptr_ = NULL;
   }
 }

 void ACMG729_1::InternalDestructEncoderInst(void* ptr_inst) {
   if (ptr_inst != NULL) {
     // WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
   }
   return;
 }

 int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
   // allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
   //                22000, 24000, 26000, 28000, 30000, 32000};
   // TODO(tlegrand): This check exists in one other place two. Should be
   // possible to reuse code.
   switch (rate) {
     case 8000: {
       my_rate_ = 8000;
       break;
     }
     case 12000: {
       my_rate_ = 12000;
       break;
     }
     case 14000: {
       my_rate_ = 14000;
       break;
     }
     case 16000: {
       my_rate_ = 16000;
       break;
     }
     case 18000: {
       my_rate_ = 18000;
       break;
     }
     case 20000: {
       my_rate_ = 20000;
       break;
     }
     case 22000: {
       my_rate_ = 22000;
       break;
     }
     case 24000: {
       my_rate_ = 24000;
       break;
     }
     case 26000: {
       my_rate_ = 26000;
       break;
     }
     case 28000: {
       my_rate_ = 28000;
       break;
     }
     case 30000: {
       my_rate_ = 30000;
       break;
     }
     case 32000: {
       my_rate_ = 32000;
       break;
     }
     default: {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                    "SetBitRateSafe: Invalid rate G729_1");
       return -1;
     }
   }

   // Re-init with new rate
   if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_,
                               flag_g729_mode_) >= 0) {
     encoder_params_.codec_inst.rate = my_rate_;
     return 0;
   } else {
     return -1;
   }
 }

 #endif

 }  // namespace acm1

 }  // namespace webrtc
	/*
	* 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/source/acm_g7291.h"

	#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
	#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
	#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
	#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
	#include "webrtc/system_wrappers/interface/trace.h"
	#ifdef WEBRTC_CODEC_G729_1
	// NOTE! G.729.1 is not included in the open-source package. Modify this file
	// or your codec API to match the function calls and names of used G.729.1 API
	// file.
	#include "g7291_interface.h"
	#endif

	namespace webrtc {

	namespace acm1 {

	#ifndef WEBRTC_CODEC_G729_1

	ACMG729_1::ACMG729_1(int16_t /* codec_id */)
	: encoder_inst_ptr_(NULL),
	decoder_inst_ptr_(NULL),
	my_rate_(32000),
	flag_8khz_(0),
	flag_g729_mode_(0) {
	return;
	}

	ACMG729_1::~ACMG729_1() {
	return;
	}

	int16_t ACMG729_1::InternalEncode(
	uint8_t* /* bitstream */,
	int16_t* /* bitstream_len_byte */) {
	return -1;
	}

	int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
	int16_t /* bitstream_len_byte */,
	int16_t* /* audio */,
	int16_t* /* audio_samples */,
	int8_t* /* speech_type */) {
	return -1;
	}

	int16_t ACMG729_1::InternalInitEncoder(
	WebRtcACMCodecParams* /* codec_params */) {
	return -1;
	}

	int16_t ACMG729_1::InternalInitDecoder(
	WebRtcACMCodecParams* /* codec_params */) {
	return -1;
	}

	int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
	const CodecInst& /* codec_inst */) {
	return -1;
	}

	ACMGenericCodec* ACMG729_1::CreateInstance(void) {
	return NULL;
	}

	int16_t ACMG729_1::InternalCreateEncoder() {
	return -1;
	}

	void ACMG729_1::DestructEncoderSafe() {
	return;
	}

	int16_t ACMG729_1::InternalCreateDecoder() {
	return -1;
	}

	void ACMG729_1::DestructDecoderSafe() {
	return;
	}

	void ACMG729_1::InternalDestructEncoderInst(void* /* ptr_inst */) {
	return;
	}

	int16_t ACMG729_1::SetBitRateSafe(const int32_t /rate/) {
	return -1;
	}

	#else //===================== Actual Implementation =======================

	struct G729_1_inst_t_;

	ACMG729_1::ACMG729_1(int16_t codec_id)
	: encoder_inst_ptr_(NULL),
	decoder_inst_ptr_(NULL),
	my_rate_(32000), // Default rate.
	flag_8khz_(0),
	flag_g729_mode_(0) {
	// TODO(tlegrand): We should add codec_id as a input variable to the
	// constructor of ACMGenericCodec.
	codec_id_ = codec_id;
	return;
	}

	ACMG729_1::~ACMG729_1() {
	if (encoder_inst_ptr_ != NULL) {
	WebRtcG7291_Free(encoder_inst_ptr_);
	encoder_inst_ptr_ = NULL;
	}
	if (decoder_inst_ptr_ != NULL) {
	WebRtcG7291_Free(decoder_inst_ptr_);
	decoder_inst_ptr_ = NULL;
	}
	return;
	}

	int16_t ACMG729_1::InternalEncode(uint8_t* bitstream,
	int16_t* bitstream_len_byte) {

	// Initialize before entering the loop
	int16_t num_encoded_samples = 0;
	*bitstream_len_byte = 0;

	int16_t byte_length_frame = 0;

	// Derive number of 20ms frames per encoded packet.
	// [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
	int16_t num_20ms_frames = (frame_len_smpl_ / 320);
	// Byte length for the frame. +1 is for rate information.
	byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 -
	flag_g729_mode_);

	// The following might be revised if we have G729.1 Annex C (support for DTX);
	do {
	*bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_,
	&in_audio_[in_audio_ix_read_],
	(int16_t*) bitstream,
	my_rate_, num_20ms_frames);

	// increment the read index this tell the caller that how far
	// we have gone forward in reading the audio buffer
	in_audio_ix_read_ += 160;

	// sanity check
	if (*bitstream_len_byte < 0) {
	// error has happened
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"InternalEncode: Encode error for G729_1");
	*bitstream_len_byte = 0;
	return -1;
	}

	num_encoded_samples += 160;
	} while (*bitstream_len_byte == 0);

	// This criteria will change if we have Annex C.
	if (*bitstream_len_byte != byte_length_frame) {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"InternalEncode: Encode error for G729_1");
	*bitstream_len_byte = 0;
	return -1;
	}

	if (num_encoded_samples != frame_len_smpl_) {
	*bitstream_len_byte = 0;
	return -1;
	}

	return *bitstream_len_byte;
	}

	int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
	int16_t /* bitstream_len_byte */,
	int16_t* /* audio */,
	int16_t* /* audio_samples */,
	int8_t* /* speech_type */) {
	return 0;
	}

	int16_t ACMG729_1::InternalInitEncoder(
	WebRtcACMCodecParams* codec_params) {
	//set the bit rate and initialize
	my_rate_ = codec_params->codec_inst.rate;
	return SetBitRateSafe((uint32_t) my_rate_);
	}

	int16_t ACMG729_1::InternalInitDecoder(
	WebRtcACMCodecParams* /* codec_params */) {
	if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"InternalInitDecoder: init decoder failed for G729_1");
	return -1;
	}
	return 0;
	}

	int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
	const CodecInst& codec_inst) {
	if (!decoder_initialized_) {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"CodeDef: Decoder uninitialized for G729_1");
	return -1;
	}

	// Fill up the structure by calling
	// "SET_CODEC_PAR" & "SET_G729_FUNCTION."
	// Then call NetEQ to add the codec to it's
	// database.
	SET_CODEC_PAR((codec_def), kDecoderG729_1, codec_inst.pltype,
	decoder_inst_ptr_, 16000);
	SET_G729_1_FUNCTIONS((codec_def));
	return 0;
	}

	ACMGenericCodec* ACMG729_1::CreateInstance(void) {
	return NULL;
	}

	int16_t ACMG729_1::InternalCreateEncoder() {
	if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"InternalCreateEncoder: create encoder failed for G729_1");
	return -1;
	}
	return 0;
	}

	void ACMG729_1::DestructEncoderSafe() {
	encoder_exist_ = false;
	encoder_initialized_ = false;
	if (encoder_inst_ptr_ != NULL) {
	WebRtcG7291_Free(encoder_inst_ptr_);
	encoder_inst_ptr_ = NULL;
	}
	}

	int16_t ACMG729_1::InternalCreateDecoder() {
	if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"InternalCreateDecoder: create decoder failed for G729_1");
	return -1;
	}
	return 0;
	}

	void ACMG729_1::DestructDecoderSafe() {
	decoder_exist_ = false;
	decoder_initialized_ = false;
	if (decoder_inst_ptr_ != NULL) {
	WebRtcG7291_Free(decoder_inst_ptr_);
	decoder_inst_ptr_ = NULL;
	}
	}

	void ACMG729_1::InternalDestructEncoderInst(void* ptr_inst) {
	if (ptr_inst != NULL) {
	// WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
	}
	return;
	}

	int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
	// allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
	// 22000, 24000, 26000, 28000, 30000, 32000};
	// TODO(tlegrand): This check exists in one other place two. Should be
	// possible to reuse code.
	switch (rate) {
	case 8000: {
	my_rate_ = 8000;
	break;
	}
	case 12000: {
	my_rate_ = 12000;
	break;
	}
	case 14000: {
	my_rate_ = 14000;
	break;
	}
	case 16000: {
	my_rate_ = 16000;
	break;
	}
	case 18000: {
	my_rate_ = 18000;
	break;
	}
	case 20000: {
	my_rate_ = 20000;
	break;
	}
	case 22000: {
	my_rate_ = 22000;
	break;
	}
	case 24000: {
	my_rate_ = 24000;
	break;
	}
	case 26000: {
	my_rate_ = 26000;
	break;
	}
	case 28000: {
	my_rate_ = 28000;
	break;
	}
	case 30000: {
	my_rate_ = 30000;
	break;
	}
	case 32000: {
	my_rate_ = 32000;
	break;
	}
	default: {
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
	"SetBitRateSafe: Invalid rate G729_1");
	return -1;
	}
	}

	// Re-init with new rate
	if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_,
	flag_g729_mode_) >= 0) {
	encoder_params_.codec_inst.rate = my_rate_;
	return 0;
	} else {
	return -1;
	}
	}

	#endif

	} // namespace acm1

	} // namespace webrtc