diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/g711')
7 files changed, 800 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc new file mode 100644 index 0000000000..46ac671b30 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc @@ -0,0 +1,102 @@ +/* + * Copyright (c) 2015 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 "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" + +#include <utility> + +#include "modules/audio_coding/codecs/g711/g711_interface.h" +#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" + +namespace webrtc { + +void AudioDecoderPcmU::Reset() {} + +std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + return LegacyEncodedAudioFrame::SplitBySamples( + this, std::move(payload), timestamp, 8 * num_channels_, 8); +} + +int AudioDecoderPcmU::SampleRateHz() const { + return 8000; +} + +size_t AudioDecoderPcmU::Channels() const { + return num_channels_; +} + +int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); + // Adjust the encoded length down to ensure the same number of samples in each + // channel. + const size_t encoded_len_adjusted = + PacketDuration(encoded, encoded_len) * + Channels(); // 1 byte per sample per channel + int16_t temp_type = 1; // Default is speech. + size_t ret = + WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return static_cast<int>(ret); +} + +int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / Channels()); +} + +void AudioDecoderPcmA::Reset() {} + +std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + return LegacyEncodedAudioFrame::SplitBySamples( + this, std::move(payload), timestamp, 8 * num_channels_, 8); +} + +int AudioDecoderPcmA::SampleRateHz() const { + return 8000; +} + +size_t AudioDecoderPcmA::Channels() const { + return num_channels_; +} + +int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz); + // Adjust the encoded length down to ensure the same number of samples in each + // channel. + const size_t encoded_len_adjusted = + PacketDuration(encoded, encoded_len) * + Channels(); // 1 byte per sample per channel + int16_t temp_type = 1; // Default is speech. + size_t ret = + WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return static_cast<int>(ret); +} + +int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + // One encoded byte per sample per channel. + return static_cast<int>(encoded_len / Channels()); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h new file mode 100644 index 0000000000..3fa42cba30 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h @@ -0,0 +1,81 @@ +/* + * Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_ +#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <vector> + +#include "api/audio_codecs/audio_decoder.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +class AudioDecoderPcmU final : public AudioDecoder { + public: + explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { + RTC_DCHECK_GE(num_channels, 1); + } + + AudioDecoderPcmU(const AudioDecoderPcmU&) = delete; + AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete; + + void Reset() override; + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, + uint32_t timestamp) override; + int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; + int SampleRateHz() const override; + size_t Channels() const override; + + protected: + int DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) override; + + private: + const size_t num_channels_; +}; + +class AudioDecoderPcmA final : public AudioDecoder { + public: + explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { + RTC_DCHECK_GE(num_channels, 1); + } + + AudioDecoderPcmA(const AudioDecoderPcmA&) = delete; + AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete; + + void Reset() override; + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, + uint32_t timestamp) override; + int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; + int SampleRateHz() const override; + size_t Channels() const override; + + protected: + int DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) override; + + private: + const size_t num_channels_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc new file mode 100644 index 0000000000..65e2da479d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -0,0 +1,126 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" + +#include <cstdint> + +#include "modules/audio_coding/codecs/g711/g711_interface.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +bool AudioEncoderPcm::Config::IsOk() const { + return (frame_size_ms % 10 == 0) && (num_channels >= 1); +} + +AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) + : sample_rate_hz_(sample_rate_hz), + num_channels_(config.num_channels), + payload_type_(config.payload_type), + num_10ms_frames_per_packet_( + static_cast<size_t>(config.frame_size_ms / 10)), + full_frame_samples_(config.num_channels * config.frame_size_ms * + sample_rate_hz / 1000), + first_timestamp_in_buffer_(0) { + RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; + RTC_CHECK_EQ(config.frame_size_ms % 10, 0) + << "Frame size must be an integer multiple of 10 ms."; + speech_buffer_.reserve(full_frame_samples_); +} + +AudioEncoderPcm::~AudioEncoderPcm() = default; + +int AudioEncoderPcm::SampleRateHz() const { + return sample_rate_hz_; +} + +size_t AudioEncoderPcm::NumChannels() const { + return num_channels_; +} + +size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const { + return num_10ms_frames_per_packet_; +} + +size_t AudioEncoderPcm::Max10MsFramesInAPacket() const { + return num_10ms_frames_per_packet_; +} + +int AudioEncoderPcm::GetTargetBitrate() const { + return static_cast<int>(8 * BytesPerSample() * SampleRateHz() * + NumChannels()); +} + +AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl( + uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) { + if (speech_buffer_.empty()) { + first_timestamp_in_buffer_ = rtp_timestamp; + } + speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end()); + if (speech_buffer_.size() < full_frame_samples_) { + return EncodedInfo(); + } + RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_); + EncodedInfo info; + info.encoded_timestamp = first_timestamp_in_buffer_; + info.payload_type = payload_type_; + info.encoded_bytes = encoded->AppendData( + full_frame_samples_ * BytesPerSample(), + [&](rtc::ArrayView<uint8_t> encoded) { + return EncodeCall(&speech_buffer_[0], full_frame_samples_, + encoded.data()); + }); + speech_buffer_.clear(); + info.encoder_type = GetCodecType(); + return info; +} + +void AudioEncoderPcm::Reset() { + speech_buffer_.clear(); +} + +absl::optional<std::pair<TimeDelta, TimeDelta>> +AudioEncoderPcm::GetFrameLengthRange() const { + return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10), + TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}}; +} + +size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) { + return WebRtcG711_EncodeA(audio, input_len, encoded); +} + +size_t AudioEncoderPcmA::BytesPerSample() const { + return 1; +} + +AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const { + return AudioEncoder::CodecType::kPcmA; +} + +size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) { + return WebRtcG711_EncodeU(audio, input_len, encoded); +} + +size_t AudioEncoderPcmU::BytesPerSample() const { + return 1; +} + +AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const { + return AudioEncoder::CodecType::kPcmU; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h new file mode 100644 index 0000000000..d50be4b457 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -0,0 +1,128 @@ +/* + * 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 MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ +#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ + +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/units/time_delta.h" + +namespace webrtc { + +class AudioEncoderPcm : public AudioEncoder { + public: + struct Config { + public: + bool IsOk() const; + + int frame_size_ms; + size_t num_channels; + int payload_type; + + protected: + explicit Config(int pt) + : frame_size_ms(20), num_channels(1), payload_type(pt) {} + }; + + ~AudioEncoderPcm() override; + + int SampleRateHz() const override; + size_t NumChannels() const override; + size_t Num10MsFramesInNextPacket() const override; + size_t Max10MsFramesInAPacket() const override; + int GetTargetBitrate() const override; + void Reset() override; + absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange() + const override; + + protected: + AudioEncoderPcm(const Config& config, int sample_rate_hz); + + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) override; + + virtual size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) = 0; + + virtual size_t BytesPerSample() const = 0; + + // Used to set EncodedInfoLeaf::encoder_type in + // AudioEncoderPcm::EncodeImpl + virtual AudioEncoder::CodecType GetCodecType() const = 0; + + private: + const int sample_rate_hz_; + const size_t num_channels_; + const int payload_type_; + const size_t num_10ms_frames_per_packet_; + const size_t full_frame_samples_; + std::vector<int16_t> speech_buffer_; + uint32_t first_timestamp_in_buffer_; +}; + +class AudioEncoderPcmA final : public AudioEncoderPcm { + public: + struct Config : public AudioEncoderPcm::Config { + Config() : AudioEncoderPcm::Config(8) {} + }; + + explicit AudioEncoderPcmA(const Config& config) + : AudioEncoderPcm(config, kSampleRateHz) {} + + AudioEncoderPcmA(const AudioEncoderPcmA&) = delete; + AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete; + + protected: + size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) override; + + size_t BytesPerSample() const override; + + AudioEncoder::CodecType GetCodecType() const override; + + private: + static const int kSampleRateHz = 8000; +}; + +class AudioEncoderPcmU final : public AudioEncoderPcm { + public: + struct Config : public AudioEncoderPcm::Config { + Config() : AudioEncoderPcm::Config(0) {} + }; + + explicit AudioEncoderPcmU(const Config& config) + : AudioEncoderPcm(config, kSampleRateHz) {} + + AudioEncoderPcmU(const AudioEncoderPcmU&) = delete; + AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete; + + protected: + size_t EncodeCall(const int16_t* audio, + size_t input_len, + uint8_t* encoded) override; + + size_t BytesPerSample() const override; + + AudioEncoder::CodecType GetCodecType() const override; + + private: + static const int kSampleRateHz = 8000; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c new file mode 100644 index 0000000000..5fe1692ccb --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c @@ -0,0 +1,59 @@ +/* + * Copyright (c) 2011 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 <string.h> + +#include "modules/third_party/g711/g711.h" +#include "modules/audio_coding/codecs/g711/g711_interface.h" + +size_t WebRtcG711_EncodeA(const int16_t* speechIn, + size_t len, + uint8_t* encoded) { + size_t n; + for (n = 0; n < len; n++) + encoded[n] = linear_to_alaw(speechIn[n]); + return len; +} + +size_t WebRtcG711_EncodeU(const int16_t* speechIn, + size_t len, + uint8_t* encoded) { + size_t n; + for (n = 0; n < len; n++) + encoded[n] = linear_to_ulaw(speechIn[n]); + return len; +} + +size_t WebRtcG711_DecodeA(const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType) { + size_t n; + for (n = 0; n < len; n++) + decoded[n] = alaw_to_linear(encoded[n]); + *speechType = 1; + return len; +} + +size_t WebRtcG711_DecodeU(const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType) { + size_t n; + for (n = 0; n < len; n++) + decoded[n] = ulaw_to_linear(encoded[n]); + *speechType = 1; + return len; +} + +int16_t WebRtcG711_Version(char* version, int16_t lenBytes) { + strncpy(version, "2.0.0", lenBytes); + return 0; +} diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h new file mode 100644 index 0000000000..c92e6cc1c8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h @@ -0,0 +1,136 @@ +/* + * Copyright (c) 2011 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 MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_ +#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_ + +#include <stddef.h> +#include <stdint.h> + +// Comfort noise constants +#define G711_WEBRTC_SPEECH 1 +#define G711_WEBRTC_CNG 2 + +#ifdef __cplusplus +extern "C" { +#endif + +/**************************************************************************** + * WebRtcG711_EncodeA(...) + * + * This function encodes a G711 A-law frame and inserts it into a packet. + * Input speech length has be of any length. + * + * Input: + * - speechIn : Input speech vector + * - len : Samples in speechIn + * + * Output: + * - encoded : The encoded data vector + * + * Return value : Length (in bytes) of coded data. + * Always equal to len input parameter. + */ + +size_t WebRtcG711_EncodeA(const int16_t* speechIn, + size_t len, + uint8_t* encoded); + +/**************************************************************************** + * WebRtcG711_EncodeU(...) + * + * This function encodes a G711 U-law frame and inserts it into a packet. + * Input speech length has be of any length. + * + * Input: + * - speechIn : Input speech vector + * - len : Samples in speechIn + * + * Output: + * - encoded : The encoded data vector + * + * Return value : Length (in bytes) of coded data. + * Always equal to len input parameter. + */ + +size_t WebRtcG711_EncodeU(const int16_t* speechIn, + size_t len, + uint8_t* encoded); + +/**************************************************************************** + * WebRtcG711_DecodeA(...) + * + * This function decodes a packet G711 A-law frame. + * + * Input: + * - encoded : Encoded data + * - len : Bytes in encoded vector + * + * Output: + * - decoded : The decoded vector + * - speechType : 1 normal, 2 CNG (for G711 it should + * always return 1 since G711 does not have a + * built-in DTX/CNG scheme) + * + * Return value : >0 - Samples in decoded vector + * -1 - Error + */ + +size_t WebRtcG711_DecodeA(const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType); + +/**************************************************************************** + * WebRtcG711_DecodeU(...) + * + * This function decodes a packet G711 U-law frame. + * + * Input: + * - encoded : Encoded data + * - len : Bytes in encoded vector + * + * Output: + * - decoded : The decoded vector + * - speechType : 1 normal, 2 CNG (for G711 it should + * always return 1 since G711 does not have a + * built-in DTX/CNG scheme) + * + * Return value : >0 - Samples in decoded vector + * -1 - Error + */ + +size_t WebRtcG711_DecodeU(const uint8_t* encoded, + size_t len, + int16_t* decoded, + int16_t* speechType); + +/********************************************************************** + * WebRtcG711_Version(...) + * + * This function gives the version string of the G.711 codec. + * + * Input: + * - lenBytes: the size of Allocated space (in Bytes) where + * the version number is written to (in string format). + * + * Output: + * - version: Pointer to a buffer where the version number is + * written to. + * + */ + +int16_t WebRtcG711_Version(char* version, int16_t lenBytes); + +#ifdef __cplusplus +} +#endif + +#endif // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_ diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc new file mode 100644 index 0000000000..f3a42f5d79 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc @@ -0,0 +1,168 @@ +/* + * 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. + */ + +/* + * testG711.cpp : Defines the entry point for the console application. + */ + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> + +/* include API */ +#include "modules/audio_coding/codecs/g711/g711_interface.h" + +/* Runtime statistics */ +#include <time.h> +#define CLOCKS_PER_SEC_G711 1000 + +/* function for reading audio data from PCM file */ +bool readframe(int16_t* data, FILE* inp, size_t length) { + size_t rlen = fread(data, sizeof(int16_t), length, inp); + if (rlen >= length) + return false; + memset(data + rlen, 0, (length - rlen) * sizeof(int16_t)); + return true; +} + +int main(int argc, char* argv[]) { + char inname[80], outname[40], bitname[40]; + FILE* inp; + FILE* outp; + FILE* bitp = NULL; + int framecnt; + bool endfile; + + size_t framelength = 80; + + /* Runtime statistics */ + double starttime; + double runtime; + double length_file; + + size_t stream_len = 0; + int16_t shortdata[480]; + int16_t decoded[480]; + uint8_t streamdata[1000]; + int16_t speechType[1]; + char law[2]; + char versionNumber[40]; + + /* handling wrong input arguments in the command line */ + if ((argc != 5) && (argc != 6)) { + printf("\n\nWrong number of arguments or flag values.\n\n"); + + printf("\n"); + printf("\nG.711 test application\n\n"); + printf("Usage:\n\n"); + printf("./testG711.exe framelength law infile outfile \n\n"); + printf("framelength: Framelength in samples.\n"); + printf("law : Coding law, A och u.\n"); + printf("infile : Normal speech input file\n"); + printf("outfile : Speech output file\n\n"); + printf("outbits : Output bitstream file [optional]\n\n"); + exit(0); + } + + /* Get version and print */ + WebRtcG711_Version(versionNumber, 40); + + printf("-----------------------------------\n"); + printf("G.711 version: %s\n\n", versionNumber); + /* Get frame length */ + int framelength_int = atoi(argv[1]); + if (framelength_int < 0) { + printf(" G.722: Invalid framelength %d.\n", framelength_int); + exit(1); + } + framelength = static_cast<size_t>(framelength_int); + + /* Get compression law */ + strcpy(law, argv[2]); + + /* Get Input and Output files */ + sscanf(argv[3], "%s", inname); + sscanf(argv[4], "%s", outname); + if (argc == 6) { + sscanf(argv[5], "%s", bitname); + if ((bitp = fopen(bitname, "wb")) == NULL) { + printf(" G.711: Cannot read file %s.\n", bitname); + exit(1); + } + } + + if ((inp = fopen(inname, "rb")) == NULL) { + printf(" G.711: Cannot read file %s.\n", inname); + exit(1); + } + if ((outp = fopen(outname, "wb")) == NULL) { + printf(" G.711: Cannot write file %s.\n", outname); + exit(1); + } + printf("\nInput: %s\nOutput: %s\n", inname, outname); + if (argc == 6) { + printf("\nBitfile: %s\n", bitname); + } + + starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */ + + /* Initialize encoder and decoder */ + framecnt = 0; + endfile = false; + while (!endfile) { + framecnt++; + /* Read speech block */ + endfile = readframe(shortdata, inp, framelength); + + /* G.711 encoding */ + if (!strcmp(law, "A")) { + /* A-law encoding */ + stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata); + if (argc == 6) { + /* Write bits to file */ + if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) != + stream_len) { + return -1; + } + } + WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType); + } else if (!strcmp(law, "u")) { + /* u-law encoding */ + stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata); + if (argc == 6) { + /* Write bits to file */ + if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) != + stream_len) { + return -1; + } + } + WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType); + } else { + printf("Wrong law mode\n"); + exit(1); + } + /* Write coded speech to file */ + if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) { + return -1; + } + } + + runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime); + length_file = ((double)framecnt * (double)framelength / 8000); + printf("\n\nLength of speech file: %.1f s\n", length_file); + printf("Time to run G.711: %.2f s (%.2f %% of realtime)\n\n", runtime, + (100 * runtime / length_file)); + printf("---------------------END----------------------\n"); + + fclose(inp); + fclose(outp); + + return 0; +} |