2014-12-10 07:29:08 +00:00
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/*
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* Copyright (c) 2014 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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2017-09-15 06:47:31 +02:00
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#ifndef MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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2014-12-10 07:29:08 +00:00
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2016-02-14 01:10:03 -08:00
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#include <memory>
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2014-12-10 07:29:08 +00:00
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#include <vector>
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2017-09-15 06:47:31 +02:00
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#include "api/audio_codecs/audio_encoder.h"
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#include "common_audio/vad/include/vad.h"
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#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
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#include "rtc_base/constructormagic.h"
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2014-12-10 07:29:08 +00:00
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namespace webrtc {
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class Vad;
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2015-01-27 20:53:56 +00:00
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class AudioEncoderCng final : public AudioEncoder {
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2014-12-10 07:29:08 +00:00
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public:
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struct Config {
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2016-03-08 06:01:31 -08:00
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Config();
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Config(Config&&);
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~Config();
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2014-12-10 07:29:08 +00:00
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bool IsOk() const;
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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size_t num_channels = 1;
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2015-09-08 05:57:53 -07:00
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int payload_type = 13;
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2016-03-08 06:01:31 -08:00
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std::unique_ptr<AudioEncoder> speech_encoder;
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2015-09-08 05:57:53 -07:00
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Vad::Aggressiveness vad_mode = Vad::kVadNormal;
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int sid_frame_interval_ms = 100;
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int num_cng_coefficients = 8;
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2014-12-10 07:29:08 +00:00
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// The Vad pointer is mainly for testing. If a NULL pointer is passed, the
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// AudioEncoderCng creates (and destroys) a Vad object internally. If an
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// object is passed, the AudioEncoderCng assumes ownership of the Vad
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// object.
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2015-09-08 05:57:53 -07:00
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Vad* vad = nullptr;
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2014-12-10 07:29:08 +00:00
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};
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2016-03-08 06:01:31 -08:00
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explicit AudioEncoderCng(Config&& config);
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2015-03-04 12:58:35 +00:00
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~AudioEncoderCng() override;
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2014-12-10 07:29:08 +00:00
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2015-03-04 12:58:35 +00:00
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int SampleRateHz() const override;
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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size_t NumChannels() const override;
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2015-02-18 12:00:32 +00:00
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int RtpTimestampRateHz() const override;
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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2015-06-18 14:58:34 +02:00
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int GetTargetBitrate() const override;
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2016-03-04 00:54:32 -08:00
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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2015-09-08 05:57:53 -07:00
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void Reset() override;
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bool SetFec(bool enable) override;
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bool SetDtx(bool enable) override;
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bool SetApplication(Application application) override;
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2015-09-08 23:15:33 -07:00
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void SetMaxPlaybackRate(int frequency_hz) override;
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2016-06-23 03:58:36 -07:00
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rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
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override;
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2016-11-30 06:49:59 -08:00
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void OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) override;
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2017-03-23 15:29:50 -07:00
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void OnReceivedUplinkRecoverablePacketLossFraction(
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float uplink_recoverable_packet_loss_fraction) override;
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2017-01-12 10:17:38 -08:00
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void OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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2018-06-19 13:26:36 +02:00
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absl::optional<int64_t> bwe_period_ms) override;
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2014-12-10 07:29:08 +00:00
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private:
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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EncodedInfo EncodePassive(size_t frames_to_encode,
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2016-03-01 00:41:31 -08:00
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rtc::Buffer* encoded);
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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EncodedInfo EncodeActive(size_t frames_to_encode,
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2016-03-01 00:41:31 -08:00
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rtc::Buffer* encoded);
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2015-03-10 15:41:26 +00:00
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size_t SamplesPer10msFrame() const;
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2014-12-10 07:29:08 +00:00
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2016-03-08 06:01:31 -08:00
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std::unique_ptr<AudioEncoder> speech_encoder_;
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2014-12-10 07:29:08 +00:00
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const int cng_payload_type_;
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const int num_cng_coefficients_;
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2015-09-08 05:57:53 -07:00
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const int sid_frame_interval_ms_;
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2014-12-10 07:29:08 +00:00
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std::vector<int16_t> speech_buffer_;
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2015-05-22 15:13:41 +02:00
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std::vector<uint32_t> rtp_timestamps_;
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2014-12-10 07:29:08 +00:00
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bool last_frame_active_;
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2016-02-14 01:10:03 -08:00
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std::unique_ptr<Vad> vad_;
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2016-04-25 07:55:58 -07:00
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std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
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2015-09-22 14:06:29 -07:00
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
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2014-12-10 07:29:08 +00:00
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};
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} // namespace webrtc
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2015-09-22 14:06:29 -07:00
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2017-09-15 06:47:31 +02:00
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#endif // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
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