audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
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 0000000..d5a644d
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -0,0 +1,305 @@
+/*
+ * 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 <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.
+ int 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,
+ int channels,
+ int sample_rate_hz,
+ AudioDecoder* audio_decoder);
+
+ //
+ // 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_