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| author | Chih-hung Hsieh <chh@google.com> | 2016-01-20 17:50:13 +0000 |
|---|---|---|
| committer | android-build-merger <android-build-merger@google.com> | 2016-01-20 17:50:13 +0000 |
| commit | b3cb8ab4ede8bb77f0bdef2715efc2c1e6267072 (patch) | |
| tree | 28c4cf735dd5bd9cc8f1ccd06fff8a173b20d1cb /webrtc/modules/audio_coding/acm2/acm_receiver.h | |
| parent | a4acd9d6bc9b3b033d7d274316e75ee067df8d20 (diff) | |
| parent | 9a337512d97e37afc142dee4fd50a41b741a87d2 (diff) | |
| download | webrtc-nougat-dr1-release.tar.gz | |
Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release
am: 9a337512d9
* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
Add tests for verifying transport feedback for audio and video.
Eliminate defines in talk/
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Remove assert which was incorrectly added to TcpPort::OnSentPacket.
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Update with new default boringssl no-aes cipher suites. Re-enable tests.
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
Re-land: "Use an explicit identifier in Config"
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
Remove libfuzzer trybot from default trybot set.
Add ramp-up tests for transport sequence number with and w/o audio.
Delete remnants of non-square pixel support from cricket::VideoFrame.
Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Re-enable tests that failed under Linux_Msan.
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
Roll chromium_revision 346fea9..099be58 (369082:369139)
Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
Add build_protobuf variable.
...
Diffstat (limited to 'webrtc/modules/audio_coding/acm2/acm_receiver.h')
| -rw-r--r-- | webrtc/modules/audio_coding/acm2/acm_receiver.h | 307 |
1 files changed, 307 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h new file mode 100644 index 0000000000..b150612f69 --- /dev/null +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h @@ -0,0 +1,307 @@ +/* + * Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ + +#include <map> +#include <string> +#include <vector> + +#include "webrtc/base/array_view.h" +#include "webrtc/base/optional.h" +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/base/thread_annotations.h" +#include "webrtc/common_audio/vad/include/webrtc_vad.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" +#include "webrtc/modules/audio_coding/neteq/include/neteq.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct CodecInst; +class CriticalSectionWrapper; +class NetEq; + +namespace acm2 { + +class AcmReceiver { + public: + struct Decoder { + int acm_codec_id; + uint8_t payload_type; + // This field is meaningful for codecs where both mono and + // stereo versions are registered under the same ID. + size_t channels; + int sample_rate_hz; + }; + + // Constructor of the class + explicit AcmReceiver(const AudioCodingModule::Config& config); + + // Destructor of the class. + ~AcmReceiver(); + + // + // Inserts a payload with its associated RTP-header into NetEq. + // + // Input: + // - rtp_header : RTP header for the incoming payload containing + // information about payload type, sequence number, + // timestamp, SSRC and marker bit. + // - incoming_payload : Incoming audio payload. + // - length_payload : Length of incoming audio payload in bytes. + // + // Return value : 0 if OK. + // <0 if NetEq returned an error. + // + int InsertPacket(const WebRtcRTPHeader& rtp_header, + rtc::ArrayView<const uint8_t> incoming_payload); + + // + // Asks NetEq for 10 milliseconds of decoded audio. + // + // Input: + // -desired_freq_hz : specifies the sampling rate [Hz] of the output + // audio. If set -1 indicates to resampling is + // is required and the audio returned at the + // sampling rate of the decoder. + // + // Output: + // -audio_frame : an audio frame were output data and + // associated parameters are written to. + // + // Return value : 0 if OK. + // -1 if NetEq returned an error. + // + int GetAudio(int desired_freq_hz, AudioFrame* audio_frame); + + // + // Adds a new codec to the NetEq codec database. + // + // Input: + // - acm_codec_id : ACM codec ID; -1 means external decoder. + // - payload_type : payload type. + // - sample_rate_hz : sample rate. + // - audio_decoder : pointer to a decoder object. If it's null, then + // NetEq will internally create a decoder object + // based on the value of |acm_codec_id| (which + // mustn't be -1). Otherwise, NetEq will use the + // given decoder for the given payload type. NetEq + // won't take ownership of the decoder; it's up to + // the caller to delete it when it's no longer + // needed. + // + // Providing an existing decoder object here is + // necessary for external decoders, but may also be + // used for built-in decoders if NetEq doesn't have + // all the info it needs to construct them properly + // (e.g. iSAC, where the decoder needs to be paired + // with an encoder). + // + // Return value : 0 if OK. + // <0 if NetEq returned an error. + // + int AddCodec(int acm_codec_id, + uint8_t payload_type, + size_t channels, + int sample_rate_hz, + AudioDecoder* audio_decoder, + const std::string& name); + + // + // Sets a minimum delay for packet buffer. The given delay is maintained, + // unless channel condition dictates a higher delay. + // + // Input: + // - delay_ms : minimum delay in milliseconds. + // + // Return value : 0 if OK. + // <0 if NetEq returned an error. + // + int SetMinimumDelay(int delay_ms); + + // + // Sets a maximum delay [ms] for the packet buffer. The target delay does not + // exceed the given value, even if channel condition requires so. + // + // Input: + // - delay_ms : maximum delay in milliseconds. + // + // Return value : 0 if OK. + // <0 if NetEq returned an error. + // + int SetMaximumDelay(int delay_ms); + + // + // Get least required delay computed based on channel conditions. Note that + // this is before applying any user-defined limits (specified by calling + // (SetMinimumDelay() and/or SetMaximumDelay()). + // + int LeastRequiredDelayMs() const; + + // + // Resets the initial delay to zero. + // + void ResetInitialDelay(); + + // Returns the sample rate of the decoder associated with the last incoming + // packet. If no packet of a registered non-CNG codec has been received, the + // return value is empty. Also, if the decoder was unregistered since the last + // packet was inserted, the return value is empty. + rtc::Optional<int> last_packet_sample_rate_hz() const; + + // Returns last_output_sample_rate_hz from the NetEq instance. + int last_output_sample_rate_hz() const; + + // + // Get the current network statistics from NetEq. + // + // Output: + // - statistics : The current network statistics. + // + void GetNetworkStatistics(NetworkStatistics* statistics); + + // + // Enable post-decoding VAD. + // + void EnableVad(); + + // + // Disable post-decoding VAD. + // + void DisableVad(); + + // + // Returns whether post-decoding VAD is enabled (true) or disabled (false). + // + bool vad_enabled() const { return vad_enabled_; } + + // + // Flushes the NetEq packet and speech buffers. + // + void FlushBuffers(); + + // + // Removes a payload-type from the NetEq codec database. + // + // Input: + // - payload_type : the payload-type to be removed. + // + // Return value : 0 if OK. + // -1 if an error occurred. + // + int RemoveCodec(uint8_t payload_type); + + // + // Remove all registered codecs. + // + int RemoveAllCodecs(); + + // + // Set ID. + // + void set_id(int id); // TODO(turajs): can be inline. + + // + // Gets the RTP timestamp of the last sample delivered by GetAudio(). + // Returns true if the RTP timestamp is valid, otherwise false. + // + bool GetPlayoutTimestamp(uint32_t* timestamp); + + // + // Get the audio codec associated with the last non-CNG/non-DTMF received + // payload. If no non-CNG/non-DTMF packet is received -1 is returned, + // otherwise return 0. + // + int LastAudioCodec(CodecInst* codec) const; + + // + // Get a decoder given its registered payload-type. + // + // Input: + // -payload_type : the payload-type of the codec to be retrieved. + // + // Output: + // -codec : codec associated with the given payload-type. + // + // Return value : 0 if succeeded. + // -1 if failed, e.g. given payload-type is not + // registered. + // + int DecoderByPayloadType(uint8_t payload_type, + CodecInst* codec) const; + + // + // Enable NACK and set the maximum size of the NACK list. If NACK is already + // enabled then the maximum NACK list size is modified accordingly. + // + // Input: + // -max_nack_list_size : maximum NACK list size + // should be positive (none zero) and less than or + // equal to |Nack::kNackListSizeLimit| + // Return value + // : 0 if succeeded. + // -1 if failed + // + int EnableNack(size_t max_nack_list_size); + + // Disable NACK. + void DisableNack(); + + // + // Get a list of packets to be retransmitted. + // + // Input: + // -round_trip_time_ms : estimate of the round-trip-time (in milliseconds). + // Return value : list of packets to be retransmitted. + // + std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const; + + // + // Get statistics of calls to GetAudio(). + void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; + + private: + const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header, + uint8_t payload_type) const + EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); + + uint32_t NowInTimestamp(int decoder_sampling_rate) const; + + rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; + int id_; // TODO(henrik.lundin) Make const. + const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_); + AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); + ACMResampler resampler_ GUARDED_BY(crit_sect_); + // Used in GetAudio, declared as member to avoid allocating every 10ms. + // TODO(henrik.lundin) Stack-allocate in GetAudio instead? + rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_); + rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_); + CallStatistics call_stats_ GUARDED_BY(crit_sect_); + NetEq* neteq_; + // Decoders map is keyed by payload type + std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_); + bool vad_enabled_; + Clock* clock_; // TODO(henrik.lundin) Make const if possible. + bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); + rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); +}; + +} // namespace acm2 + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |