Merge changes I5302146e,Ib9b96c8c into main (cbda9520) · Commits · e / os / android_packages_modules_Bluetooth

system/audio/Android.bp

+0 −4

Original line number	Diff line number	Diff line
		@@ -14,9 +14,6 @@ cc_library_static {
		"asrc/asrc_resampler.cc",
		"asrc/asrc_tables.cc",
		],
		include_dirs: [
		"packages/modules/Bluetooth/system/gd",
		],
		shared_libs: [
		"libchrome",
		],
		@@ -44,7 +41,6 @@ cc_library_host_shared {
		stl: "libc++_static",
		include_dirs: [
		"packages/modules/Bluetooth/system",
		"packages/modules/Bluetooth/system/gd",
		],
		generated_headers: [
		"BluetoothGeneratedDumpsysDataSchema_h",

system/audio/asrc/asrc_resampler.cc

+10 −12

Original line number	Diff line number	Diff line
		@@ -19,15 +19,15 @@
		#include <base/logging.h>
		#include <base/strings/stringprintf.h>

		#include <algorithm>
		#include <cmath>
		#include <utility>

		#include "asrc_tables.h"
		#include "hal/nocp_iso_clocker.h"

		namespace bluetooth::audio::asrc {

		class SourceAudioHalAsrc::ClockRecovery : ::bluetooth::hal::NocpIsoHandler {
		class SourceAudioHalAsrc::ClockRecovery : public ClockHandler {
		const int interval_;

		std::mutex mutex_;
		@@ -113,8 +113,7 @@ class SourceAudioHalAsrc::ClockRecovery : ::bluetooth::hal::NocpIsoHandler {
		}

		int dt_current = int(timestamp_us - link.local_time);
		if (std::abs(dt_current) < std::abs(link.decim_dt[1]))
		link.decim_dt[1] = dt_current;
		link.decim_dt[1] = std::min(link.decim_dt[1], dt_current);

		if (link.local_time - link.decim_t0 < 1000 * 1000) return;

		@@ -237,11 +236,9 @@ class SourceAudioHalAsrc::ClockRecovery : ::bluetooth::hal::NocpIsoHandler {
		state_{
		.link = {{.state = LinkState::RESET}, {.state = LinkState::RESET}},
		.active_link_id = -1},
		reference_timing_{0, 0, 0} {
		::bluetooth::hal::NocpIsoClocker::Register(this);
		}
		reference_timing_{0, 0, 0} {}

		~ClockRecovery() override { ::bluetooth::hal::NocpIsoClocker::Unregister(); }
		~ClockRecovery() override {}

		__attribute__((no_sanitize("integer"))) uint32_t Convert(
		uint32_t stream_time) {
		@@ -466,16 +463,16 @@ inline int32_t SourceAudioHalAsrc::Resampler::Filter(const int32_t* in,

		#endif

		SourceAudioHalAsrc::SourceAudioHalAsrc(int channels, int sample_rate,
		int bit_depth, int interval_us,
		int num_burst_buffers,
		int burst_delay_ms)
		SourceAudioHalAsrc::SourceAudioHalAsrc(
		std::shared_ptr<ClockSource> clock_source, int channels, int sample_rate,
		int bit_depth, int interval_us, int num_burst_buffers, int burst_delay_ms)
		: sample_rate_(sample_rate),
		bit_depth_(bit_depth),
		interval_us_(interval_us),
		stream_us_(0),
		drift_us_(0),
		out_counter_(0),
		clock_source_(std::move(clock_source)),
		resampler_pos_{0, 0} {
		buffers_size_ = 0;

		@@ -509,6 +506,7 @@ SourceAudioHalAsrc::SourceAudioHalAsrc(int channels, int sample_rate,
		// when the PCM bit_depth is higher than 16 bits.

		clock_recovery_ = std::make_unique<ClockRecovery>(interval_us_);
		clock_source_->Bind(clock_recovery_.get());
		resamplers_ = std::make_unique<std::vector<Resampler>>(channels, bit_depth_);

		// Deduct from the PCM stream characteristics, the size of the pool buffers

system/audio/asrc/asrc_resampler.h

+17 −3

Original line number	Diff line number	Diff line
		@@ -23,6 +23,19 @@

		namespace bluetooth::audio::asrc {

		class ClockHandler {
		public:
		virtual ~ClockHandler() = default;
		virtual void OnEvent(uint32_t timestamp, int link_id,
		int num_of_completed_packets) = 0;
		};

		class ClockSource {
		public:
		virtual ~ClockSource() = default;
		virtual void Bind(ClockHandler*) = 0;
		};

		class SourceAudioHalAsrc {
		public:
		// The Asynchronous Sample Rate Conversion (ASRC) is set up from the PCM
		@@ -36,9 +49,9 @@ class SourceAudioHalAsrc {
		// `burst_delay_ms` helps to ensure that the synchronization with the
		// transmission intervals is done.

		SourceAudioHalAsrc(int channels, int sample_rate, int bit_depth,
		int interval_us, int num_burst_buffers = 2,
		int burst_delay_ms = 500);
		SourceAudioHalAsrc(std::shared_ptr<ClockSource> clock_source, int channels,
		int sample_rate, int bit_depth, int interval_us,
		int num_burst_buffers = 2, int burst_delay_ms = 500);

		~SourceAudioHalAsrc();

		@@ -74,6 +87,7 @@ class SourceAudioHalAsrc {

		class ClockRecovery;
		std::unique_ptr<ClockRecovery> clock_recovery_;
		std::shared_ptr<ClockSource> clock_source_;

		class Resampler;
		std::unique_ptr<std::vector<Resampler>> resamplers_;

system/audio/asrc/asrc_resampler_test.cc

+6 −6

Original line number	Diff line number	Diff line
		@@ -19,17 +19,17 @@
		#include <cstdio>
		#include <iostream>

		namespace bluetooth::hal {
		void NocpIsoClocker::Register(NocpIsoHandler*) {}
		void NocpIsoClocker::Unregister() {}
		} // namespace bluetooth::hal

		namespace bluetooth::audio::asrc {

		class MockClockSource : public ClockSource {
		void Bind(ClockHandler*) override {}
		};

		class SourceAudioHalAsrcTest : public SourceAudioHalAsrc {
		public:
		SourceAudioHalAsrcTest(int channels, int bitdepth)
		: SourceAudioHalAsrc(channels, 48000, bitdepth, 10000) {}
		: SourceAudioHalAsrc(std::make_unique<MockClockSource>(), channels, 48000,
		bitdepth, 10000) {}

		template <typename T>
		void Resample(double ratio, const T* in, size_t in_length, size_t* in_count,

system/bta/hearing_aid/hearing_aid.cc

+84 −1

Original line number	Diff line number	Diff line
		@@ -32,12 +32,14 @@
		#include <mutex>
		#include <vector>

		#include "audio/asrc/asrc_resampler.h"
		#include "bta/include/bta_gatt_api.h"
		#include "bta/include/bta_gatt_queue.h"
		#include "bta/include/bta_hearing_aid_api.h"
		#include "btm_iso_api.h"
		#include "device/include/controller.h"
		#include "embdrv/g722/g722_enc_dec.h"
		#include "hal/link_clocker.h"
		#include "hardware/bt_gatt_types.h"
		#include "include/check.h"
		#include "internal_include/bt_trace.h"
		@@ -275,6 +277,13 @@ class HearingAidImpl : public HearingAid {
		const int DROP_FREQUENCY_THRESHOLD =
		bluetooth::common::init_flags::get_asha_packet_drop_frequency_threshold();

		// Resampler context for audio stream.
		// Clock recovery uses L2CAP Flow Control Credit Ind acknowledgments
		// from either the left or right connection, whichever is first
		// connected.
		std::shared_ptr<bluetooth::hal::L2capCreditIndEvents> asrc_clock_source;
		std::unique_ptr<bluetooth::audio::asrc::SourceAudioHalAsrc> asrc;

		public:
		~HearingAidImpl() override = default;

		@@ -354,6 +363,51 @@ class HearingAidImpl : public HearingAid {
		}
		}

		// Reset and configure the ASHA resampling context using the input device
		// devices as reference for the BT clock estimation.
		void ConfigureAsrc() {
		if (!IS_FLAG_ENABLED(asha_asrc)) {
		log::info("Asha resampling disabled: feature flag off");
		return;
		}

		// Create a new ASRC context if required.
		if (asrc == nullptr) {
		asrc_clock_source =
		std::make_shared<bluetooth::hal::L2capCreditIndEvents>();
		asrc = std::make_unique<bluetooth::audio::asrc::SourceAudioHalAsrc>(
		asrc_clock_source, /channels/ 2,
		/sample_rate/ codec_in_use == CODEC_G722_24KHZ ? 24000 : 16000,
		/bit_depth/ 16,
		/interval_us/ default_data_interval_ms * 1000,
		/num_burst_buffers/ 0,
		/burst_delay/ 0);
		}

		for (auto& device : hearingDevices.devices) {
		if (!device.accepting_audio) {
		continue;
		}

		uint16_t lcid = GAP_ConnGetL2CAPCid(device.gap_handle);
		uint16_t rcid = 0;
		L2CA_GetRemoteCid(lcid, &rcid);

		auto conn = btm_acl_for_bda(device.address, BT_TRANSPORT_LE);
		log::info("Updating ASRC context for handle=0x{:x}, cid=0x{:x}",
		conn->Handle(), rcid);

		asrc_clock_source->Update(device.isLeft(), conn->Handle(), rcid);
		}
		}

		// Reset the ASHA resampling context.
		void ResetAsrc() {
		log::info("Resetting the Asha resampling context");
		asrc_clock_source = nullptr;
		asrc = nullptr;
		}

		uint16_t UpdateBleConnParams(const RawAddress& address) {
		/* List of parameters that depends on the chosen Connection Interval */
		uint16_t min_ce_len = MIN_CE_LEN_20MS_CI;
		@@ -1171,6 +1225,10 @@ class HearingAidImpl : public HearingAid {
		} else {
		log::info("audio_running={}", audio_running);
		}

		// Close the ASRC context.
		ResetAsrc();

		audio_running = false;
		stop_audio_ticks();

		@@ -1213,6 +1271,9 @@ class HearingAidImpl : public HearingAid {
		return;
		}

		// Open the ASRC context.
		ConfigureAsrc();

		// TODO: shall we also reset the encoder ?
		encoder_state_release();
		encoder_state_init();
		@@ -1352,6 +1413,16 @@ class HearingAidImpl : public HearingAid {
		return diff_credit < (init_credit / 2 - 1);
		}

		void OnAudioDataReadyResample(const std::vector<uint8_t>& data) {
		if (asrc == nullptr) {
		return OnAudioDataReady(data);
		}

		for (auto const resampled_data : asrc->Run(data)) {
		OnAudioDataReady(*resampled_data);
		}
		}

		void OnAudioDataReady(const std::vector<uint8_t>& data) {
		/* For now we assume data comes in as 16bit per sample 16kHz PCM stereo */
		bool need_drop = false;
		@@ -1414,6 +1485,18 @@ class HearingAidImpl : public HearingAid {
		l2cap_flush_threshold = 1;
		}

		// Skipping packets completely messes up the resampler context.
		// The condition for skipping packets seems to be easily triggered,
		// causing dropouts that could have been avoided.
		//
		// When the resampler is enabled, the flush threshold is set
		// to the number of credits specified for the ASHA l2cap streaming
		// channel. This will ensure it is only triggered in case of
		// critical failure.
		if (IS_FLAG_ENABLED(asha_asrc)) {
		l2cap_flush_threshold = 8;
		}

		// TODO: monural, binarual check

		// divide encoded data into packets, add header, send.
		@@ -2069,7 +2152,7 @@ void encryption_callback(const RawAddress* address, tBT_TRANSPORT, void*,
		class HearingAidAudioReceiverImpl : public HearingAidAudioReceiver {
		public:
		void OnAudioDataReady(const std::vector<uint8_t>& data) override {
		if (instance) instance->OnAudioDataReady(data);
		if (instance) instance->OnAudioDataReadyResample(data);
		}
		void OnAudioSuspend(const std::function<void()>& stop_audio_ticks) override {
		if (instance) instance->OnAudioSuspend(stop_audio_ticks);